@book{ahlin92,
author = {Lars Ahlin and Jens Zander},
title = {Digital Radiokommunikation - system och metoder},
publisher = {Studentlitteratur},
year = {1992},
address = {Lund},
annote = {Contains some info on FEC}
}
@misc{Allman98,
title = {Enhancing {TCP} Over Satellite Channels using Standard Mechanisms},
author = {Mark Allman and Dan Glover},
month = aug,
year = 1998,
note = {Work in Progress},
annote = { This papers identifies ways in which TCP communication over satellite paths can more efficiently utilize the available capacity. In particular, it focuses only on mechanisms that are currently available and that are on the IETF standards track. First the characteristics for satellite communications is outlined. Specifically the problems of noise and bandwidth are first discussed. Then other characteristics that can degrade the performance of TCP and that differ from most terrestrial channels is presented: \begin{itemize} \item Long feedback loop \item Large delay*bandwidth product \item Transmission errors \item Asymmetric use \item Variable round Trip Times \item Intermittent connectivity \end{itemize} Two non-TCP mechanisms are suggested to help increase TCP performance. The first is Path MTU Discovery to determine the maximum packet size that can used without undergoing IP fragmentation. The second mechanism is forward error correction (FEC). Both of these processes, of course, have disadvantages. Using Path MTU Discovery may cause a large delay before TCP is able to start sending data. And FEC requires special hardware and can introduce delay and timing jitter due to the processing time of the coder/decoder. The real problem, however, is the fact that TCP has no mechanism to distinguish between congestion induced drops and those caused by corruption. From there, the author goes on to discuss standard TCP mechanisms that may help to fully utilize satellite channels. \begin{itemize} \item TCP Congestion Control \begin{itemize} \item Slow Start \item Congestion Avoidance \item Fast Retransmit \item Fast Recovery \end{itemize} \item TCP Large Windows \begin{itemize} \item Window Scaling \item PAWS \item RTTM \end{itemize} \item TCP SACKs \end{itemize} By combining the different techniques that are outlined above, better utilization of satellite channels can be realized. },
status = {INTERNET DRAFT}
}
@article{Amer94,
author = {P. D. Amer and C.Chassot and T. Connolly and M. Diaz and P. Conrad},
title = {Partial order transport service for multimedia and other applications},
journal = {{IEEE/ACM} Transactions on Networking},
year = {1994},
volume = {2},
number = {5},
pages = {440-456},
month = {October},
bibdate = {Saturday, November 21, 1998 at 12:08:56 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@article{Amer98,
author = {P. Amer and S. Iren and G. Sezen and P. Conrad and M. Taube and A. Caro.},
title = {Network-conscious {GIF} image transmission over {Internet}},
journal = {Proc 4th Int'l Workshop on High Performance Protocol Architectures (HIPPARCH) },
year = {1998},
month = {June},
annote = {This paper describe how they modified the GIF Standard to make it suitable for partial order connections. They confine the LZW to one packet instead of the whole image which reduce the compression gain. They also insert position information in each packet so it can be decoded and displayed without any dependance on any other packet.},
url = {papers/hipparch98.ps.gz},
submitter = {Johan Garcia},
bibdate = {Thursday, October 08, 1998 at 08:49:41 (DFT)}
}
@inproceedings{Amir96,
author = {E. Amir and S. McCanne and M. Vetterli},
title = {A layered {DCT} coder for {Internet} video},
booktitle = {Proc. {IEEE} Int. Conf. Image},
year = {1996},
address = {Lausanne, Switzerland},
month = {Oct.},
annote = {Disusses a layered DCT-scheme derived from a (misunderstood?) progressive JPEG. Dicusses some implementation optimizations. },
url = {papers/layered_dct.ps},
submitter = {Johan Garcia}
}
@article{Amir97,
author = {Elan Amir and Steven McCanne and Randy Katz},
title = {Receiver-driven bandwidth adaptation for light-weight sessions },
journal = {Proceedings of ACM Multimedia '97},
year = {1997},
pages = {415-426},
annote = {Describes a protocol (SCUBA) that allows receiver preferences to control the rate adaption performed by multiple multimedia transmitters (multicast). Not very interesting, except the fact that one of the applications they discuss is a media gateway. The media gateway builds on the exact same idea as the "web gateway" we are considering as an application.},
url = {papers/p415-amir.pdf}
}
@article{Asplund98,
author = {Katarina Asplund},
title = {Working Draft 980703 },
journal = {Working draft, University of Karlstad},
year = {1998},
month = {July},
url = {http://www.cs.kau.se/~katarina/prtp/draft980703.html },
submitter = {Katarina Asplund},
bibdate = {Sunday, October 04, 1998 at 19:46:35 (DFT)}
}
@misc{Atmforum97,
title = {Audiovisual Multimedia Services: Video on Demand Specification 1.1},
author = {{ATM Forum}},
month = {March},
year = 1997,
annote = {The ATM Forum together with ITU-T works on defining the many plauslible ATM applications. In 1994, the Audiovisual Multimedia Service (AMS) workgroup listed 17 different applications which their companies wanted specified. Based on priority, they began working on defining a video-on-demand specification. This specification can be downloaded from the reference. It is stored in the file "af-saa-0049.001.pdf". The document is primarily concerned with the interfaces at the hosts in order to provide the services. Especially the relationships between the AMS QoS parameters and the ATM Layer parameters I found useful to us.},
url = {http://www.atmforum.com/atmforum/specs/approved.html},
submitter = {Karl-Johan Grinnemo}
}
@misc{Atmservice,
title = {{ATM} Service Categories: The Benefits to the User},
author = {{ATM Forum}},
annote = {B-ISDN and the technology underlying this service, ATM, are thought to accommodate every conceivable application requirement, especially those originating from multimedia applications. Therefore this paper could be interesting to us in that it describes the service categories and QoS parameters used in ATM to describe application requirements. },
url = {http://www.atmforum.com/atmforum/library/service_categories.html},
key = {Atmforum},
submitter = {Karl-Johan Grinnemo}
}
@article{Ayanoglu95,
author = {E. Ayanoglu and S. Paul and T.F. LaPorta and K.K. Sabnani and R.D. Gitlin},
title = {{AIRMAIL}: A Link-Layer Protocol for Wireless Networks},
journal = {ACM Wireless Networks},
year = 1995,
volume = 1,
number = 1,
pages = {47--60},
month = feb
}
@article{Badrinath95,
author = {B. R. Badrinath and Arup Acharyan},
title = {{MRSVP}: A Reservation Protocol for an Integrated Services Packet Network},
journal = {{DARPA} Technical Report},
year = {1995},
submitter = {Karl-Johan Grinnemo},
bibdate = {Friday, November 27, 1998 at 10:30:16 (NFT)}
}
@article{Bakre94,
author = {A. Bakre and B. R. Badrinath},
title = {{I-TCP}: {Indirect TCP} for Mobile Hosts},
journal = {Technical Report DCS-TR-314, Rutgers University},
year = {1994},
month = {October},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 22, 1998 at 16:29:51 (NFT)}
}
@article{Bakre95,
author = {Ajay Bakre and B. R. Badrinath},
title = {{I-TCP}: {Indirect TCP} for Mobile Hosts},
journal = {15th International Conference on Distributed Computing Systems},
year = {1995},
month = may,
annote = {Technical report DCS-TR-314 / WINLAB TR-89},
url = {http://www.cs.rutgers.edu/\~{}badri/journal/contents11.html}
}
@article{Bakshi96,
author = {B. Bakshi and P. Krishna and N. Vaidya and D. Pradhan},
title = {Improving Performance of {TCP} over Wireless Networks},
journal = {Technical Report 96-014, Texas A \& M University},
year = {1996},
month = {May},
bibdate = {Sunday, November 22, 1998 at 16:39:22 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Balakrishnan95,
author = {H. Balakrishnan and S. Seshan and E. Amir and R. H. Katz},
title = {Improving {TCP/IP} Performance over Wireless Networks},
booktitle = {Proceedings of the First Annual International Conference on Mobile Computing and Networking (MOBICOM)},
year = {1995},
month = nov,
address={Berkeley, CA, USA},
annote = {TCP is a tuned to perform well in a traditional network. In this paper, a design and implementation of a protocol "snoop" is described. Snoop works at the network layer and monitors every packet between a mobile and fixed host. The "snoop" protocol resides on an intermediary, so called base station, that is interposed between the fixed and the mobile host. The main idea of the "snoop" protocol is to cache packets at the base station and perform local retransmissions to the mobile host.},
url = {http://www.cs.berkeley.edu/~ss/papers.html}
}
@article{Balakrishnan97,
author = {Hari Balakrishnan and Venkata N. Padmanabhan and Srinivasan Seshan and Randy H. Katz},
title = {A comparison of mechanisms for improving {TCP} performance over wireless links },
journal = {IEEE/ACM Transactions on Networking},
year = {1997},
volume = {5},
number = {6},
pages = {756-769},
month = dec,
annote = {Compares different approaches for improving TCP performance over wireless links using simulation. Schemes considered include end-to-end proposals (Selective acknowledgements), split-connection proposals (I-TCP), and link-layer proposals (local retransmission and FEC). Provides a good overview of the different schemes proposed. Recomended reading.},
url = {papers/p756-balakrishnan.pdf }
}
@article{Ballardie93,
author = {Anthony Ballardie and Paul Francis and Jon Crowcroft},
title = {Core Based Trees},
journal = {Proc ACM SIGCOMM 93},
year = {1993},
month = sep,
address = {San Francisco}
}
@article{Birman87,
author = {Kenneth Birman and T. A. Joseph},
title = {Reliable Communication in the Presence of Failures},
journal = {ACM Transactions on Computer Systems},
year = 1987,
volume = 5,
pages = {47-76}
}
@article{Birman91,
author = {Kenneth Birman and Andr\'{e} Schiper and Pat Stephenson},
title = {Lightweight Causal and Atomic Group Multicast},
journal = {ACM Transactions on Computer Systems},
year = 1991,
volume = 9 ,
number = 3,
pages = {272-314},
month = {aug},
annote = {Describes the implementation of Isis, which provides CBCAST a fault-tolerant causally consistent multicast primitive. CBCAST can be extended into ABCAST when a total delivery order is needed. The implementation of CBCAST is based on vector time.}
}
@article{Blake98,
author = {S. Blake and D. Black and M. Carlson and E. Davies and Z. Wang and W. Weiss},
title = {An Architecture for Differentiated Services},
journal = {Work in progress. Internet Draft draft-ietf-diffserv-arch-02.txt},
year = {1998},
month = {October},
submitter = {Katarina Asplund},
bibdate = {Thursday, November 26, 1998 at 17:11:02 (NFT)}
}
@inproceedings{Bolot96,
author = {J-C Bolot and T. Turletti},
title = {Adaptive Error Control for Packet Video in the {Internet}},
booktitle = {Proceedings CIP '96, Lausanne},
year = {1996},
month = {September},
annote = {This paper discusses error control mechanisms based on FEC for packet video. The mechanism is much like that presented in the paper "Control Mechanisms for Packet Audio in the Internet"},
bibdate = {Sunday, October 04, 1998 at 19:35:59 (DFT)},
submitter = {Katarina Asplund}
}
@inproceedings{Bolot96a,
author = {J-C Bolot and A Vega-Garcia},
title = {Control Mechanisms for Packet Audio in the {Internet}},
booktitle = {Proceedings IEEE INFOCOM'96, San Fransisco, CA},
year = {1996},
month = {March},
annote = {One way to support real time applications such as interactive audio given a best effort service, is to use control mechanisms that adapt the audio coding and decoding processes based on the characteristics of the channels. This paper describes and analyzes a combined error and rate control mechanism.},
url = {http://www.cs.kau.se/cs/prtp/comments/NS02.html },
submitter = {Katarina Asplund},
bibdate = {Sunday, October 04, 1998 at 19:37:37 (DFT)}
}
@article{Bolot98,
author = {Jean-Crysostome Bolot and André Vega-Garcia},
title = {The Case for {FEC}-Based Error Control for Packet Audio in the {Internet} },
journal = {to appear in ACM Multimedia Systems},
year = {1998},
annote = {Examines packet loss characteristics for audio packets and presents an analytic model for packet audio over the Internet. Also presents a FEC mechanism using a low bitrate encoding (LPC) to provide FEC-data to a PCM encoded audio packet stream. This allows up to 30% of non-continous packet loss before sound detoriates. An adaption of this mechanism could be useful as part of a demonstration application for a PRTP. },
url = {http://www.cs.kau.se/cs/prtp/papers/Bolo96_Audio-FEC.ps },
bibdate = {Sunday, October 04, 1998 at 19:40:23 (DFT)},
submitter = {Johan Garcia}
}
@article{Bottou98,
author = {L. Bottou and P. Haffner and P.G. Howard and P. Simard and Y. Bengio and Y. LeCun},
title = {High Quality Document Image Compression with {DjVu}},
journal = {Journal of Electronic Imaging},
year = {1998},
annote = {This paper describes a novel technique that is to be used to efficiently store scanned documents, color or bw. It uses several steps, (1) separate text and drawings from pictures and the background. (2) Encode the text and images with a bi-level encoder. (3) Encode pictures and background with a wavelet coder. The technique is very efficient and a scanned textpage typically needs < 20k.},
url = {papers/djvu.ps.gz},
bibdate = {Thursday, October 08, 1998 at 09:28:09 (DFT)},
submitter = {Johan Garcia}
}
@incollection{Boumans96,
author = {Jak Boumans},
title = {A Decade of Compact Disc Media},
booktitle = {Contours of Multimedia},
publisher = {John Libbey Media},
year = 1996,
editor = {Nicholas W. Jankowski and Lucien Hanssen},
number = 19,
series = {Acamedia Research Monograph},
pages = {22-31}
}
@inproceedings{Brakmo94,
author = {L. S. Brakmo and S. W. O'Malley and L. L. Peterson},
title = {{TCP Vegas}: New Techniques for Congestion Detection and Avoidance},
booktitle = {Proceedings, 1994 SIGCOMM Conference},
year = 1994,
pages = {24--35},
address = {London, UK},
month = aug # { 31st - } # sep # { 2nd},
annote = { \input{Vegas} }
}
@article{Braun95,
author = {T. Braun and C. Diot and A. Hoglander and V. Roca},
title = {An Experimental User Level Implementation of {TCP}},
journal = {Technical Report, INRIA Sophia Antipolis, France },
year = {1995},
number = {2650},
month = {September},
annote = {The report describe a user level implementation of TCP. The implementation is realised in two parts, one in Kernel-space (demultiplexing), and the rest in user space. It is designed to be compatible to standard TCP implementations, but allow easy experimenting. Performance is reported lower than for kernel TCP due mainly to more user/kernel switches. Source code for the implementation is available.},
url = {papers/rr-2650.ps }
}
@article{Brown97,
author = {Kevin Brown and Suresh Singh},
title = {{M}-{TCP}: {TCP} for Mobile Cellular Networks},
journal = {ACM Computer Communication Review},
year = {1997},
volume = {27},
number = {5},
month = oct,
keywords = {mobile, TCP, protocol}
}
@article{Brunstrom,
author = {Anna Brunstrom},
title = {Slides from Technical Reference Group Meeting 980916},
journal = {Meeting Slides, University of Karlstad},
year = {1998},
month = {September},
url = {papers/ref0916/ref0916.html}
}
@article{Brunstrom98,
author = {A. Brunstrom and K. Asplund and E. Jonsson},
title = {A Partially Reliable Transport Protocol for Stream-Oriented Multimedia Applications},
journal = {Karlstad University},
year = {1998},
submitter = {Karl-Johan Grinnemo},
bibdate = {Saturday, November 21, 1998 at 12:12:55 (NFT)}
}
@article{Brunstrom98a,
author = {Anna Brunstrom},
title = {Research Project Proposal: Analysis and Implementation of a Partially Reliable Transport Protocol for Multimedia Applications},
journal = {Project Proposal, University of Karlstad},
year = {1998},
month = {February},
annote = {The original project proposal for the "PRTP project". Note that the timeschedule for the project have been slightly revised as can be seen in the prestudy specification.},
url = {papers/proposal/prop.html}
}
@article{Brunstrom98b,
author = {Anna Brunstrom},
title = {Analysis and Implementation of a Partially Reliable Transport Protocol for Multimedia Applications: Prestudy Specification},
journal = {Prestudy Specification, University of Karlstad},
year = {1998},
month = {May},
url = {papers/prestudy/prestudy.html }
}
@article{Brunstrom98c,
author = {Anna Brunstrom},
title = {Slides from Steering Committee Meeting 980820},
journal = {Meeting Slides, University of Karlstad},
year = {1998},
month = {August},
url = {papers/styr}
}
@inproceedings{Burger98,
author = {Cora Burger and Kurt Rothermel and Mecklenburg},
title = {Interactive Protocol Simulation Applets for Distance Education},
booktitle = {Interactive Distributed Multimedia Systems and Telecommunication Services},
year = 1998,
editor = {Thomas Plagemann and Vera Goebel},
pages = {29-40},
publisher = {Springer-Verlag},
month = {September},
annote = {my stuff here},
volume = 1483,
series = {Lecture Notes in Computer Science}
}
@article{Caceres94,
author = {Ram\'{o}n C\'{a}ceres and Liviu Iftode},
title = {Improving the Performance of Reliable Transport Protocols in Mobile Computing Environments},
journal = {IEEE JSAC Special Issue on Mobile Computing Network},
year = {1994},
note = {http://www.cs.rutgers.edu/\~{}badri/journal/contents11.html}
}
@article{Caceres95,
author = {Ram\'{o}n C\'{a}ceres and Liviu Iftode},
title = {Improving the Performance of Reliable Transport Protocols in Mobile Computing Environments},
journal = {IEEE Journal on Selected Areas in Communications},
year = {1995},
volume = {13},
month = {June}
}
@article{Cain97,
author = {Brad Cain and Steve Deering and Ajit Thyagarajan},
title = {{Internet Group Management Protocol}, Version 3},
journal = {Work in progress (Internet-Draft draft-ietf-idmr-igmp-v3-00.txt)},
year = {1997},
month = {November},
url = {http://www.ietf.org/internet-drafts/draft-ietf-idmr-igmp-v3-00.txt},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Wednesday, October 14, 1998 at 19:52:38 (DFT)}
}
@inproceedings{Campbell96,
author = {Andrew Campbell and Geoff Coulson},
title = {A {QoS} adaptive transport system: design, implementation and experience},
booktitle = {Proceedings ACM Multimedia 96},
year = {1996},
pages = {117-127},
annote = {Describes an implementation of quite extensive QoS architecture. It is built on top of an ATM network. It includes dynamic QoS adaptation based on layered encoding, jitter control and mechanisms for controlling and managing flows.},
url = {papers/p117-campbell.pdf}
}
@article{Cardelli,
author = {Luca Cardelli},
title = {Mobile Computational Ambients},
journal = {Ambit},
annote = {This paper describes how ambients make it possible to transfer a complete computing environment from a stationary personal computer to a laptop. Formally, an ambient is a "bounded place where computation happens". The concept is more easily grasped if you think of it in terms of an example. An opened document in Word together with its current settings comprises an ambient.},
url = {http://www.luca.demon.co.uk/Ambit/Ambit.html}
}
@article{Chang84,
author = {Jo-Mei Chang and N. F. Maxemchuk},
title = {Reliable Broadcast Protocols},
journal = {ACM Transactions on Computer Systems},
year = 1984,
volume = 2,
number = 3,
pages = {251-273},
month = {August}
}
@inproceedings{Chang98,
author = {Edward Y. Chang},
title = {An Image Coding and Reconstruction scheme for Mobile Computing},
booktitle = {Proceedings IDMS, Oslo, Norway},
year = {1998},
editor = {Thomas Plagemann and Vera Goebel},
publisher = {Springer},
month = {Sept.},
annote = {Describes a scheme to make JPEG coding more error resilient by different DCT coefficients in different packets in order to make reconstruction easier. Can handle up to 50% loss rate.},
submitter = {Johan Garcia}
}
@article{Chen95,
author = {Z. Chen and S. Tan and R. Campbell and Y. Li},
title = {Real time video and audio in the World Wide Web},
journal = {Proc. Fourth International World Wide Web Conference},
year = {1995},
submitter = {Katarina Asplund},
bibdate = {Friday, November 27, 1998 at 10:03:43 (NFT)}
}
@article{Cisco98,
author = {{Cisco Systems Inc.}},
title = {{Cisco IP/TV}},
journal = {{http://www.cisco.com/warp/public\\/732/net\_enabled/iptv/}},
year = {1998},
month = {November},
bibdate = {Wednesday, November 04, 1998 at 15:15:19 (NFT)},
submitter = {Johan Garcia}
}
@techreport{Civanlar98,
author = {M. Reha Civanlar and Glenn L. Cash and Barry G. Haskell},
title = {{AT&T}'s Error Resilient Video Transmission Technique},
institution = {IETF Internet-draft AT&T's Error Resilient Video Transmission Technique (work in progress)},
year = {1998},
month = {July},
annote = {The video transmission is split into high priority and low priority segments. The high priority segments is transmitted on a reliable channel prior to sending the rest on an unreliable channel. The split into high and low priority could be interesting to PRTP since it can provide varying reliability guarantees. },
url = {ftp://ftp.ietf.org/internet-drafts/draft-civanlar-hplp-00.txt },
submitter = {Anna Brunstrom}
}
@inproceedings{Clark88,
author = {D. D. Clark},
title = {The Design Philosophy of the {DARPA} Internet Protocols},
booktitle = {Proceedings of ACM SIGCOMM '88},
year = {1988},
pages = {106--114},
address = {Stanford, CA},
month = aug
}
@inproceedings{Clark90,
author = {David D. Clark and David L. Tennenhouse},
title = {Architectural Considerations for a New Generation of Protocols},
booktitle = {Proceedings ACM SIGCOMM},
year = {1990},
annote = {The classical paper preenting ALF (Application Level Framing) and ILP (Integrated Layer Processing). },
submitter = {Johan Garcia},
bibdate = {Thursday, October 08, 1998 at 10:41:25 (DFT)}
}
@article{Clark92,
author = {D. Clark and S. Shenker and L. Zhang},
title = {Supporting real-time applications in an integrated services packet network: architecture and mechanism},
journal = {Proc. SIGCOMM92 Conference},
year = {1992},
pages = {14-26},
bibdate = {Thursday, November 26, 1998 at 16:54:50 (NFT)},
submitter = {Katarina Asplund}
}
@article{Cohen98,
author = {Reuven Cohen and Srinivas Ramanathan},
title = {{TCP} for high performance in hybrid fiber coaxial broad-band access networks },
journal = {IEEE/ACM Transactions on Networking},
year = {1998},
volume = {6},
number = {1},
pages = {15-29},
month = {February},
annote = {Analysis the impact of high packet-loss on TCP, this time in Hybrid Fiber Coaxial Networks. Suggests that small modifications in the socket size, tuning the TCP MSS value, using finer granularity retransmission timers and a "superfast" retransmit could be used to improve performance. All these changes are localized to the server.},
url = {papers/p15-cohen.pdf }
}
@book{Comer95,
author = {D. E. Comer},
title = {Internetworking with {TCP/IP}},
year = {1995},
volume = {1},
submitter = {Karl-Johan Grinnemo},
journal = {Prentice Hall},
bibdate = {Saturday, November 21, 1998 at 12:15:01 (NFT)}
}
@article{Conrad96,
author = {P. T. Conrad and E. Golden and P. D. Amer and R. Marasli},
title = {A multimedia document retrieval system using partially-ordered/partially-reliable transport service},
journal = {Proceedings of Multimedia Computing and Networking},
year = {1996},
month = {January},
bibdate = {Saturday, November 21, 1998 at 12:18:19 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Creusere96,
author = {Charles Creusere},
title = {A family of image compression algorithms which are robust to transmission errors },
booktitle = {Proceedings of the SPIE},
year = {1996},
address = {Denver, CO},
month = {August},
note = {To appear},
annote = {The paper suggests a way of divide the wavelet coefficients into separate groups, and interleaving them to achieve a degree of loss tolerance. Normally a decoder could have problems to detect small data losses, this is also addresed in the paper. },
url = {papers/robust_spie.ps },
volume = {2825},
submitter = {Johan Garcia}
}
@article{Dalal78,
author = {Yogen K. Dalal and Robert M. Metcalfe},
title = {Reverse Path Forwarding of Broadcast Packets},
journal = {Communications of the ACM},
year = {1978},
volume = {21},
number = {12},
pages = {1040--1048},
month = dec,
abstract = {A broadcast packet is for delivery to all nodes of a network. Algorithms for accomplishing this delivery through a store-and-forward packet switching computer network include (1) transmission of separately addressed packets, (2) multidestination addressing, (3) hot potato forwarding, (4) spanning tree forwarding, and (5) source based forwarding. To this list of algorithms is added (6) reverse path forwarding, a broadcast routing method which exploits routing procedures and data structures already available for packet switching. Reverse path forwarding is a practical algorithm for broadcast routing in store-and-forward packet switching computer networks. The algorithm is described as being practical because it is not optimal according to metrics developed for its analysis in this paper, and also because it can be implemented in existing networks with less complexity than that required for the known alternatives.},
journalabr = {Commun ACM},
bibdate = {Tue Mar 25 13:26:09 MST 1997},
corpsource = {Xerox Corp., Palo Alto, CA, USA},
classification = {723},
classcodes = {C5620 (Computer networks and techniques)},
keywords = {broadcast packets; broadcast routing; computer networks; packet switching; reverse path forwarding},
issn = {0001-0782},
coden = {CACMA2},
treatment = {P Practical},
acknowledgement = ack-nhfb
}
@inproceedings{Danskin97,
author = {Danskin, John M. and Davis, Geoffrey M. and Song, Xiyong},
title = {Fast Lossy Internet Image Transmission},
booktitle = {Proceedings of ACM Multimedia},
year = {1995},
month = {Nov},
annote = {A Image transmission system that uses joint source/channel coding to trade higher compression for more FEC redundancy according to network conditions. Uses a wavelet based coding and implements a transport protocol on top of UDP in order to be TCP-friendly (Vegas style)},
url = {http://www.cs.dartmouth.edu/~jmd/decs/xsong/flit/paper.html},
submitter = {Johan Garcia}
}
@phdthesis{Deering91,
author = {Steve Deering},
title = {Multicast Routing in a Datagram Internetwork},
school = {Stanford University},
year = 1991
}
@article{Deering93,
author = {Steve Deering},
title = {Internet multicast routing: State of the art and open research issues},
journal = {Multimedia Integrated Conferencing for Europe (MICE) Seminar at the Swedish Institute of Computer Science, Stockholm},
year = {1993},
month = {October},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Wednesday, October 14, 1998 at 17:06:14 (DFT)}
}
@article{Deering98,
author = {Stephen Deering and Deborah Estrin and Dino Farinacci and Van Jacobson and Ahmed Helmy and David Meyer and Liming Wei},
title = {Protocol Independent Multicast Version 2 Dense Mode Specification},
journal = {Work in progress (Internet-Drafts draft-ietf-pim-v2-dm-00.txt)},
year = {1998},
month = {August},
url = {http://www.ietf.org/internet-drafts/draft-ietf-pim-v2-dm-00.txt},
bibdate = {Tuesday, October 13, 1998 at 19:52:14 (DFT)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Deering98b,
author = {S. Deering and D. Estrin and D. Farinacci and A. Helmy and D. Thaler and and M. Handley and V. Jacobson and C. Liu and P. Sharma and L. Wei},
title = {Protocol Independent Multicast-Sparse Mode ({PIM-SM}): Motivation and Architecture},
journal = {Work in progress (Internet-Draft draft-ietf-idmr-pim-arch-05.txt)},
year = {1998},
month = {August},
url = {http://www.ietf.org/internet-drafts/draft-ietf-idmr-pim-arch-05.txt},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Tuesday, October 13, 1998 at 20:14:09 (DFT)}
}
@article{Demers90,
author = {A. Demers and S. Keshav and S. Shenker},
title = {Analysis and Simulation of a Fair Queueing Algorithm},
journal = {Journal of Internetworking: Research and Experience 1},
year = {1990},
pages = {3-26},
submitter = {Katarina Asplund},
bibdate = {Thursday, November 26, 1998 at 17:17:23 (NFT)}
}
@inproceedings{Dempsey94,
author = {B. Dempsey and M. Lucas and A. Weaver},
title = {Design and implementation of a high quality video distribution system using {XTP} reliable multicast},
booktitle = {Proceedings 2nd Int. Workshop on Advanced Communications and Applications for High-Speed Networks},
year = {1994},
month = {September},
annote = {Gives a quite detailed description of the implementation of a video distribution system on top of XTP's reliable multicast protocol.},
url = {papers/iwaca94.ps}
}
@article{Dempsey94b,
author = {B. J. Dempsey},
title = {Retransmission-Based Error Control For Continuous Media},
journal = {PhD thesis, University of Virginia},
year = {1994},
submitter = {Katarina Asplund},
bibdate = {Friday, November 27, 1998 at 10:10:07 (NFT)}
}
@article{Dempsey96,
author = {Bert J. Dempsey and Jorg Liebeherr and Alfred C. Weaver},
title = {On retransmission-based error control for continuous media traffic in packet-switched networks.},
journal = {Computer Networks and ISDN Systems},
year = {1996},
volume = {28},
pages = {719-736},
annote = {Provides analytic eveidence on the feasibility of retransmission based error recovery even for voice communication. Pretty much a piece from Dempsey's dissertation. Skip the math.},
url = {papers/isdn96.ps}
}
@inproceedings{Diot95,
author = {C. Diot and C. Huitema and T. Turletti},
title = {Multimedia Applications should be Adaptive},
booktitle = {Proceedings HPCS'95, Mystic (CN)},
year = {1995},
month = {August},
annote = {The paper argues that multimedia application should be implemented in a network-conscsious manner. This means that they are aware on the type of underlying network they are operating upon, and that they are able to handle network conditions that occur on this network. The authors exemplify with a video conferencing system that reduces the output data rate depending on the packet loss ration of the network and argue that adaption have several benefits over resource reservation, primarily less complexity. },
url = {papers/hpcs95.ps },
submitter = {Johan Garcia},
bibdate = {Sunday, October 04, 1998 at 19:23:07 (DFT)}
}
@article{Diot97,
author = {C. Diot and W. Dabbous and J. Crowcroft.},
title = {Multipoint Communication: A Survey of Protocols, Functions and Mechanisms},
journal = {IEEE Journal on Selected Area in Communication. Special Issue on GroupCommunication.},
year = {1997},
month = {May},
annote = {Survey covers most issues related to multicast. Contains a vast number of references. },
url = {papers/jsac-xcast.ps.gz},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Wednesday, October 14, 1998 at 17:20:57 (DFT)}
}
@inproceedings{Dooly98,
author = {Daniel R. Dooly and Sally A. Goldman and Stephen D. Scott},
title = {{TCP} Dynamic Acknowledgment Delay: Theory and Practice},
booktitle = {Proceedings of the 30th Annual {ACM} Symposium on Theory of Computing ({STOC}-98)},
year = {1998},
pages = {389--398},
publisher = {ACM Press},
address = {New York},
month = may # {23--26~},
isbn = {0-89791-962-9}
}
@inproceedings{Ensor98,
author = {J. Robert Ensor and Gianpaolo U. Carraro and John T. Edmark},
title = {Visual Techniques to Accommodate Varying Network Performance in Virtual Environments},
booktitle = {Interactive Distributed Multimedia Systems and Telecommunication Services},
year = 1998,
editor = {Thomas Plagemann and Vera Goebel},
pages = {41-46},
publisher = {Springer-Verlag},
month = {September},
series = {Lecture Notes in Computer Science},
volume = 1483
}
@article{Eriksson94,
author = {Hans Eriksson},
title = {{MBONE}: the multicast backbone},
journal = {Communications of the ACM},
year = {1994},
volume = {37},
number = {8},
pages = {54--60},
month = aug,
url = {papers/p54-eriksson.pdf},
subject = {{\bf C.2.1}: Computer Systems Organization, COMPUTER-COMMUNICATION NETWORKS, Network Architecture and Design, Network communications. {\bf C.2.1}: Computer Systems Organization, COMPUTER-COMMUNICATION NETWORKS, Network Architecture and Design, Internet. {\bf C.2.1}: Computer Systems Organization, COMPUTER-COMMUNICATION NETWORKS, Network Architecture and Design, Network topology. {\bf F.2.2}: Theory of Computation, ANALYSIS OF ALGORITHMS AND PROBLEM COMPLEXITY, Nonnumerical Algorithms and Problems, Routing and layout. {\bf C.2.2}: Computer Systems Organization, COMPUTER-COMMUNICATION NETWORKS, Network Protocols.},
coden = {CACMA2},
keywords = {design; management; performance},
acknowledgement = ack-nhfb,
bibdate = {Thu May 30 09:41:10 MDT 1996},
issn = {0001-0782}
}
@article{Fall96,
author = {Fall, K and Floyd, S},
title = {Simulation-based Comparisons of {Tahoe}, {Reno} and {SACK} {TCP}},
journal = {Computer Communication Review},
year = 1996,
volume = 26,
number = 3 ,
pages = {5-21},
month = {July}
}
@article{Fischer92,
author = {Yuval Fischer},
title = {Fractal Image Compression},
journal = {SIGGRAPH '92 Course notes },
year = {1992},
annote = {An introduction to fractal image compression. },
url = {papers/fractal_paper.ps.gz},
bibdate = {Thursday, October 08, 1998 at 11:20:37 (DFT)},
submitter = {Johan Garcia}
}
@article{Fischer94,
author = {Fischer, Y. and Rogovin, D. and Shen, T.P.},
title = {A Comparison of Fractal Methods with {DCT} and Wavelets},
journal = {Proc. SPIE, Neural and Stochastic Methods in Image and Signal Processing III, Su-Shing Chen; Ed.},
year = {1994},
volume = {2304},
pages = {132-143},
month = {June},
annote = {The paper contains a brief description of fractal image compression methods with sample compression results. They also present comparative results between two fractal schemes, discrete cosine transform, and a wavelet method. They show that, with the PSNR as a measure of image quality, some fractal schemes perform best over the range of compressions of most interest. },
url = {papers/spie.ps.gz},
submitter = {Johan Garcia},
bibdate = {Thursday, October 22, 1998 at 15:39:54 (DFT)}
}
@unpublished{Floyd95,
author = {S. Floyd},
title = {{TCP} and Successive Fast Retransmits},
note = {ftp://ftp.ee.lbl.gov/papers/fastretrans.ps},
year = {February 24, 1995},
annote = { This paper discusses problems found in the Tahoe and Reno implementations of TCP. In these implementations Fast Retransmit can be invoked more than once in one round trip time. This problem occurs for TCP connections with a large congestion window which experience multiple packet drops within one window of data. One fix to this problem is to not treat duplicate ACKs that acknowledge packets from the window of a previous Fast Retransmit as an inication of congestion. This can be implemented in Tahoe TCP by adding an extra variable \em{high\_seq} to keep track of the highest sequence number outstanding when the TCP initiated a Fast Retransmit or responded to an ECN (Explicit Congestion Notification). On problem with this fix is that the TCP source cannot distinguish duplicate acks resulting from retransmitted packets that had previously been correctly received and duplicate acks resulting from packet losses. A more appropriate solution to this problem is to use Selective ACKs. },
journal = {Unpublished technical note}
}
@article{Floyd96,
author = {S. Floyd and V. Jacobson and S. McCanne and C. G. Liu and L. Zhang},
title = {A Reliable Multicast Framework for Light-weight Sessions and Application Level Framing},
journal = {{IEEE/ACM} Transactions on Networking},
year = {1996},
month = {November},
annote = {Describes the SRM protocol used in wb.},
url = {papers/srm.ps},
bibdate = {Wednesday, October 14, 1998 at 16:47:05 (DFT)},
submitter = {Anna Brunstr\"{o}m}
}
@techreport{Floyd96a,
author = {S. Floyd},
title = {Issues of {TCP} with {SACK}},
institution = {Lawrence Berkeley National Laboratory},
year = 1996 ,
month = {March}
}
@article{Forman98,
author = {George H. Forman},
title = {Ensuring Responsiveness Despite Resource Variability and Volatility},
journal = {External TR HPL-98-15, Hewlett-Packard Labs},
year = {1998},
annote = {Applications running in a mobile computing environment are prone to a great deal of variability in the response time they experience from system services. In this paper, Forman elaborates on the causes of response time variability, and the challenges it puts on the applications running on a mobile host. He also enumerates som ways to obtaining acceptable response behaviour despite the variability of the wireless channel. },
url = {http://www.cs.washington.edu/homes/forman/pubs.html}
}
@article{G11496,
author = {ITU-T},
title = {Recommendation {G.114} - One-way transmission time},
journal = {Geneva, Switzerland},
year = {1996},
month = {February},
submitter = {Johan Garcia},
bibdate = {Friday, November 06, 1998 at 09:25:15 (NFT)}
}
@article{G71188,
author = {ITU-T},
title = {Recommendation {G.711} - Pulse code modulation ({PCM}) of voice frequencies},
journal = {ITU-T},
year = {1988},
submitter = {Johan Garcia},
bibdate = {Friday, November 06, 1998 at 09:14:08 (NFT)}
}
@article{G723196,
author = {ITU-T},
title = {Recommendation {G.723.1} - Dual rate speech coder for multimedia communications},
journal = {Geneva, Switzerland},
year = {1996},
month = {March},
submitter = {Johan Garcia},
bibdate = {Friday, November 06, 1998 at 09:16:30 (NFT)}
}
@article{G72690,
author = {ITU-T},
title = {Recommendation {G.726} - 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation},
journal = {Geneva, Switzerland},
year = {1990},
month = {December},
bibdate = {Friday, November 06, 1998 at 09:18:12 (NFT)},
submitter = {Johan Garcia}
}
@article{G72790,
author = {ITU-T},
title = {Recommendation {G.727} - 5-, 4-, 3- and 2-bits sample embedded adaptative differential pulse code modulation ({ADPCM})},
journal = {Geneva, Switzerland},
year = {1990},
month = {December},
submitter = {Johan Garcia},
bibdate = {Friday, November 06, 1998 at 09:20:32 (NFT)}
}
@article{G72996,
author = {ITU-T},
title = {Recommendation {G.729} - Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction ({CS-ACELP})},
journal = {Geneva, Switzerland},
year = {1996},
month = {March},
bibdate = {Friday, November 06, 1998 at 09:22:23 (NFT)},
submitter = {Johan Garcia}
}
@inbook{Garcia-Luna-Aceves98,
author = {J. J. Garcia-Luna-Aceves and Brian Neil Levine},
title = {Multimedia Communications: Protocols and Appplications},
chapter = {6: End-to-End Reliable Multicast},
pages = {169-191},
publisher = {Prentice Hall},
year = 1998,
annote = {Good survey on reliable multicast}
}
@article{Garcia98,
author = {Johan Garcia},
title = {Multimedia Coding for Partially Reliable Channels},
journal = {Working draft, University of Karlstad},
year = {1998},
month = {August},
annote = {The paper provides a classification of coding properties, an overview of some coding techniques and a discussion on some of the problems of partially reliable TCP. The major obstacle is how to cater for the need of explicit decoder resynchronization that arises when we have data loss and no application level visible packet boundaries. The paper elaborates on the ideas from a presentation at the 980805 research meeting.},
url = {papers/multimedia_encoding.ps},
submitter = {Johan Garcia},
bibdate = {Sunday, October 04, 1998 at 19:45:16 (DFT)}
}
@article{Gecsei97,
author = {Jan Gecsei},
title = {Adaption in Distributed Multimedia Systems},
journal = {IEEE Multimedia},
year = {1997},
volume = {4},
number = {2},
pages = {58--66},
month = {Apr.--June},
annote = {Discusses adaption froam a broad viewpoint.},
submitter = {Johan Garcia}
}
@article{Georgiadis96,
author = {L. Georgiadis and R. Guerin and V. Peris and R. Rajan},
title = {Efficient Support of Delay and Rate Guarantees in an Internet},
journal = {Proceedings of SIGCOMM96},
year = {1996},
pages = {106-116},
month = {August},
submitter = {Katarina Asplund},
bibdate = {Thursday, November 26, 1998 at 16:45:46 (NFT)}
}
@book{Gibson98,
author = {J.D. Gibson and T. Berger and T. Lookabaugh and D. Lindberg and R. L. Baker},
title = {Digital Compression for Multimedia},
publisher = {Morgan Kaufmann Publishers},
year = {1998},
address = {San Fransisco, CA},
annote = {A deatiled book covering speech and video coding(H.26X and MPEG 1 & 2), both principles and standards. Contains a small chapter on JPEG as well.},
submitter = {Johan Garcia},
bibdate = {Thursday, October 08, 1998 at 08:52:41 (DFT)}
}
@misc{Gif89a,
title = {{Graphics Interchange Format}, Version 89a },
howpublished = {Technical Report, {Compuserve Incorporated}, Columbus, Ohio},
month = {July},
year = {1989},
annote = { The GIF image coding standard. GIF is a lossless coding using Variable-Length-Code LZW compression, a variation on the Lempel-Ziv compression. It can code bw and color pictures with max 256 colors. (ca 25 pg) },
url = { papers/h_gif89a.html },
submitter = {Johan Garcia},
key = {Gif89a}
}
@article{Gong92,
author = {F. Gong and G. Parulkar},
title = {An Application-Oriented Error Control Scheme for High-Speed Networks},
journal = {Technical Report WUCS.92-37 Department of Computer Science, Washington University},
year = {1992},
month = {November},
submitter = {Karl-Johan Grinnemo},
bibdate = {Saturday, November 21, 1998 at 12:26:56 (NFT)}
}
@article{Gong96,
author = {F. Gong and G. Parulkar},
title = {An Application-Oriented Error Control Scheme for High-Speed Networks},
journal = {IEEE/ACM Transactions on Networking},
year = {1996},
volume = {4},
number = {5},
pages = {669-683},
month = {October},
bibdate = {Friday, November 27, 1998 at 10:23:49 (NFT)},
submitter = {Katarina Asplund}
}
@article{Goyal98,
author = {Rohit Goyal and Raj Jain and Shiv Kalyanaraman and Sonia Fahmy and Bobby Vandalore},
title = {Improving the Performance of {TCP} over the {ATM-UBR} service},
journal = {Computer Communications},
year = {1998},
volume = {21},
number = {10},
pages = {898-911},
month = jul,
annote = {To appear.},
url = {http://www.cis.ohio-state.edu/~jain/papers/cc.htm},
coden = {COCOD7},
bibdate = {Mon Aug 17 17:45:43 1998},
issn = {0140-3664}
}
@article{Gringeri98,
author = {Gringeri, Steven and Khasnabish, Bhumip and Lewis, Arianne and Shuaib, Khaled and Egorov, Roman and Basch, Bert},
title = {Transmission of {MPEG-2} Video Streams over {ATM}},
journal = {IEEE Multimedia},
year = {1998},
volume = {5},
number = {1},
pages = {58--71},
month = {Jan.-March},
annote = {Reports on experiments of using comercial encoders/decoders of networks with induced losses. Also discusses error concealment and hierarchical coding},
submitter = {Johan Garcia}
}
@article{Gupta98,
author = {Rajarshi Gupta and Mike Chen and Steven McCanne and Jean Walrand},
title = {{WebTP}: A Receiver-Driven Web Transport Protocol },
journal = {Submitted to Infocom'99},
year = {1998},
annote = {Suggests that a new transport protocol is needed for Web transport. Interesting in that it outlines problems with TCP, some of which could be reduced by PRTP, and in that it suggests that the protocol should be controlled at the receiver side since that's where user preferences can be accounted for.},
url = {papers/infocom.ps}
}
@article{H26193,
author = {ITU-T},
title = {Recommendation {H.261} - Video codec for audiovisual services at p x 64 kbit/s},
journal = {Geneva, Switzerland},
year = {1993},
month = {March},
bibdate = {Friday, November 06, 1998 at 09:26:40 (NFT)},
submitter = {Johan Garcia}
}
@article{H26396,
author = {ITU-T},
title = {Recommendation {H.263} - Video coding for low bit rate communication},
journal = {Geneva, Switzerland},
year = {1996},
month = {March},
submitter = {Johan Garcia},
bibdate = {Friday, November 06, 1998 at 09:27:37 (NFT)}
}
@article{H32398,
author = {ITU-T},
title = {Recommendation {H.323} - Packet-based multimedia communications systems},
journal = {Geneva, Switzerland},
year = {1998},
month = {February},
bibdate = {Friday, November 06, 1998 at 09:32:57 (NFT)},
submitter = {Johan Garcia}
}
@article{H323primer,
title = {A Primer on the {H.323}. Series Standard},
journal = {Whitepaper, DataBeam Corporation, Lexington, KY},
year = {1998},
annote = {Provides a good architectural overview of the structure of H.323 and the component standards for video, audio, data and control. },
url = {papers/h323.ps},
bibdate = {Tuesday, November 24, 1998 at 13:11:52 (NFT)},
submitter = {Johan Garcia},
key = {H323primer}
}
@article{H32498,
author = {ITU-T},
title = {Recommendation {H.324} - Terminal for low bit rate Multimedia Communication},
journal = {Geneva, Switzerland},
year = {1998},
month = {February},
submitter = {Johan Garcia},
bibdate = {Friday, November 06, 1998 at 09:35:32 (NFT)}
}
@inproceedings{Han96,
author = {Han, Richard Y. and Messerschmitt, David G.},
title = {Asymptotically Reliable Transport of Multimedia/Graphics Over Wireless Channels},
booktitle = {Proceedings of Multimedia and Networking},
year = {1996},
address = {San Jose, CA},
month = {Jan. 29-31},
annote = {Proposes a 'leacky-ARQ' scheme that forwards corrupt packets to the application while still requesting retransmissions. Useful because of the error characteristics of mobile links, not so useful in regular networks where packets as whole are lost.},
url = {papers/asymp.pdf}
}
@article{Han96b,
author = {Jefferson Han and Brian Smith },
title = {{CU-SeeMe VR} immersive desktop teleconferencing},
journal = {ACM Multimedia 96},
pages = {199--207},
url = {http://simon.cs.cornell.edu/Info/Projects/multimedia/Papers/Vr/vr.htm},
bibdate = {Wednesday, October 14, 1998 at 13:33:43 (DFT)},
submitter = {Johan Garcia}
}
@article{Hardman95,
author = {Hardman, V. and Sasse, M.A. and Handley, M. and Watson, A.},
title = {Reliable Audio for use over the {Internet}},
journal = {Proceedings of INET, Oahu, Hawaii},
year = {1995},
annote = {Paper describing much the same as the work done on audio coding at INRIA. Provides some good general background material though.},
submitter = {Johan Garcia},
bibdate = {Thursday, October 22, 1998 at 15:51:00 (DFT)}
}
@article{Hardman98,
author = {Vicky Hardman and Martina Angela Sasse and Isidor Kouvelas},
title = {Successfull multiparty audio communication over the {Internet}},
journal = {Communications of the ACM},
year = {1998},
volume = {41},
number = {5},
pages = {74-80},
month = {May},
annote = {Describes the problems involved in audio transmission over the Mbone and how these problems have been addressed in RAT. Most of the discussion is relevant for point-to-point audio communication as well. Possible reference for numbers on packetloss frequency, loss tolerance etc. }
}
@inproceedings{Hartung97,
author = {Hartung, John and Jacquin, Arnaud and Pawlyk, James and Shipley, Kathleen},
title = {A real-time scalable software video codec for collaborative applications over packet networks},
booktitle = {Proceeding of ACM Multimedia},
year = {1998},
address = {Bristol, UK},
month = {Sept.},
annote = {Uses three layers and layer specific rate control and some simple reconstruction-minded source coding.},
url = {http://www.kom.e-technik.tu-darmstadt.de/pr/workshop/chair/AC MMM98/electronic_proceedings/hartung/index.html},
submitter = {Johan Garcia}
}
@article{Heidemann97,
author = {John Heidemann and Katia Obraczka and Joe Touch},
title = {Modeling the performance of {HTTP} over several transport protocols },
journal = {IEEE/ACM Transactions on Networking},
year = {1997},
volume = {5},
number = {5},
pages = {616-630},
month = {October},
annote = {Analytic study of HTTP over TCP, persistent-connection TCP, T/TCP, and a UDP based request-response protocol. Intro and related work section worth skimming.},
url = {papers/p616-heidemann.pdf}
}
@article{Heidemann97a,
author = {John Heidemann},
title = {Performance interactions between {P-HTTP} and {TCP} implementations},
journal = {Computer Communication Review},
year = {1997},
volume = {27},
number = {2},
pages = {64-73},
month = {April},
annote = {Describes three different problems associated with the then (1997) current TCP and Apache versions. The problems were examined in a LAN setting and two of the problems related to the use of delayed acks. The first problem occur because of application bug workaround irregularities in the sending of mime headers which causes a non-full mime header to be sent which as seconde segment. Since the second segment is only partial, an ack-every-other will not be triggered, instead the ack will be delayed 200ms. Since the window at this time is 2 segments, no data can be sent in the meantime. The second problem also causes delayed ack waiting at the end of connections when Nagles alg is enabled and there is an odd number of segments with the last segment being smaller than half of the recv win (not bypassing nagle). The third problem discussed is the reseting of the congestion window that some TCP implementations do after a RTT or RTO. When using a PHTTP connection on the same host and moving between different webpages, each new webpage will cause slow start. This is deemed too conservative by the author, but seems somewhat reasonable as the network load may have changed considerably between web page requests. Suggests a rate-based approach to pace outgoing packets after a pause in transmission.},
url = {papers/Heidemann97PHTTP_TCP_interact.ps.gz},
bibdate = {Monday, March 25, 2002 at 07:53:35 (CET)},
submitter = {Johan Garcia}
}
@article{Hellwig89,
author = {K. Hellwig and P. Vary and D. Massaloux and J.P. Petit},
title = {Speech Codec for European Mobile Radio System},
journal = {IEEE Global Telecommunications },
year = {1989},
pages = {1065-1069},
annote = {Describes GSM},
submitter = {Johan Garcia},
bibdate = {Monday, November 16, 1998 at 19:06:12 (NFT)}
}
@inproceedings{Hemami95,
author = {S. S. Hemami},
title = {Digital Image Coding for Robust Multimedia Transmission},
booktitle = {Symposium on Multimedia Communications and Video Coding},
year = {1995},
address = {New York, NY},
month = {Oct.},
annote = {Discusses different techniques for image reconstruction, both decoder only approximation according to non-blocking assumption and reconstruction optimized source coding. See Chang for JPEG variation.},
url = {papers/img_coding_robust.ps},
submitter = {Johan Garcia}
}
@mastersthesis{Hoe95,
author = {J. C. Hoe},
title = {Start-up Dynamics of {TCP}'s Congestion Control and Avoidance Schemes},
school = {Massachusetts Institute of Technology},
year = 1995,
month = {June}
}
@inproceedings{Hoe96,
author = {J. C. Hoe},
title = {Improving the Start-up Behavior of a Congestion Control Sheme for {TCP}},
booktitle = {Proceedings of the {ACM} {SIGCOMM} Conference on Applications, Technologies, Architectures, and Protocols for Computer Communications},
year = {1996},
pages = {270--280},
publisher = {ACM Press},
address = {New York},
month = aug # {~26--30~},
isbn = {0-89791-790-1},
series = {ACM SIGCOMM Computer Communication Review},
volume = {26,4}
}
@article{I.321,
author = {ITU-T},
title = {Recommendation {I.321} - {B-ISDN} protocol reference model and its applications},
journal = {Geneva,Switzerland},
year = {1991},
month = {April},
bibdate = {Thursday, November 26, 1998 at 17:30:11 (NFT)},
submitter = {Katarina Asplund}
}
@article{IEEEMultimedia98,
title = {Special issue on Multimedia in Medicine},
journal = {IEEE Multimedia},
year = {1997},
volume = {4},
number = {2},
month = {April},
key = {IEEEMultimedia},
submitter = {Johan Garcia},
bibdate = {Wednesday, November 04, 1998 at 15:28:52 (NFT)}
}
@inproceedings{Ingvaldsen98,
author = {Tone Ingvaldsen and Espen Klovning and Mike Wilkins},
title = {A Study of Delay Factors in {CSCE} Applications and Their Importance},
booktitle = {Proceedings of IDMS},
year = {1998},
editor = {Thomas Plagemann and Vera Goebel},
publisher = {Springer},
address = {Oslo, Norway},
month = {Sept.},
annote = {States that some applications that are less interactive can tolerate upp to 900ms latency while still be acceptable to the users.},
subitter = {Johan Garcia}
}
@article{Ioannidis93,
author = {J. Ioannidis and G. Q. Maguire Jr.},
title = {The Design and Implementation of Mobile Internetworking Architecture},
journal = {Proceedings of the Winter USENIX Conference},
year = {1993},
month = {January},
bibdate = {Sunday, November 22, 1998 at 15:49:28 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@article{Iren98,
author = {S. Iren and P. Amer and P. Conrad},
title = {{NETCICATS}: Network-conscious Image Compression and Transmission System},
journal = {Proc 4th Int'l Workshop on Multimedia Information, Istanbul, Turkey},
year = {1998},
month = {September},
annote = {This papers presents the NETCICATS testing system for network-conscious images (not so intersesting) , but also says some about network-conscious Wavelet Zerotree encoding.},
url = {papers/mis98.ps.gz},
submitter = {Johan Garcia},
bibdate = {Thursday, October 08, 1998 at 11:03:25 (DFT)}
}
@inproceedings{Iren98a,
author = {Sami Iren and Paul D. Amer and Phillip T. Conrad},
title = {Network-conscious Compressed Images over Wireless Networks},
booktitle = {Interactive Distributed Multimedia Systems and Telecommunication Services},
year = 1998,
editor = {Thomas Plagemann and Vera Goebel},
pages = {149-157},
publisher = {Springer-Verlag},
month = {September},
annote = {my stuff here},
series = {Lecture Notes in Computer Science},
volume = 1483
}
@article{Jacobs97,
author = {Stephen Jacobs and Alexandros Eleftheriadis},
title = {A Real Time Protocol that Guarantess Fairness with {TCP}},
journal = {submitted to ACM/IEEE Transactions on Networking},
year = {1997},
month = {October},
annote = {The paper describes a partially reliable protocol to be used for multimedia communication. The protocol is based on UDP but has a congestion windows that allows it to adapt to network conditions in the same way as TCP. It feedbacks rate information to the sending application to make it produce data at the available rate. The receiver can send selective ARQ, and since it knows its playout buffer length and an estimate of RTT is sent in each packet, it will only send ARQs for packets that can be resent without buffer underrun. The paper also presents experimental results obtained with the protocol.},
url = {papers/ton97.ps}
}
@inproceedings{Jacobson88old,
author = {Van Jacobson},
title = {{Congestion Avoidance and Control}},
booktitle = {{Proceedings, SIGCOMM '88 Workshop}},
year = {1988},
pages = {314--329},
organization = {ACM SIGCOMM},
publisher = {ACM Press},
month = aug,
note = {Stanford, CA}
}
@misc{Jacobson90,
title = {Modified {TCP} Congestion Avoidance Algorithm},
author = {Van Jacobson},
howpublished = {end2end-interest mailing list},
month = apr,
year = 1990 ,
note = {ftp://ftp.isi.edu/end2end/end2end-interest-1990.mail}
}
@techreport{Jain87,
author = {R. Jain and K. Ramakrishnan and D. Chiu},
title = {Congestion Avoidance in Computer Networks with a Connectionless Network Layer},
institution = {DEC},
year = {1987},
number = {DEC-TR-506},
note = {DEC-TR-506, reprinted in C. Partridge, Ed., {\em Innovations in Internetworking}, 140--156, published by Artech House, October 1988.},
url = {http://www.cis.ohio-state.edu/~jain/papers/cr5.htm},
abstract = {Widespread use of computer networks and the use of varied technology for the interconnection of computers has made congestion a significant problem.\par In this report, we summarize our research on congestion avoidance. We compare the concept of congestion avoidance with that of congestion control.\par Briefly, congestion control is a recovery mechanism, while congestion avoidance is a prevention mechanism. A congestion control scheme helps the network to recover from the congestion state while a congestion avoidance scheme allows a network to operate in the region of low delay and high throughput with minimal queuing, thereby preventing it from entering the congested state in which packets are lost due to buffer shortage.\par A number of possible alternatives for congestion avoidance were identified. From these alternatives we selected one called the binary feedback scheme in which the network uses a single bit in the network layer header to feed back the congestion information to its users, which then increase or decrease their load to make optimal use of the resources. The concept of global optimality in a distributed system is defined in terms of efficiency and fairness such that they can be independently quantified and apply to any number of resources and users.\par The proposed scheme has been simulated and shown to be globally efficient, fair, responsive, convergent, robust, distributed, and configuration-independent.},
pages = {17},
bibdate = {Mon Aug 17 17:45:43 1998}
}
@article{Jayant93,
author = {N. Jayant},
title = {High Quality Networking of Audio-Visual Information},
journal = {IEEE Communications},
year = {1993},
pages = {84-95},
month = {September},
bibdate = {Saturday, November 21, 1998 at 12:28:59 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@misc{Jpeg92,
title = {Recommendation {T.81} - Digital compression and coding of continuous-tone still images},
author = {ITU-T},
howpublished = {Geneva, Switzerland},
month = {September},
year = {1992},
annote = {The JPEG standard. (182 pg)},
url = {papers/itu-11505_t81e.ps},
submitter = {Johan Garcia},
bibdate = {Friday, November 27, 1998 at 09:15:56 (NFT)}
}
@article{Karlsson89,
author = {G. Karlsson and M. Vetterli},
title = {Packet Video and Its Integration into the Network Architecture},
journal = {IEEE Journal on Selected Areas in Communications},
year = {1989},
pages = {739-751},
month = {June},
bibdate = {Saturday, November 21, 1998 at 12:31:32 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Karn87,
author = {Phil Karn and Craig Partridge},
title = {Improving Round-Trip Time Estimates in Reliable Transport Protocols},
booktitle = {{Proceedings, SIGCOMM '87 Workshop}},
year = {1987},
pages = {2--7},
organization = {ACM SIGCOMM},
publisher = {ACM Press},
month = aug,
note = {Stowe, Vermont}
}
@inproceedings{Karn93,
author = {Phil Karn},
title = {The {Qualcomm CDMA} Digital Cellular System},
booktitle = {Proceedings of the {USENIX} Mobile and Location-Independent Computing Symposium},
year = 1993,
pages = {35--39},
month = aug,
isbn = {1-880446-51-0},
acknowledgement = ack-nhfb,
affiliation = {Qualcomm, Inc.},
day = {2--3},
bibdate = {Thu Feb 22 08:12:14 MST 1996}
}
@article{Kojo94,
author = {M. Kojo and K. Raatikainen and T. Alanko},
title = {Connecting Mobile Workstations to the {Internet} over a Digital Cellular Telephone Network},
journal = {Proceedings Workshop on Mobile and Wireless Information Systems (Mobidata), Rutgers University, NJ},
year = {1994},
month = {November},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 22, 1998 at 16:43:34 (NFT)}
}
@article{Krawczyk97,
author = {J. Krawczyk},
title = {Designing Tunnels for Interoperability with {RSVP}},
journal = {Internet Draft},
year = {1997},
month = {March},
bibdate = {Friday, November 27, 1998 at 10:27:35 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Kunkelmann98,
author = {Thomas Kunkelmann and Uwe Horn},
title = {Video Encryption Based on Data Partitioning and Scalable Coding -- A Comparison},
booktitle = {Interactive Distributed Multimedia Systems and Telecommunication Services},
year = 1998,
editor = {Thomas Plagemann and Vera Goebel},
pages = {95-106},
publisher = {Springer-Verlag},
month = {September},
annote = {my stuff here},
volume = 1483,
series = {Lecture Notes in Computer Science}
}
@book{Kuo98,
author = {Franklin Kuo and Wolfgang Effelsberg and J. J. Garcia-Luna-Aceves},
title = {Multimedia Communications: Protocols and Appplications},
publisher = {Prentice Hall},
year = 1998
}
@article{Kurose93,
author = {Jim Kurose},
title = {Open Issues and Challenges in Providing Quality of Service Guarantees in High-Speed Networks},
journal = {ACM Computer Communication Review},
year = {1993},
volume = {23},
number = {1},
pages = {6-15},
month = {January},
submitter = {Katarina Asplund},
bibdate = {Thursday, November 26, 1998 at 16:36:57 (NFT)}
}
@article{Lakshman97,
author = {T. V. Lakshman and Upamanyu Madhow},
title = {The performance of {TCP/IP} for networks with high bandwidth-delay products and random loss },
journal = {IEEE/ACM Transactions on Networking},
year = {1997},
volume = {5},
number = {3},
pages = {336-350},
month = {June},
annote = {Compares the performance of TCP-Tahoe (Jacobsons original congestion control) and TCP-Reno (revised scheme using fast retransmit/recovery) through analysis and simulation. Quite mathematical. },
url = {papers/p336-lakshman.pdf}
}
@article{Li98,
author = {Kien Li and Annika Wennstr\"{o}m},
title = {Technical Report on {WAP}-Applications},
journal = {KS/EIN Technical Report},
year = {1998},
month = {July},
annote = {The first part of the document describes the services and applications that WAP targets. In the second part of the document, a prototype for finding employees in Ericsson's phone book is presented. I find this document very useful for those who wants to see WAP from a user's perspective.}
}
@techreport{Liljeberg96,
author = {Mika Liljeberg and Heikki Helin and Markku Kojo and Kimmo Raatikainen},
title = {Enhanced Services for {World-Wide Web} in Mobile {WAN} Environment},
institution = {Department of Computer Science, University of Helsinki},
year = {1996},
number = {C-1996-28},
month = {April},
annote = {The report describes the MOWGLI system which incorporates a Data Channel Service (i.e. tailored, more effective, transport protocol) for the mobile link, HTTP header compression, content filtering & recompression (GIF->JPEG), user-initiated prefetching and cacheing. The reported performance of the Dara channel Service is 150-200% of standard TCP and suggests that a gain is also achieveable by PRTP, although slow-start problems will still be present },
url = {ftp://ftp.cs.helsinki.fi/pub/Reports/by_Title/Enhanced_Services_for_World-Wide_Web_in_Mobile_WAN_Environment.ps.gz},
bibdate = {Wednesday, October 14, 1998 at 09:20:37 (DFT)},
submitter = {Johan Garcia}
}
@article{Lin96,
author = {John C. Lin and Sanjoy Paul},
title = {{RMTP}: A Reliable Multicast Transport Protocol },
journal = {Proceedings of {IEEE} Infocom},
year = {1996},
pages = {1414-1424},
month = {March},
annote = {RMTP uses a tree-based approach to implement reliable multicast. A receiver-initiated mechanism is used to maintain reliability within the local groups.},
url = {papers/rmtp.ps},
bibdate = {Wednesday, October 14, 1998 at 16:55:49 (DFT)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Lucas97,
author = {Matthew T. Lucas and Bert J. Dempsey and Alfred C. Weaver},
title = {{MESH}: Distributed Error Recovery for Multimedia Streams in Wide-Area Multicast Networks},
journal = {IEEE International Conference on Communication (ICC '97), Montreal, CA},
year = {1997},
pages = {1127--1133},
month = {June},
annote = {Shows that retransmission based error recovery can be a viable alternative to forward error correction even for multicast over a WAN. However the protocol seem very complex to me. Consequently the results are obtained through simulation results rather than implementation and measurements.},
url = {papers/icc97.ps}
}
@article{MPEG193,
author = {ISO/IEC},
title = {{MPEG-1} Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s},
journal = {ISO/IEC 11172},
year = {1993},
bibdate = {Friday, November 06, 1998 at 09:49:56 (NFT)},
submitter = {Johan Garcia}
}
@article{MPEG296,
author = {ISO/IEC},
title = {{MPEG-2} Generic coding of moving pictures and associated audio information},
journal = {ISO/IEC 13818},
year = {1996},
bibdate = {Friday, November 06, 1998 at 09:51:54 (NFT)},
submitter = {Johan Garcia}
}
@article{MPEG498,
author = {ISO/IEC},
title = {Overview of the {MPEG-4} Standard},
journal = {ISO/IEC JTC1/SC29/WG11 N2323, http://www.cselt.it/mpeg/standards\\/mpeg-4/mpeg-4.htm},
year = {1998},
month = {July},
submitter = {Johan Garcia},
bibdate = {Friday, November 06, 1998 at 10:00:28 (NFT)}
}
@article{Mahdavi98,
author = {J. Mahdavi and S. Floyd },
title = {The {TCP}-Friendly Website},
journal = {http://www.psc.edu/networking/tcp\_friendly.html},
year = {1998},
month = {November},
url = {http://www.psc.edu/networking/tcp\_friendly.html},
submitter = {Sean Schneyer},
bibdate = {Tuesday, November 17, 1998 at 11:18:19 (NFT)}
}
@article{Marasli96,
author = {R. Marasli and P. D Amer and P. T. Conrad},
title = {Retransmission-Based Partially Reliable Transport Service: An Analytic Model},
journal = {Proceedings of INFOCOM '96},
year = {1996},
month = {March},
bibdate = {Saturday, November 21, 1998 at 12:37:04 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@article{Marasli97,
author = {R. Marasli and P. D. Amer and P. T. Conrad},
title = {Partially Reliable Transport Service},
journal = {Proceedings of the 2nd IEEE Symposium on Computers and Communications ({ISCC})},
year = {1997},
month = {July},
submitter = {Karl-Johan Grinnemo},
bibdate = {Saturday, November 21, 1998 at 12:43:57 (NFT)}
}
@article{Marasli97b,
author = {R. Marasli},
title = {Partially Ordered and Partially Reliable Transport Protocols: Performance Analysis},
journal = {PhD thesis, University of Delaware},
year = {1997},
bibdate = {Saturday, November 21, 1998 at 12:33:35 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@article{Mayer-Patel96,
author = {Ketan Mayer-Patel and David Simpson and David Wu and Lawrence A. Rowe},
title = {Synchronized continuous media playback through the {World Wide Web}},
journal = {Proceedings ACM Multimedia 96},
year = {1996},
pages = {435-436},
annote = {Gives a highlevel description of an application called cmplayer. However the app and the CMT toolkit on which it is built is available at .},
url = {papers/p435-mayer-patel.pdf}
}
@article{McCanne95,
author = {S. McCanne and V. Jacobson},
title = {Vic: A Flexible Framework for Packet Video},
journal = {{ACM} Multimedia},
year = {1995},
pages = {511-522},
month = {November},
bibdate = {Wednesday, October 14, 1998 at 20:10:30 (DFT)},
submitter = {Anna Brunstr\"{o}m}
}
@inproceedings{McCanne96,
author = {S. McCanne and V. Jacobson and M. Vetterli},
title = {Receiver-driven Layered Multicast},
booktitle = {Proceedings of the {ACM} {SIGCOMM} Conference on Applications, Technologies, Architectures, and Protocols for Computer Communications},
year = {1996},
pages = {117--130},
publisher = {ACM Press},
address = {New York},
month = aug # {26--30~},
url = {papers/McCa96_Receiver-SIGCOMM96.ps},
isbn = {0-89791-790-1},
series = {ACM SIGCOMM Computer Communication Review},
volume = {26,4}
}
@article{Meer98,
author = {H. de Meer and J-P Richter and A. Puliafito and O. Tomarchio},
title = {Tunnel Agents for Enhanced Internet {QoS}},
journal = {IEEE Concurrency},
year = {1998},
volume = {6},
number = {2},
pages = {30-39},
annote = {Suggests using agent technology for providing QoS features in routers that don't support RSVP along a path of (mostly RSVP) routers. Tangential to our work. No nedd to read unless you find the topic interesting in itself.}
}
@article{Merz97,
author = {Michael Merz and Konrad Froitzheim and Peter Schulthess and Heiner Wolf},
title = {Iterative transmission of media streams },
journal = {Proceedings of ACM Multimedia '97},
year = {1997},
pages = {283-290},
annote = {Describes a method to transfer video (and other) media streams in a progressive fashion. The video is broken down and part of each frame or a selective set of frames is transmitted first, allowing quick viewing although of lesser quality. Appears most useful for short prerecorded videoclips.},
url = {papers/p283-merz.pdf}
}
@techreport{Microsoft97,
author = {{Microsoft Corporation}},
title = {Microsoft {NetMeeting} 2.0: Overview and Frequently Asked Questions},
institution = {Microsoft Corporation},
year = {1997},
month = {July},
annote = {This paper describes NetMeeting 2.0, a videoconferencing program from Microsoft. Users of NetMeeting can collaborate and share information with two or more conference participants in real-time. With a video-capture card and camera, users of NetMeeting can send and receive video images for face-to-face communication during a conference. The video-conferencing capability of NetMeeting is based on the ITU H.323 standard. Using a sound card, microphone, and speakers, the participants of a conference can talk to each other over Internet and corporate intranets. The Internet telephony capability of NetMeeting is based on the ITU H.323 standard. },
submitter = {Karl-Johan Grinnemo}
}
@article{Microsoft98,
author = {{Microsoft Corporation}},
title = {Windows Media Technologies },
journal = {http://www.microsoft.com/windows/windowsmedia/default.asp},
year = {1998},
month = {November},
submitter = {Johan Garcia},
bibdate = {Wednesday, November 04, 1998 at 15:13:02 (NFT)}
}
@article{Mitzel94,
author = {D. Mitzel and D. Estrin and S. Shenker and L. Zhang},
title = {An Architectural Comparison of {ST-II} and {RSVP}},
journal = {Proceedings of Infocom 94},
year = {1994},
bibdate = {Thursday, November 26, 1998 at 17:24:24 (NFT)},
submitter = {Katarina Asplund}
}
@article{Ohta91,
author = {H. Ohta and K. Kitami},
title = {A Cell Loss Recovery Method using {FEC} in {ATM} Networks},
journal = {IEEE Journal on Selected Areas in Communication},
year = {1991},
pages = {1471-1483},
month = {December},
bibdate = {Saturday, November 21, 1998 at 12:46:03 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Ott97,
author = {Teunis J. Ott and Neil Aggarwal},
title = {{TCP} over {ATM}: {ABR} or {UBR} ?},
booktitle = {Proceedings of the 1997 {ACM} {SIGMETRICS} International Conference on Measurement and Modeling of Computer Systems},
year = {1997},
pages = {52--63},
publisher = {ACM Press},
address = {New York},
month = jun # {15--18~},
isbn = {0-89791-909-2},
series = {Performance Evaluation Review},
volume = {25,1}
}
@misc{Painter97,
title = {A Review of Algorithms for Perceptual Coding of Audio Signals },
author = {Painter, E.M and Spanias, A.S.},
howpublished = {Portions submitted to DSP-97.},
annote = {The introduction in section 1 and conclusions in section 6 can be used to get some overview of the field of audio coding (i.e music, not voice). Other sections discuss among other things wavelet subband coding and MPEG layer 3. (30 pg, 122 refs) },
url = {papers/dsp97.ps },
submitter = {Johan Garcia}
}
@techreport{Papadopoulos95,
author = {Papadopoulos, Christos and Parulkar, Guru},
title = {Error Control for Continous Media and Multipoint Applications},
year = {1995},
month = {Dec},
annote = {Contains a discussion on error control and proposes an error control mechanism based on selective retransmission, conditional (only request not.outdated packets) retransmission, playout buffering gap-based loss detection. Some interseting general discussion on error control.},
url = {papers/wucs95-35.ps},
howpublished = {Tecnical Report WUCS-95-35, Washington University }
}
@article{Papadopoulos96,
author = {Papadopoulos, Christos and Parulkar, Guru M.},
title = {Retransmission-Based Error Control for Continuous Media Applications},
journal = {Proc. 6th Intl. Workshop on Network and Operating System Support for Digital Audio and Video (NOSSDAV)},
year = {1996},
month = {November},
annote = {Discusses the same mechanisms as Papadopoulos95 but also presents some measurements on an implementation over Ethernet / ATM.},
url = {papers/nossdav96.ps},
submitter = {Johan Garcia},
bibdate = {Wednesday, October 07, 1998 at 14:53:57 (DFT)}
}
@article{Paul95,
author = {S. Paul and E. Ayanoglu and T. F. LaPorta},
title = {An Assymetric Link-Layer Protocol for Digital Cellular Communications},
journal = {Proceedings of INFOCOM '95},
year = {1995},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 22, 1998 at 16:27:33 (NFT)}
}
@article{Paxson98,
author = {V. Paxson, Editor},
title = {Known {TCP} Implementation Problems},
journal = {Internet-Draft draft-ietf-tcpimpl-prob-04.txt},
year = {1998},
month = {August},
annote = {Hopefully the implementation we start with doesn't exhibit any of these problems, but it's still semi-interesting skimming through the material.},
url = {ftp://ftp.ietf.org/internet-drafts/draft-ietf-tcpimpl-prob-04.txt}
}
@article{Pearce97,
author = {M. Pearce},
title = {{TCP} Performance Over Wireless Channels},
journal = {Motorola Internal Report},
year = {1997},
month = {January},
bibdate = {Sunday, November 22, 1998 at 16:40:51 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@article{Perkins94,
author = {C. E. Perkins and A. Myles and D. B. Johnson},
title = {{IMHP}: A Mobile Host Protocol for the Internet},
journal = {Computer Networks},
year = {1994},
month = {December},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 22, 1998 at 15:51:46 (NFT)}
}
@article{Perkins94b,
author = {C. E. Perkins and A. Myles},
title = {Mobile {IP}},
journal = {Proceedings of International Telecommunications Symposium},
year = {1994},
pages = {415-419},
month = {August},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 22, 1998 at 15:54:45 (NFT)}
}
@article{Peterson89,
author = {Larry L. Peterson and Nick C. Buchholz and Richard D. Schlichting},
title = {Preserving and Using Context Information in Interprocess Communication},
journal = {{ACM} Transactions on Computer Systems},
year = 1989,
volume = 7,
number = 3,
pages = {217-246},
month = {August},
annote = {Description of Psynch, which provides causally consistent communication within process groups. The implementation is based on maintaining a view of the context graph at each node.}
}
@inproceedings{Pingali94,
author = {Sridhar Pingali and Don Towsley and James F. Kurose},
title = {A Comparison of Sender-Initiated and Receiver-Initiated Reliable Multicast Protocols},
booktitle = {Proceedings of the Sigmetrics Conference on Measurement and Modeling of Computer Systems},
year = {1994},
pages = {221--230},
publisher = {ACM Press},
address = {New York, NY, USA},
month = may,
isbn = {0-89791-659-X}
}
@article{Poduri98,
author = {K. Poduri and K. Nichols},
title = {Simulation Studies of Increased Initial {TCP} Window Size},
journal = {Internet-Draft draft-ietf-tcpimpl-poduri-02.txt (work in progress)},
year = {1998},
month = {August},
annote = {Interesting primarily in that we can see how they set up their simulations, what parameters they used, what they measured etc. - good input when we design our simulations. The simulator is publically available and could be an alternative. Also, increased initial window size is one TCP modification that can improve TCP performance for real-time apps.},
url = {ftp://ftp.ietf.org/internet-drafts/draft-ietf-tcpimpl-poduri-02.txt }
}
@article{Pusateri98,
author = {T. Pusateri},
title = {Distance Vector Multicast Routing Protocol},
journal = {Work in progress (Internet-Drafts draft-ietf-idmr-dvmrp-v3-07.txt)},
year = {1998},
month = {August},
url = {http://www.ietf.org/internet-drafts/draft-ietf-idmr-dvmrp-v3-07.txt},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Tuesday, October 13, 1998 at 19:40:13 (DFT)}
}
@misc{RFC0760,
author="J. Postel",
title="{DoD standard Internet Protocol}",
series="Request for Comments",
number="760",
howpublished="RFC 760",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1980,
month=jan,
url="http://www.ietf.org/rfc/rfc760.txt",
}
@misc{RFC0777,
title = {{RFC 777}: {Internet Control Message Protocol}},
author = {J. Postel},
month = apr,
year = {1981},
bibdate = {Thu Oct 16 09:34:09 MDT 1997},
day = {1}
}
@misc{rfc0791,
author="J. Postel",
title={Internet Protocol},
series="Request for Comments",
number="791",
howpublished="RFC 791 (Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1981,
month=sep,
url="http://www.ietf.org/rfc/rfc791.txt",
}
@misc{rfc0768,
author="J. Postel",
title={User Datagram Protocol},
series="Request for Comments",
number="768",
howpublished="RFC 768 (Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1980,
month=aug,
url="http://www.ietf.org/rfc/rfc768.txt",
}
@misc{rfc0793,
author="J. Postel",
title={Transmission Control Protocol},
series="Request for Comments",
number="793",
howpublished="RFC 793 (Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1981,
month=sep,
note="Updated by RFCs 1122, 3168, 6093",
url="http://www.ietf.org/rfc/rfc793.txt",
}
@misc{RFC0801,
title = {{RFC 801}: {NCP\slash TCP} transition plan},
author = {J. Postel},
month = nov,
year = {1981},
day = {1}
}
@misc{RFC0896,
title = {{RFC 896}: Congestion control in {IP}\slash {TCP} internetworks},
author = {J. Nagle},
month = jan ,
year = {1984},
day = {6}
}
@misc{RFC0908,
title = {{RFC 908}: Reliable Data Protocol},
author = {D. Velten and R. M. Hinden and J. Sax},
month = jul,
year = {1984},
note = {Updated by RFC1151. Status: EXPERIMENTAL.},
bibdate = {Thu Oct 16 09:34:09 MDT 1997},
day = {1}
}
@misc{RFC0998,
title = {{RFC 998}: {NETBLT}: {A} bulk data transfer protocol},
author = {D. D. Clark and M. L. Lambert and L. Zhang},
month = mar ,
year = {1987},
bibdate = {Thu Oct 16 09:34:09 MDT 1997},
day = {1}
}
@misc{RFC1045,
title = {{RFC 1045}: {VMTP}: Versatile Message Transaction Protocol: Protocol specification},
author = {D. R. Cheriton},
month = feb,
year = {1988},
day = {1}
}
@misc{RFC1072,
title = {{RFC 1072}: {TCP} extensions for long-delay paths},
author = {V. Jacobson and R. T. Braden},
month = oct,
year = 1988,
annote = { This document claims that current implementations (current as of the writing of the document) of TCP have a serious performance bottleneck for transmissions along paths with high bandwidth and long round-trip delays. These paths are referred to as a "long, fat pipe" and the network containing such a path is referred to as an "LFN". The document defines three problems with the current TCP over LFN paths and solutions to each of them: \begin{description} \item[Window size Limitations] A new TCP option is to be defined. This option will allow for larger windows by specifying a scale factor. This scale factor is to be used to multiply the window size value found in a TCP header to obtain the true window size. \item[Cumulative Acknowledgments] Selective acknowledgements (SACK) are suggested to solve this problem. \item[Round Trip Timing] Accurate round trip times (RTT) are important for both reliability and retransmission of TCP segments. A TCP "echo" option is suggested for dealing with this area. \end{description} },
day = 1,
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1075,
title = {{RFC 1075}: Distance Vector Multicast Routing Protocol},
author = {D. Waitzman and C. Partridge and S. E. Deering},
month = nov,
year = {1988},
day = {1}
}
@misc{RFC1112,
title = {{RFC 1112}: Host extensions for {IP} multicasting},
author = {S. E. Deering},
month = aug,
year = {1989},
day = {1},
bibdate = {Sat Jan 10 08:59:55 MST 1998}
}
@misc{RFC1151,
title = {{RFC 1151}: Version 2 of the Reliable Data Protocol ({RDP})},
author = {C. Partridge and R. M. Hinden},
month = apr,
year = {1990},
day = {1}
}
@misc{RFC1185,
title = {{RFC 1185}: {TCP} extension for high-speed paths},
author = {V. Jacobson and R. T. Braden and L. Zhang},
month = oct,
year = {1990},
annote = { This RFC describes a small extension to TCP to support reliable operation over very high-speed paths. TCP was designed when transmission speeds were a mere 56kbps. As faster transmission speeds are being introduced we are coming closer and closer to the limitations imposed by the TCP protocol itself. Specifically, this RFC addresses two TCP issues: reliability and performance. Because of the way in which TCP operates, performance depends on not just the transfer rate, but on the product of the transfer rate and the round-trip delay. High transfer rates, however, can affect reliability by violating the assumptions behind the TCP mechanism for duplication detection and sequencing. Two ways in which duplication of sequence numbers can occur are described: \begin{enumerate} \item Sequence number wrap-around on the current connection \item Segment from an earlier connection incarnation \end{enumerate} The reliability of TCP depends on the existence of a bound on the lifetime of a segment. This bound is known as the Maximum Segment Lifetime (MSL). One way to avoid the problem of cycling the sequence space is to increase the size of the TCP sequence number field. The authors suggests that previous proposals to correct this problem using this method have "tended towards complexity and ugliness". A simple solution to this problem is defined using the time-stamping methods defined in RFC 1072\cite{RFC1072}. },
day = {1},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1190,
title = {{RFC 1190}: Experimental {Internet Stream Protocol}: Version 2 ({ST-II})},
author = {C. Topolcic},
month = oct,
year = {1990},
bibdate = {Thu Oct 16 09:34:09 MDT 1997},
day = {1}
}
@misc{rfc1191,
author="J.C. Mogul and S.E. Deering",
title={Path {MTU} discovery},
series="Request for Comments",
number="1191",
howpublished="RFC 1191 (Draft Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1990,
month=nov,
url="http://www.ietf.org/rfc/rfc1191.txt",
}
@misc{RFC1256,
title = {{RFC 1256}: {ICMP} Router Discovery Messages},
author = {S. Deering},
month = sep,
year = {1991},
day = {1},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1263,
title = {{RFC 1263}: {TCP} extensions considered harmful},
author = {S. O'Malley and L. L. Peterson},
month = oct,
year = {1991},
annote = { This paper is a direct response to proposals made in RFC 1072\cite{RFC1072} and RFC 1185\cite{RFC1185}. This paper specifically addresses the changes suggested in the previously mentioned RFCs and also addresses possible enhancements to the way in which protocols in general can deal with evolution. The paper argues against trying to maintain backward compatibility with older releases claiming that designing a backward compatible change to a new protocol is extremely difficult and that these implementations tend to be complex and ugly. The authors propose an approach to protocol change which they call protocol evolution. The goal of protocol evolution is to escape the limitations of trying to maintain backward compatibility without incurring all of the costs of creating a new protocol. Several points are given to support the advantages of using protocol evolution: \begin{itemize} \item{Simplest type of modification to make to a protocol, thus less likely to be buggy} \item{No need to consider effects on previous versions of the protocol} \item{Most of the protocol is reused in the new version, eliminating the design and debugging costs of a completely new protocol} \item{No artificial limit on the amount of change that can be made, thus the useful lifetime can be extended indefinitely} \item{Backwards compatibility is automatically maintained since the old version of the protocol is still around} \end{itemize} An entire framework is presented on how this scheme could be realized. The author argues that the only additional cost of this scheme is the memory required for keeping around two copies of the protocol that is evolving. This, however, does not take into account the fact that a protocol can evolve several times. Does this mean we must keep a copy of every possible version of a particular protocol? This point could be a serious arguing point against the use of evolution over backward compatibility. Although the other seems to favor protocol evolution over backward compatibility, he does offer a suggestion to when backward compatible changes are most appropriate. These changes are best suited for changes that do not affect the protocol's header, and do not require the other end of the connection to be changed as well. },
bibdate = {Thu Oct 16 09:34:09 MDT 1997},
day = {1}
}
@misc{rfc1323,
author="V. Jacobson and R. Braden and D. Borman",
title={{TCP} Extensions for High Performance},
series="Request for Comments",
number="1323",
howpublished="RFC 1323 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1992,
month=may,
url="http://www.ietf.org/rfc/rfc1323.txt",
annote = { This document combines RFC 1072\cite{RFC1072} and RFC 1185\cite{RFC1185} giving a more detailed specification. The previous two documents were experimental, whereas this document is aimed at specifying a standards track protocol. The timestamp mechanisms have now been given the names RTTM (round Trip Time Measurement) and PAWS (Protect Against Wrapped Sequences). A summary of changes from RFC 1072\cite{RFC1072} and RFC 1185\cite{RFC1185} is the best way to describe this document: \begin{enumerate} \item SACK has been deferred to a later memo \item The detailed rules for sending timestamp replies has been revised \item Timestamp information can now be shared between RTTM and PAWS \item Timestamps are sent on both ACK and data segments \item The echo and echo reply options of RFC 1072 were combined into a single Timestamps option \item Problem of outdated timestamps on long-idle connections resolved. \item Timestamp check should now be performed before header prediction \item Extended options will be sent on $<$SYN,ACK$>$ segments only when they are received in the corresponding $<$SYN$>$ segments. \end{enumerate} },
}
@misc{RFC1584,
title = {{RFC 1584}: Multicast Extensions to {OSPF}},
author = {J. Moy},
month = mar,
year = {1994},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1633,
title = {{RFC 1633}: Integrated Services in the {Internet} Architecture: an Overview},
author = {R. Braden and D. Clark and S. Shenker},
month = jun,
year = {1994},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1644,
title = {{RFC 1644}: {T\slash TCP} --- {TCP} Extensions for Transactions Functional Specification},
author = {R. Braden},
month = jul,
year = {1994},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1693,
title = {{RFC} 1693: An Extension to {TCP}: Partial Order Service},
author = {T. Connolly and P. Amer and P. Conrad},
month = nov ,
year = {1994}
}
@misc{RFC1700,
title = {{RFC 1700}: {ASSIGNED NUMBERS}},
author = {J. Reynolds and J. Postel},
month = oct,
year = {1994},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1883,
title = {{RFC 1883}: {Internet Protocol}, Version 6 ({IPv6}) Specification},
author = {S. Deering and R. Hinden},
month = dec,
year = {1995},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1889,
title = {{RFC 1889}: {RTP}: {A} Transport Protocol for Real-Time Applications},
author = {H. Schulzrinne and S. Casner and R. Frederick and V. Jacobson},
month = jan,
year = {1996},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1890,
title = {{RFC 1890}: {RTP} Profile for Audio and Video Conferences with Minimal Control},
author = {{Audio-Video Transport Working Group} and H. Schulzrinne},
month = jan,
year = {1996},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC1945,
title = {{RFC 1945}: {Hypertext Transfer Protocol} --- {HTTP/1.0}},
author = {T. Berners-Lee and R. Fielding and H. Frystyk},
month = may,
year = {1996},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC2001,
author="W. Stevens",
title={{TCP} Slow Start, Congestion Avoidance, Fast Retransmit, and Fast Recovery Algorithms},
series="Request for Comments",
number="2001",
howpublished="RFC 2001 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1997,
month=jan,
alt_note="Obsoleted by RFC 2581",
url="http://www.ietf.org/rfc/rfc2001.txt",
}
@misc{RFC2002,
title = {{RFC 2002}: {IP} Mobility Support},
author = {C. Perkins},
month = oct,
year = {1996},
bibdate = {Sat Mar 21 15:53:14 1998}
}
@misc{rfc2018,
author="M. Mathis and J. Mahdavi and S. Floyd and A. Romanow",
title={{TCP} Selective Acknowledgment Options},
series="Request for Comments",
number="2018",
howpublished="RFC 2018 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1996,
month=oct,
url="http://www.ietf.org/rfc/rfc2018.txt",
}
@misc{RFC2068,
title = {{RFC 2068}: {Hypertext Transfer Protocol} --- {HTTP/1.1}},
author = {R. Fielding and J. Gettys and J. Mogul and H. Frystyk and T. Berners-Lee},
month = jan ,
year = {1997},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC2131,
title = {{RFC 2131}: Dynamic Host Configuration Protocol},
author = {R. Droms},
month = mar ,
year = {1997},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC2189,
title = {{RFC 2189}: {Core Based Trees} ({CBT} version 2) Multicast Routing},
author = {A. Ballardie},
month = sep,
year = {1997},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC2190,
title = {{RFC 2190}: {RTP} Payload Format for {H.263} Video Streams},
author = {C. Zhu},
month = sep,
year = {1997},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC2198,
title = {{RFC 2198}: {RTP} Payload for Redundant Audio Data},
author = {C. Perkins and I. Kouvelas and O. Hodson and V. Hardman and M. Handley and J. C. Bolot and A. Vega-Garcia and S. Fosse-Parisis},
month = sep,
year = {1997},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC2201,
title = {{RFC 2201}: {Core Based Trees} ({CBT}) Multicast Routing Architecture},
author = {A. Ballardie},
month = sep,
year = {1997},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC2205,
title = {{RFC 2205}: {Resource ReSerVation Protocol} ({RSVP}) --- Version 1 Functional Specification},
author = {R. {Braden, Ed.} and L. Zhang and S. Berson and S. Herzog and S. Jamin},
month = sep,
year = {1997},
bibdate = {Thu Oct 16 09:34:09 MDT 1997}
}
@misc{RFC2210,
title = {{RFC 2210}: The Use of {RSVP} with {IETF} Integrated Services},
author = {J. Wroclawski},
month = sep,
year = {1997}
}
@misc{RFC2236,
title = {{RFC 2236}: {Internet Group Management Protocol}, Version 2},
author = {W. Fenner},
month = nov,
year = {1997},
bibdate = {Sat Jan 10 08:59:55 MST 1998}
}
@misc{RFC2250,
title = {{RFC 2250}: {RTP} Payload Format for {MPEG1\slash MPEG2} Video},
author = {D. Hoffman and G. Fernando and V. Goyal and M. Civanlar},
month = jan,
year = {1998},
bibdate = {Sat Mar 21 15:14:14 MST 1998}
}
@misc{RFC2328,
title = {{RFC 2328}: {OSPF} Version 2},
author = {J. Moy},
month = apr,
year = {1998}
}
@misc{RFC2354,
title = {{RFC 2354}: Options for Repair of Streaming Media},
author = {C. Perkins and O. Hodson},
month = jun,
year = {1998},
annote = {This paper discusses various techniques for the repair of continuous media streams subject to packet loss. The discussion is restricted to such techniques that require the involvement of the sender, and to applications which use RTP/UDP transport to deliver continuous media streams. The techniques discussed are retransmission, interleaving and forward error coding. }
}
@misc{RFC2362,
title = {{RFC 2362}: Protocol Independent Multicast-Sparse Mode ({PIM-SM}): Protocol Specification},
author = {D. Estrin and D. Farinacci and A. Helmy and D. Thaler and S. Deering and M. Handley and V. Jacobson and C. Liu and P. Sharma and L. Wei},
month = jun,
year = {1998}
}
@misc{RFC2398,
title = {{RFC 2398}: Some Testing Tools for {TCP} Implementors},
author = {S. Parker and C. Schmechel},
month = aug,
year = {1998}
}
@misc{RFC2414,
title = {{RFC 2414}: Increasing {TCP}'s Initial Window},
author = {M. Allman and S. Floyd and C. Partridge},
month = sep,
year = {1998}
}
@misc{RFC2435,
title = {{RFC 2435}: {RTP} Payload Format for {JPEG}-compressed Video},
author = {L. Berc and W. Fenner and R. Frederick and S. McCanne and P. Stewart},
month = oct,
year = {1998}
}
@misc{RFC0925,
title = {{RFC 925}: Multi-{LAN} address resolution},
author = {J. Postel},
month = oct,
year = {1984},
bibdate = {Thu Oct 16 09:34:09 MDT 1997},
day = {1}
}
@article{Real98,
author = {{RealNetworks Inc.}},
title = {{RealAudio} {G2} Music Codec -- A Musical Stream Apart},
journal = {Whitepaper },
year = {1998},
annote = {Describes the the audio part of the next generation RealNetworks G2 player which is capable of bandwidth scaling, has a high degree of discretness and uses interleaving to improve the data reconstruction capability of the decoder. This is stated to lead to packet losses of 10-15% to only minimal gradation reduction of audio quality. },
url = {http://www.real.com/devzone/library/whitepapers/music.html},
bibdate = {Wednesday, October 14, 1998 at 16:11:46 (DFT)},
submitter = {Johan Garcia}
}
@misc{Real98b,
author = {{RealNetworks Inc.}},
title = {Real Player},
note = {\url{http://www.real.com} Visited 2011-09-09},
}
@article{Reichmeyer98,
author = {Francis Reichmeyer and Kwok Chan and David Durham and Raj Yavatkar and Silvano Gai and Keith McCloghrie and Shai Herzog},
title = {{COPS} Usage for Differentiated Services},
journal = {Internet-Draft Draft-ietf-RAP-COPS-DS-00.txt (work in progress)},
year = {1998},
month = {August},
annote = {Describes enhancements to the Common Open Policy Service (COPS) protocol (developed by the IETF RSVP Admission Policy WG) to support policy services in a Differentiated Services (diff serv) environment. Bridges the gap between the RSVP and Differented Services effort. },
url = {ftp://ftp.ietf.org/internet-drafts/draft-ietf-rap-cops-ds-00.txt}
}
@article{Rosenberg98,
author = {J. Rosenberg and H. Schulzrinne},
title = {An {RTP} Payload Format for Generic Forward Error Correction},
journal = {Internet-Draft draft-ietf-avt-fec-03.txt (work in progress)},
year = {1998},
month = {July},
annote = {Describes a RTP payload for parity/Reed-Solomon FEC. The FEC information is sent on a separate channel from the media data allowing it to be used even when a subset of the receivers doesn't support the payload. Inetrnet-Drafts describing the RTP payload for various audio/video encodings can be found at .},
url = {ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-fec-03.txt}
}
@article{Rosenholtz96,
author = {Ruth Rosenholtz and Andrew B. Watson },
title = {Perceptual adaptive {JPEG} coding },
journal = {IEEE International Conference on Image Processing, Lausanne, Switzerland, pp. 901-904.},
year = {1996},
volume = {1},
pages = {901-904},
submitter = {Johan Garcia},
bibdate = {Thursday, November 05, 1998 at 15:49:50 (NFT)}
}
@article{Saltzer84,
author = {J. H. Saltzer and D. P. Reed and D. D. Clark},
title = {End-To-End Arguments in System Design},
journal = {ACM Transactions on Computer Systems},
year = {1984},
volume = {2},
number = {4},
pages = {277--288},
month = nov ,
abstract = {This paper presents a design principle that helps guide placement of functions among the modules of a distributed computer system. The principle, called the end-to-end argument, suggests that functions placed at low levels of a system may be redundant or of little value when compared with the cost of providing them at that low level. Examples discussed in the paper include bit-error recovery, security using encryption, duplicate message suppression, recovery from system crashes, and delivery acknowlegement. Low-level mechanisms to support these functions are justified only as performed enhancements.},
keywords = {design; data communication; protocol design; design principles},
key = {Saltzer et al.},
bibdate = {Wed Mar 6 11:12:06 1985}
}
@techreport{Schulzrinne92,
author = {Henning Schulzrinne},
title = {Voice communication across the {Internet}: a network voice terminal},
institution = {University of Massachusetts at Amherst, Dept. of Computer and Information Science},
year = {1992},
type = {COINS technical report},
number = {92--50},
address = {Amherst, MA, USA},
month = jun,
annote = {This report describes NEVOT, a network voice terminal meant to support multiple concurrent, both two-party and multiparty conferences, on top of a variety of transport protocols. As it is to be used as an experimental tool, it offers extensive configuration, trace and statistics options. NEVOT is available free from the Internet. Design guidelines for a packet voice protocol are given, based on experiments with existing protocols. Unlike most other data transferred across the network, voice and video streams have to obey strict synchronization constraints. The receiver has to maintain synchronization in the face of randomly distributed packet delays, asynchronous operating system scheduling and packet loss. NEVOT support three different transport protocols, namely UDP, TCP and ST-II, and its structure is therefore somewhat complicated. With UDP, the program has to deal with packets that are lost, corrupt, duplicated or out-of-order. With TCP and ST-II, delays can be unacceptable long. The report clearly shows that today?s transport protocols not are well suited for real-time applications, but instead it takes a lot of effort to build programs that make up for these. },
abstract = {Voice conferencing has attracted interest as a useful and viable first real-time application on the Internet. This report describes NEVOT a network voice terminal meant to support multiple concurrent both two-party and multi-party conferences on top of a variety of transport protocols and using audio encodings offering from vocoder to multi-channel CD quality. As it is to be used as an experimental tool, it offers extensive configuration, trace and statistics options. The design is kept modular so that additional audio encodings, transport and real-time protocols as well as user interfaces can be added readily. In the first part, the report describes the X-based graphical user interface, the configuration and operation. The second part describes the individual components of NEVOT and compares alternate implementations. An appendix covers the installation of NEVOT.},
keywords = {Packet switching (Data transmission)},
acknowledgement = ack-nhfb
}
@article{Schulzrinne98a,
author = {Schulzrinne and Casner and Frederick and Jacobson},
title = {{RTP}: A Transport Protocol for Real-Time Applications},
journal = {Internet-Draft ietf-avt-rtp-new-01.txt (work inprogress)},
year = {1998},
month = {August},
annote = {This is the updated RTP specification in preparation of the protocol moving to a Draft Standard. For a description of RTP see Katarina's comments on RFC 1889.},
url = {ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-01.txt}
}
@article{Schulzrinne98b,
author = {Schulzrinne },
title = {{RTP} Profile for Audio and Video Conferences with Minimal Control},
journal = {Internet-Draft ietf-avt-profile-new-03.txt (work in progress)},
year = {1998},
month = {August},
annote = {Interesting to see the list of audio encodings they discuss. Also contains pointers to reference implementations. Also contains the following note on running RTP over TCP: RTP over TCP and Similar Byte Stream Protocols. Under special circumstances, it may be necessary to carry RTP in protocols offering a byte stream abstraction, such as TCP, possibly multiplexed with other data. If the application does not define its own method of delineating RTP and RTCP packets, it SHOULD prefix each packet with a two-octet length field. },
url = {ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-profile-new-03.txt}
}
@article{Shenker95,
author = {S. Shenker},
title = {Fundamental Design Issues for the Future {Internet}},
journal = {IEEE J. Selected Areas in Communications},
year = {1995},
month = {September},
bibdate = {Thursday, November 26, 1998 at 16:49:15 (NFT)},
submitter = {Katarina Asplund}
}
@article{Simpson93,
author = {W. Simpson},
title = {{IPng} (Next Generation) Mobility Considerations},
journal = {Request for Comments 1688},
year = {1993},
pages = {8},
month = {December},
annote = {This RFC addresses mobility in IPng, and the implications this has on the protocol. Issues like: addressing, routing, security, bandwith are discussed.},
url = {http://www.it.kth.se/docs/rfc/rfcs/rfc1550.txt}
}
@inproceedings{Sisalem98,
author = {Dorgham Sisalem and Henning Schulzrinne},
title = {The Loss-Delay Based Adjustment Algorithm: A {TCP}-Friendly Adaptation Scheme},
booktitle = {Proceedings of NOSSDAV},
year = {1998},
address = {Cambridge, UK. },
annote = {Suggests a Loss-Delay algorithm for adapting the output data rate of multimedia applications using RTCP info.},
url = {papers/Sisa9807_Loss_ps.gz},
submitter = {Johan Garcia}
}
@article{Smith94,
author = {Brian C. Smith},
title = {Cyclic-{UDP}: A priority-Driven Best-Effort Protocol},
journal = {Unpublished manuscript },
year = {1994},
month = {May},
annote = {The paper describes a protocol that uses application level ordering and a transport-level queue to provide different prority to different media units. The protocol also uses flow control to limit the outgoing bandwidth, and by the prioritizing mechanism it is always the most important packets that will be sent.},
url = {papers/cyclicudp.ps}
}
@article{Solomon98,
author = {J. D. Solomon},
title = {{Mobile IP} - The {Internet} Unplugged},
journal = {Prentice Hall},
year = {1998},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 22, 1998 at 16:25:03 (NFT)}
}
@article{Spanias94,
author = {Spanias, A.S.},
title = {Speech Coding: A Tutorial Review },
journal = {Proceedings of the IEEE},
year = {1994},
volume = {82},
number = {10},
pages = {1541--1582},
month = {Oct.},
annote = { A thorough (95 pgs, 300+ refs) expose of different voice coding algorithms. Sections 6 gives a good summarizing table and section 1 can also be skimmed through. Other sections rather detailed and somewhat mathematical.},
url = {papers/review.ps},
submitter = {Johan Garcia}
}
@article{Stalight98,
author = {{Starlight Networks, Inc}},
title = {StarLive Datasheet},
journal = {http://www.starlight.com\\/starlight/start.htm},
year = {1998},
month = {November},
bibdate = {Wednesday, November 04, 1998 at 15:21:27 (NFT)},
submitter = {Johan Garcia}
}
@article{Sterbenz90,
author = {J. P. G. Sterbenz and G. Parulkar},
title = {Axon: Network virtual storage design},
journal = {Computer Communication},
year = {1990},
volume = {20},
pages = {50-56},
month = {April},
submitter = {Karl-Johan Grinnemo},
bibdate = {Saturday, November 21, 1998 at 12:53:43 (NFT)}
}
@article{Sterbenz90a,
author = {J. P. G. Sterbenz and G. Parulkar},
title = {Axon: A high speed communication architecture for distributed applications},
journal = {Proceedings of INFOCOM '90, Washington D.C.},
year = {1990},
month = {June},
bibdate = {Saturday, November 21, 1998 at 12:56:07 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@book{Stevens94,
author = {Richard W. Stevens},
title = {{TCP}/{IP} Illustrated, Volume 1: The Protocols},
publisher = {Addison Wesley},
year = {1994},
address = {Reading},
descriptor = {TCP, TCP/IP},
isbn = {0-201-63346-9},
note = {ISBN 0-201-63346-9}
}
@misc{Stevens98,
title = {{TCP} Congestion Control},
author = {W. Stevens and M. Allman and V. Paxson},
month = aug,
year = 1998,
note = {Work in Progress},
annote = { This document discusses three distinct topics related to TCP Congestion Control. First it looks at TCP's four congestion control algorithms: \begin{itemize} \item slow start \item congestion avoidance \item fast retransmit \item fast recovery \end{itemize} In addition the document specifies how TCP should begin transmission after a relatively long idle period. Finally, various acknowledgment generation methods are discussed. Implementation considerations are give for each of the four congestion control algorithms. This includes a discussion of what is required by an implementation and what is merely a recommendation. The discussion on the restart of idle connections recommends that slow start should be used to avoid an inappropriate burst of traffic to be transmitted. Finally, when discussing acknowledgment mechanisms, the author recommends that the delayed ACK algorithm should be used. It also emphasizes that no TCP implementation is allowed to generate more than one ACK for every incoming segment. For implementations that implement the selective acknowledgment option guidelines are given for how these algorithms should behave, but implementation details are left up to the implementer. },
status = {INTERNET DRAFT}
}
@article{Stewart98,
author = {R. R. Stewart and Q. Xie},
title = {MULTI_NETWORK DATAGRAM TRANSMISSION PROTOCOL},
journal = {Internet-Draft draft-stewart-xie-mdtp-00.txt (work inprogress)},
year = {1998},
month = {August},
annote = {Protocol implementing reliable (and unreliable) transfer at the application layer. Suggested as an alternative to TCP. Claims to achieve major improvements over TCP in areas including redundant network support, fault management and configuration flexibility. Not aimed at real-time applications but supports message boundaries.},
url = {ftp://ftp.ietf.org/internet-drafts/draft-stewart-xie-mdtp-00.txt}
}
@article{Strayer89,
author = {W. T. Strayer and B. J. Dempsey and A. C. Weaver},
title = {Making {XTP} Responsive to Real-Time Needs},
journal = {Technical Report TR-89-18, Department of Computer Science},
year = {1989},
month = {December},
submitter = {Karl-Johan Grinnemo},
bibdate = {Saturday, November 21, 1998 at 12:49:01 (NFT)}
}
@article{Strayer92,
author = {Timothy W. Strayer and Bert J. Dempsey and Alfred C. Weaver},
title = {{XTP} -- {The} Xpress Transfer Protocol},
journal = {Addison-Wesley Publishing Company},
year = {1992},
abstract = {Chapter 1 Foundations of the Xpress Transfer Protocol Section 1.1 Why ANY New Protocol? Section 1.1.1 Changes Below Section 1.1.2 Changes Above Section 1.1.3 Changes Within Section 1.1.4 Repair or Replace? Section 1.2 Why THIS New Protocol Section 1.2.1 Functional Enhancements Section 1.1.2 Performance Enhancements Section 1.3 History of XTP Section 1.4 Organization of Book Chapter 2 Network Concepts and the XTP Architectural Model Section 2.1 Network Concepts Section 2.1.1 The OSI Reference Model Section 2.1.2 Data Transfer Models Section 2.1.3 Services and Service Access Section 2.1.4 Transport Layer Section 2.1.5 Network Layer Section 2.2 XTP Architecture Section 2.2.1 XTP Transfer Layer Architecture Section 2.2.2 XTP Data Communication Model Section 2.2.3 XTP Data Delivery Service Requirements Chapter 3 Influential Protocols Section 3.1 Conventional Transport Section 3.1.1 TCP Section 3.1.2 ISO Transport Protocol Section 3.1.3 Conventional Mechanisms Section 3.2 Delta-t Section 3.3 NETBLT Section 3.4 GAM-T-103 Military Real-Time Local Area Network Architecture Section 3.5 Versatile Message Transaction Protocol Section 3.6 Datakit and the Universal Receiver Protocol Section 3.7 Summary Chapter 4 Protocol Procedures Section 4.1 Introduction Section 4.1.1 XTP Packet Structure Overview Section 4.1.2 XTP Packet Formats Section 4.1.3 XTP protocol Procedures Section 4.1.4 Conventional Notations Section 4.2 Association Management Procedures Section 4.2.1 Establishing an Association Section 4.2.2 Maintaining an Association Section 4.2.3 Terminating an Association Section 4.2.4 Interesting Paradigms Section 4.2.5 Multicast Association Section 4.3 Path Management Procedures Section 4.3.1 Path Establishment Section 4.3.2 Path Maintenance Section 4.3.3 Path Release Section 4.4 Data Flow Procedures Section 4.5 Flow Control Procedures Section 4.6 Rate Control Procedures Section 4.7 Error Control Procedures Section 4.7.1 Checksums Section 4.7.2 Acknowledgements and Retransmissions Section 4.7.3 Timers Section 4.7.4 Synchronizing Handshake Section 4.7.5 Time-to-Live Section 4.7.6 Error Notification Chapter 5 Packet Structures Section 5.1 Introduction Section 5.1.1 Notation Convention Section 5.1.2 Segments Section 5.2 XTP Header Section 5.2.1 Route Field (route) Section 5.2.2 Time-to-Live Field (ttl) Section 5.2.3 Command Field (cmd) Section 5.2.4 Key Field (key) Section 5.2.5 Synchronize Field (sync) Section 5.2.6 Sequence Number Field (seq) Section 5.2.7 Delivered Sequence Number Field (dseq) Section 5.2.8 Sort Field (sort) Section 5.2.9 Data Length Field (dlen) Section 5.2.10 Header Checksum Field (hcheck) Section 5.3 XTP Trailer Section 5.4 Control Segment Section 5.4.1 Rate Control Fields (rate and burst) Section 5.4.2 Synchronize Echo Field (echo) Section 5.4.3 Time Synchronization Fields (time and techo) Section 5.4.4 Key Exchange Field (xkey) Section 5.4.5 Route Exchange Field (xroute) Section 5.4.6 Allocation Field (alloc) Section 5.4.7 Received Sequence Number Field (rseq) Section 5.4.8 Selective Retransmission Fields (nspan and spans) Section 5.4.9 The Reserved Fields (rsvd) Section 5.5 Information Segment Section 5.5.1 Data Segment Section 5.5.2 Address Segment Section 5.5.3 Data and Address Segments Section 5.5.4 Management Segment Chapter 6 Packet Formats Section 6.1 FIRST Packet Section 6.2 DATA Packet Section 6.3 ...},
isbn = {0-201-56351-7},
keywords = {book network protocol design network protocol analysis XTP multicast}
}
@article{Strayer95,
author = {W.T. Strayer},
title = {{Xpress Transport Protocol} 4.0 Specification},
journal = {XTP Forum Inc.},
year = {1995},
submitter = {Katarina Asplund},
bibdate = {Friday, November 27, 1998 at 09:58:07 (NFT)}
}
@inproceedings{Sudan97,
author = {Madhu Sudan },
title = {Algorithmic Issues in Coding Theory},
booktitle = {Foundations of Software Technology and Theoretical Computer Science 17th Conference},
year = {1997},
editor = {S. Ramesh and G. Sivakumar },
publisher = {LNCS 1346},
address = {Kharagpur, India},
month = {Dec. 18-20},
annote = {Contains a mathematical discussion on FEC},
url = {papers/13460184.pdf}
}
@article{T12096,
author = {ITU-T},
title = {Recommendation {T.120} - Data protocols for multimedia conferencing},
journal = {Geneva, Switzerland},
year = {1996},
month = {July},
submitter = {Johan Garcia},
bibdate = {Friday, November 06, 1998 at 09:38:53 (NFT)}
}
@article{Tan98,
author = {W. Tan and A. Zakhor},
title = {Error Resilient Packet Video for the {Internet}},
journal = {Submitted to ICIP 98},
year = {1998},
annote = {Describes a compression method based on subband divison where every packet is equally important. The packets are transmitted on top of a TCP-friendly transport protocol with rate based congestion control. Experiments and simulation show good performance. Short well-written paper worth reading.},
url = {papers/icip98.ps}
}
@book{Tanenbaum96,
author = {Andrew S. Tanenbaum},
title = {Computer Networks},
publisher = {Prentice-Hall International, Inc.},
year = {1996},
note = {ISBN: 0-13-394248-1},
date = {03/09/96},
star = {5},
comment = {Topics new to the Third Edition include The TCP/IP reference model, wireless communication, Broadband ISDN, SONET, Fast Ethernet, OSPF, Mobile IP, IPv6, network performance, the domain name system, the World Wide Web, Java, and Multimedia networking. In addition, the material on the OSI protocols has been removed and replaced by detailed discussion of the TCP/IP and Internet protocols. Futhermore, all the example protocols in the book are now written in C, rather than in Pascal.},
keywords = {Communications: hardware through applications}
}
@article{Tang93,
author = {Tang, J.C. and Isaacs, E.A.},
title = {Why Do Users Like Video? {Studies} of Multimedia-Supported Collaboration},
journal = {CSCS: An International Journal},
year = {1993},
volume = {1},
number = {3},
pages = {163-196},
annote = {Contains summary of three studies. One study surveyed users' opinions of a video conference room system. A second study analyzed videotape records of a work group when meeting face-to-face, video conferencing, and phone conferencing. The analysis indicated that the noticeable audio delay made it difficult for the participants to manage turn taking and to coordinate their gazes, which in turn impaired their ability to handle conflict and to engage in humor. In the third study, a distributed team was observed under three conditions: using their existing collaboration tools, adding a desktop conferencing prototype (audio, video, and shared drawing tool), and subtracting the video capability from the prototype. During one study, the audio latency was measured to vary between 0.32-0.44 seconds (depending on processing and networking loads). This slight improvement over the 0.57 second latency in the video conference rooms was enough to noticeably affect the level of interaction that the collaborators could accomplish. },
url = {http://earthlight.com/ellen/papers/dvc-cscwj.html},
bibdate = {Wednesday, October 14, 1998 at 11:51:56 (DFT)},
submitter = {Johan Garcia}
}
@article{Teraoka93,
author = {F. Teraoka and M. Tokoro},
title = {Host Migration Transparency in {IP} Networks},
journal = {Computer Communication Review},
year = {1993},
pages = {45-46},
month = {January},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 22, 1998 at 15:56:45 (NFT)}
}
@article{Towsley93,
author = {Don Towsley},
title = {Providing quality of service in packet switched networks},
journal = {Performance Evaluation of Computer and Communication Systems, Springer-Verlag},
year = {1993},
pages = {560-586},
submitter = {Katarina Asplund},
bibdate = {Thursday, November 26, 1998 at 16:41:38 (NFT)}
}
@techreport{Turletti97,
author = {Thierry Turletti and Sacha Fosse Parisis and Jean-Chrysostome Bolot},
title = {Experiments with a Layered Transmission Scheme over the {Internet}},
year = {1997},
month = {Nov.},
annote = {Discusses a TCP-friendly multicast rate control mechanism for use with layered coding (audio coding is used as example in paper)},
url = {papers/rr-3296.ps},
howpublished = {INRIA Report No 3296},
submitter = {Johan Garcia}
}
@inproceedings{Villemur98,
author = {Thierry Villemur and Veronique Baudin and Stephane Owezarski and Michel Diaz},
title = {An Integrated Platform for Cooperative Teleteaching},
booktitle = {Interactive Distributed Multimedia Systems and Telecommunication Services},
year = 1998,
editor = {Thomas Plagemann and Vera Goebel},
pages = {59-70},
publisher = {Springer-Verlag},
month = {September},
annote = {my stuff here},
series = {Lecture Notes in Computer Science},
volume = 1483
}
@article{Vocaltec98,
author = {VocalTec Communications Ltd.},
title = {{Internet Phone 5}},
journal = {http://www.vocaltec.com/products/iphone5/index.html},
year = {1998},
month = {November},
bibdate = {Wednesday, November 04, 1998 at 15:26:29 (NFT)},
submitter = {Johan Garcia}
}
@article{WAP98,
author = {{WAP Forum Ltd.}},
title = {{Wireless Application Protocol} - Architecture Specification},
journal = {http://www.wapforum.org},
year = {1998},
month = {April},
annote = {Gives an architectual overview of WAP, and elaborates on the, so called, "WAP Model".},
url = {http://www.wapforum.org/docs/technical.htm}
}
@article{Wada93,
author = {H. Wada and T. Yozawa and T. Ohnishi and Y. Tanaka},
title = {Mobile Computing Environment Based on {Internet} Packet Forwarding},
journal = {Proceedings of the Winter USENIX Conference},
year = {1993},
pages = {503-517},
month = {January},
bibdate = {Sunday, November 22, 1998 at 15:59:40 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@misc{Wapforum98,
title = {{WAP} Specifications for {WAP} version 1.0 ({Wireless Application Protocol})},
author = {{WAP Forum}},
note = {Available at \url{http://www.wapforum.org}},
annote = {The WAP Forum aims to create a global wireless protocol specification that works across differing network technologies. At the home page referenced, you find all publicly available specifications concerning WAP version 1. To us, the specification "Wirelesss Application Protocol - Wireless Transaction Protocol Specification" maybe of interest, even though the transaction protocol is defined to provide the services necessary for interactive "browsing" with no concern about real-time applications. },
url = {http://www.wapforum.org/docs/index.htm },
submitter = {Karl-Johan Grinnemo}
}
@article{Weaver,
author = {Alfred C. Weaver},
title = {{Xpress Tranport Protocol}, Version 4.0 },
journal = {http://www.cs.virginia.edu/~netlab/xtp_stuff/xtp4_tutorial.ps},
annote = {A 15 page overview of XTP. Provides a basic knowledge of the features and fuctionality of the protocol that everyone should have.},
url = {papers/xtp4_tutorial.ps}
}
@article{Whitepine98,
author = {{White Pine Software, Inc}},
title = {{CU-SeeMe} Product Information},
journal = {http://www.wpine.com/Products/CU-SeeMe/index.html},
year = {1998},
month = {October},
annote = {Referred link to CU-SeeME},
url = {http://www.wpine.com/Products/CU-SeeMe/index.html},
bibdate = {Wednesday, October 14, 1998 at 13:00:09 (DFT)},
submitter = {Johan Garcia}
}
@article{Wolf96,
author = {Wayne Wolf and Yiqing Liang and Michael Kozuch and Heathery Yu and Michael Phillips and Marcel Weekes and Andrew Debruyne},
title = {A digital video library on the {World Wide Web}},
journal = {Proceedings ACM Multimedia 96},
year = {1996},
pages = {433-434},
annote = {High-level description of application, no need to read. },
url = {papers/p433-wolf.pdf}
}
@inproceedings{Yavatkar94,
author = {R. Yavatkar and N. Bhagwat},
title = {Improving End-to-End Performance of {TCP} over Mobile Internetworks},
booktitle = {Proceedings of the Workshop on Mobile Computing Systems and Applications (WMCSA)},
address={Santa Cruz, California, USA},
year = {1994},
month = dec,
bibdate = {Sunday, November 22, 1998 at 16:33:58 (NFT)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Yun96,
author = {Louis C. Yun and David G. Messerschmitt},
title = {Digital Video in a Fading Interference Wireless Environment},
booktitle = {Proceedings of Intl. Conference on Acoustics, Speech and Signal Processing},
year = {1996},
address = {Atlanta, GA},
month = {may},
annote = {Advocates layered coding, also as a basis for joint source/channel coding. Has a high BER requirement for MPEG.},
url = {papers/Icassp.pdf},
submitter = {Johan Garcia}
}
@article{Zenel95,
author = {Bruce Zenel and Dan Duchamp},
title = {Intelligent Communication Filtering for Limited Bandwidth Environments},
journal = {Hot-OS V},
year = {1995},
month = {May},
annote = {In this paper, Bruce Zenel et al. describes how a proxy interposed between a fixed server and a mobile client can mitigate the network limitations imposed by the wireless channel. You could see this paper as an introductory text to the paper "General Purpose Proxies: Solved and Unsolved Problems".},
url = {http://www.mcl.cs.columbia.edu/~baz/publications.html}
}
@article{Zenel97,
author = {Bruce Zenel and Dan Duchamp},
title = {General Purpose Proxies: Solved and Unsolved problems},
journal = {Hot-OS, VI},
year = {1997},
month = {May},
annote = {In his paper, Bruce describes how a proxy between a fixed and mobile host can help alleviate the network limitations imposed by the wireless communication channel. More specifically, the proxy runs special programs, so called filters, that may: - partition the fixed and wireless communication channels. Using an optimized protocol between the proxy and the mobile host, while continue talking TCP to the fixed host. - heuristically drop data. - compress data. - impose a delay on data. },
url = {http://www.mcl.cs.columbia.edu/~baz/publications.html}
}
@inproceedings{Zhang86,
author = {L Zhang},
title = {Why {TCP} Timers Don't Work Well},
booktitle = {Proc. Comm. Arch. \& Protocols Symp., SIGCOMM86, Vermont, pp. 397-405, ACM},
year = 1986,
month = aug ,
contents = {congestion}
}
@article{Zhang93,
author = {L. Zhang and S. Deering and D. Estrin and S. Shenker and D. Zappala},
title = {{RSVP}: A New Resource ReSerVation Protocol},
journal = {IEEE Network},
year = {1993},
volume = {5},
pages = {8-18},
month = {September},
submitter = {Katarina Asplund},
bibdate = {Thursday, November 26, 1998 at 17:21:42 (NFT)}
}
@inproceedings{Zhao96,
author = {J. Zhao and Y. Shimazu and K. Ohta and R. Hayasaka and Y. Matsushita},
title = {A {JPEG} Codec Adaptive to Region Importance},
booktitle = {Proceedings of the Fourth {ACM} Multimedia Conference ({MULTIMEDIA}'96)},
year = {1996},
pages = {209--218},
publisher = {ACM Press},
address = {New York, NY, USA},
month = nov,
annote = {Describes a coding technique where you "spend more bits" on the important part of a picture. For instance a face is coded more accurately than the background of the picture. Uses AI-based techniques to determine which portions of a picture is most important.},
url = {papers/p209-zhao.pdf},
isbn = {0-201-92140-X}
}
@article{Zhao97,
author = {Xinhua Zhao and Mary G. Baker},
title = {Flexible Connectivity Management for Mobile Hosts},
journal = {Computer System Laboratory, Stanford University},
year = {1997},
pages = {19},
month = {September},
annote = {One major problem concerning mobility in an IP-network is the addressing scheme to use. In this paper Xinhua Zhao et al. proposes two ways to solve this problem. First, a "foreign agent" could be used that registered the mobile host with the "home agent" by sending its IP address. The second way is for the mobile host to receive a temporary IP address to use while it visits a foreign network. The mobile host registers this address with its home agent. Zhao also elaborates on various mechanisms for flexible connectivity management, i.e. to enable mobile hosts to choose different delivery methods based on the network at hand.},
url = {http://mosquitonet.Stanford.EDU/mosquitonet.html#Publications}
}
@article{jfif98,
author = {Eric Hamilton},
title = {The {JPEG} File Interchange Format},
journal = {C-Cube Microsystems Inc, http://http.ijg.org/files/jfif.ps.gz.},
year = {1998},
annote = {Reference from RFC 2035},
bibdate = {Thursday, November 05, 1998 at 15:39:33 (NFT)},
submitter = {Johan Garcia},
key = {JPEG}
}
@misc{vat,
title = {LBNL Audio Conferencing Tool (vat)},
author = {{Network Research Group, Lawrence Berkeley National Laboratory}},
howpublished = {URL: http://www-nrg.ee.lbl.gov/vat/},
month = {November},
year = 1998
}
@article{vdonet98,
author = {{VDOnet Corp.}},
title = {{VDOLive} Documentation},
journal = {http://www.clubvdo.net\\/store/Products/Lib/vdolivewpapers.asp},
year = {1998},
month = {November},
bibdate = {Wednesday, November 04, 1998 at 15:02:17 (NFT)},
submitter = {Johan Garcia}
}
@article{vdonet98a,
author = {{VDOnet Corp.}},
title = {{VDOPhone} Documentation},
journal = {http://www.clubvdo.net\\/store/Products/Lib/phoneds.asp},
year = {1998},
month = {November},
bibdate = {Wednesday, November 04, 1998 at 15:03:35 (NFT)},
submitter = {Johan Garcia}
}
@article{vivo98,
author = {{VIVO Software Inc.}},
title = {{VivoActive VideoNow}},
journal = {http://www.vivo.com\\/products/products.html},
year = {1998},
month = {November},
submitter = {Johan Garcia},
bibdate = {Wednesday, November 04, 1998 at 14:53:56 (NFT)}
}
@article{vosaic98,
author = {{VOSAIC LLC}},
title = {{VOSAIC MediaServer}},
journal = {http://www.vosaic.com\\/products/servers/},
year = {1998},
month = {November},
submitter = {Johan Garcia},
bibdate = {Wednesday, November 04, 1998 at 14:49:16 (NFT)}
}
@misc{wb,
title = {wb - {LBNL} Whiteboard Tool},
author = {{Network Research Group, Lawrence Berkeley National Laboratory}},
howpublished = {URL: http://www-nrg.ee.lbl.gov/wb/},
month = {November},
year = 1998
}
@article{xing98,
author = {{Xing Technology Corp.}},
title = {{StreamWorks} Server},
journal = {http://www.xingtech.com\\/products/swserver/},
year = {1998},
month = {November},
bibdate = {Wednesday, November 04, 1998 at 15:11:17 (NFT)},
submitter = {Johan Garcia}
}
@article{Bakre95a,
author = {A. Bakre and R. Badrinath},
title = {Handoff and System Support for {Indirect TCP/IP}},
journal = {Proceedings of the 2nd USENIX Symposium on Mobile and Location-Independent Computing},
year = {1995},
month = {April},
submitter = {Karl-Johan Grinnemo}
}
@misc{Jpeg2000,
title = {{JPEG2000} requirements and profiles version 4.0},
author = {{JPEG2000 Requirements ad hoc group}},
month = {November},
year = 1998,
note = {Draft in progress}
}
@inproceedings{Brunstrom95-2,
author = {Anna Brunstrom and Phil Kearns},
title = {A User Level Implementation of Flush Channels},
booktitle = {Proceedings of the Seventh {IASTED/ISMM} International Conference on Parallel and Distributed Computing and Systems},
year = 1995 ,
pages = {125-128},
month = {October}
}
@article{Ahuja90-1,
author = {Mohan Ahuja},
title = {Flush Primitives for Asynchronous Distributed Systems},
journal = {Information Processing Letters},
year = 1990 ,
volume = 34,
pages = {5-12},
annote = {Have not read it}
}
@article{Ahuja90-2,
author = {Mohan Ahuja and Ajay D. Kshemkalyani and Timothy Carlson},
title = {A Basic Unit of Computation in Distributed Systems},
journal = {Proc. {ICDCS-10}},
year = 1990 ,
pages = {12-19},
annote = {The atoms and molecules paper}
}
@incollection{Ahuja91-1,
author = {Mohan Ahuja and Kannan Varadhan and Amitabh Sinha},
title = {Flush Message Passing in Communicating Sequential Processes},
booktitle = {Parallel Architecture},
publisher = {IEEE Computer Society Press},
year = 1991 ,
editor = {N. Rishe, S. Navathe and D. Tal},
pages = {31-47},
annote = {Have not read it}
}
@article{Ahuja91-2,
author = {Mohan Ahuja},
title = {An implementation of F-channels, a preferable alternative to {FIFO} channels},
journal = {Proc. 11th Int. Conf. Distributed Computing Systems},
year = 1991 ,
pages = {180-187},
annote = {Have not read it}
}
@techreport{Ahuja91-3,
author = {Mohan Ahuja},
title = {Hierarchy of Communication Speeds for Designing Concurrent Systems},
institution = {Ohio State University},
year = 1991 ,
number = {OSU-CISRC-1/91-TR1},
month = jan,
annote = {Have not read it}
}
@article{Camp92,
author = {Phil Kearns and Tracy Camp and Mohan Ahuja},
title = {An Implementation of Flush Channels Based on a Verification Methodology},
journal = {Proc. 12th Int. Conf. Distributed Computing Systems},
year = 1992,
pages = {336-343},
annote = {Have not read it}
}
@inproceedings{Amer97,
author = {Paul Amer and Philip Conrad and Edward Golden and Sami Iren and Armando Caro },
title = {Partially-ordered, partially-reliable transport service for multimedia applications },
booktitle = {Proc. Advanced Telecommunications/Information Distribution Research Program Annual Conference},
year = 1997 ,
pages = {215-220 },
address = {College Park, MD},
month = {January}
}
@article{johansson96,
author = {MAGNUS JOHANSSON },
title = {Early Analog Computers in Sweden - With Examples From Chalmers University of Technology and the Swedish Aerospace Industry},
journal = {IEEE ANNALS OF THE HISTORY OF COMPUTING},
year = {1996},
volume = {18},
number = {4},
pages = {27-33},
month = {October},
annote = {This paper gives a short overview of early analog computing in Sweden in the 1940s and 1950s. },
submitter = {Johan Garcia}
}
@inproceedings{Ng97,
author = {{F. M.~} Ng and J. Kova\v{c}evi\'{c}},
title = {Nonredundant image representations},
booktitle = icip,
year = {1997},
address = {Santa Barbara, CA},
month = {October},
submitter = {Johan Garcia}
}
@article{Rizzo97,
author = {L. Rizzo and L. Vicisano},
title = {A Reliable Multicast data Distribution Protocol based on software {FEC} techniques},
journal = {Proceedings of the Fourth IEEE HPCS '97 Workshop, Chalkidiki, Greece},
year = {1997},
month = jun,
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Pfeifer98,
author = {Tom Pfeifer and Thomas Magedanz and Stephan Hubener},
title = {Mobile Guide - Location-Aware Applications from the Lab to the Market},
booktitle = {Proceedings of IDMS},
year = {1998},
editor = {Thomas Plagemann and Vera Goebel},
publisher = {Springer},
address = {Oslo, Norway},
month = {Sept.},
annote = {Describes a mobile, navigating information retrieval platform},
submitter = {Johan Garcia}
}
@article{Lundgren98,
author = {Fredrik Lundgren },
title = {Mobil Internet- nu med Turbo },
journal = {Telia customer information web, http://www.dof.se/utblick/arkiv/artikel/artikel3/artikel.htm},
annote = {Länk till Mobile Internet Turbo info.},
submitter = {Johan Garcia}
}
@article{Tan98a,
author = {W. Tan and A. Zakhor},
title = {{Internet} Video using Error Resilient Scalable Compression and Cooperative Transport Protocol},
journal = {Proceedings of the IEEE International Conference on Image Processing (ICIP), Chicago, IL},
year = {1998},
volume = {3},
pages = {458-462},
month = {October},
annote = {Presents an interleved wavelet packetizing strategy coupled with a TCP-friendly unreliable transmission protocol.},
submitter = {Johan Garcia}
}
@inproceedings{Cosman98,
author = { P. Cosman and J. Rogers and P. G. Sherwood and K. Zeger},
title = {Image Transmission over Channels with Bit Errors and Packet Erasures},
booktitle = {Proceedings 32nd Asilomar Conference on Signals, Systems, and Computers, Monterey, California},
year = {1998},
month = {November},
annote = {Proposes a hybrid FEC/packetized wavelet joint source/channel image transmission scheme suitable both for channels with bit error bursts typical of mobile commo, as well as packet loss. Nice loss distribution as well.},
submitter = {Johan Garcia}
}
@article{Dabbous95,
author = {W. Dabbous and C. Diot.},
title = {High Performance Protocol Architecture},
journal = {Proceedings of PCN '95. Istanbul.},
year = {1995},
month = {October},
submitter = {Johan Garcia}
}
@article{Fox96,
author = {Armando Fox and Eric A. Brewer},
title = {Reducing {WWW} Latency and Bandwidth Requirements by Real-Time Distillation},
journal = {Proc. Fifth International World Wide Web Conference, Paris, France},
year = {1996},
month = {May},
url = {http://www5conf.inria.fr/fich_html/papers/P48/Overview.html},
submitter = {Johan Garcia}
}
@article{Chin98,
author = {K. V. Chin and S. C. Hui and S. Foo},
title = {Enhancing the quality of {Internet} voice communication for {Internet} telephony systems},
journal = {Journal of Network and Computer Applications},
year = {1998},
volume = {21},
number = {3},
pages = {203-218},
month = {July},
annote = {Internet real time voice communication involves transmitting digitized voice signals which can be lost in the packet-switched network environment of TCP/IP, thereby causing intermittent voice losses and quality degradations. Various methods such as silence substitution, waveform substitution, sample interpolation, Xor mechanism, embedded speech coding, and the combined rate and control mechanism have been proposed to enhance voice delivery and to minimize the quality impact caused by these losses. This paper proposes a quality-based dynamic voice recovery mechanism that combines network transmission control and voice recovery to deliver voice signals with optimal intelligibility and quality. This is accomplished by considering the subjective rating of different codecs that are used in the coding and transmission of digital audio and network packet loss conditions. The dynamic mechanism results in voice delivery that, at minimum, satisfies voice intelligibility while tolerating moderate packet loss caused by network congestion. This mechanism has been successfully incorporated into the Internet Telephone Software System developed at the School of Applied Science, Nanyang Technological University. Johan - Describes a Internet telephony system based on RTP/UDP transport and network load adaption, much like Bolot.},
url = {papers/chin98.pdf},
submitter = {Anna Brunstr\"{o}m}
}
@article{Parr98,
author = {Gerard Parr and Kevin Curran},
title = {Optimal multimedia transport on the {Internet}},
journal = {Journal of Network and Computer Applications},
year = {1998},
volume = {21},
number = {2},
pages = {149-161},
month = {April},
annote = {Not as interesting as the title implies. Describes a quality of service framework where applications only pay for services they need: programmers can request qualities of service such as reliable multicast, virtual synchrony, encrypted communication and a protocol composition framework that extends to incorporate yet unsupported communication protocols and qualities of service. The paper focuses on the real time wide area network dissemination architecture protocol (RWANDA) which overcomes synchronous limitations by providing an asynchronous group communication model where applications only pay for the required quality of service (QoS) such as multicast, virtual synchrony and encrypted communication.},
url = {papers/rwanda.pdf},
submitter = {Anna Brunstr\"{o}m}
}
@article{Lu98,
author = {Guojun Lu and Chester Kang},
title = {An efficient communication scheme for media on-demand services with hard {QoS} guarantees},
journal = {Journal of Network and Computer Applications},
year = {1998},
volume = {21},
number = {1},
pages = {1-15},
month = {January},
annote = {The critical issue of multimedia communications is to provide quality of service (QoS) guarantees while system resources are efficiently used. The system utilization is usually low when hard QoS guarantees are required, due to "burstiness" of multimedia traffic. We propose a scheme that provides hard QoS guarantees for media on-demand applications while fully utilizing system resources. The basic idea of the scheme is to convert variable bit-rate streams into constant bit-rate streams for transmission. The constant bit-rate is equal to the average bit-rate of the stream. This arrangement not only fully utilizes the system resources, but also simplifies the server and network design. The scheme requires extra buffering at the receiver (client). We show how to determine the buffering delay and end-to-end delay. Our experimental results show that using our scheme, the end-to-end delay is acceptable to most media on-demand applications, and in the case of heavily loaded multi-hop networks, may be lower than that experienced when no traffic smoothing is carried out.},
url = {papers/lu98.pdf},
submitter = {Anna Brunstr\"{o}m}
}
@article{Larzon99,
author = { L. Larzon and M. Degermark and S. Pink},
title = {The {UDP} Lite Protocol},
journal = {Internet-Draft, draft-larzon-udplite-00.txt (work in progress)},
year = {1999},
month = {March},
annote = {A version of UDP that provides a partial checksum intended to cover the header data. Checksumming the payload can be skipped for applications that can handle damaged data (ie. multimedia with robust decoders).},
url = {http://www.ietf.org/internet-drafts/draft-larzon-udplite-00.txt},
submitter = {Anna Brunstr\"{o}m}
}
@article{Kinoshita95,
author = {Kinoshita and Yamamuro},
title = {Image Quality with Reiterative {JPEG} Compression},
journal = {Journal of Imaging Science and Technology},
year = {1995},
volume = {39},
number = {4},
pages = {306-312},
month = {July},
annote = {Claim that (1) the iterative factor of the repetitive JPEG operation had no influence on image quality, and (2) the first compression determined base image quality. },
submitter = {Johan Garcia}
}
@article{Bolot93,
author = {J-C. Bolot},
title = {End-to-end packet delay and loss behavior in the {Internet}},
journal = {Proc. ACM Sigcomm, San Fransisco, CA },
year = {1993},
pages = {289-298},
month = {September},
submitter = {Johan Garcia}
}
@article{Talukdar99,
author = {A. K. Talukdar, B. R. Badrinath, A. Acharya},
title = {Integrated services packet networks with mobile hosts: Architecture and performance},
journal = {Wireless Networks},
year = {1999},
volume = {5},
number = {2},
pages = {111-124},
month = {March},
annote = {This paper proposes a service model for mobile hosts that can support adaptive applications which can withstand service degradation, as well as applications which require mobility independent service guarantees.},
url = {http://www.cs.kau.se/cs/prtp/papers/talukdar99.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, April 21, 1999 at 11:27:03 (MEST)}
}
@article{Degermark99,
author = {M. Degermark and M. Engan and B. Nordgren and S. Pink},
title = {Low-loss {TCP/IP} header compression for wireless networks},
journal = {Wireless Networks},
year = {1997},
volume = {3},
number = {5},
pages = {375-387},
month = Oct,
annote = {In this paper, two schemes for compresing UDP and TCP headers are presented. A new mechanism, compression slow-start, is introduced. This mechanism allows quick installation of high compression rates.},
url = {http://www.cs.kau.se/cs/prtp/papers/degermark99.pdf},
bibdate = {Wednesday, April 21, 1999 at 14:11:38 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Durst97,
author = {R. C. Durst, G. J. Miller},
title = {{TCP} Extensions for space communications},
journal = {Wireless Networks},
year = {1997},
volume = {3},
number = {5},
pages = {389-403},
month = {October},
annote = {This paper proposes som extensions to TCP that addresses problems related to wireless links in general, and satellite links in particular. Simulation and live experiments conducted on SPCS-TP, the name of the protocol, indicate significant improvements in throughput over unmodified TCP on error-prone links.},
url = {http://www.cs.kau.se/cs/prtp/papers/durst97.pdf},
bibdate = {Wednesday, April 21, 1999 at 15:04:50 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Bhagwat97,
author = {P. Bhagwat, P. Bhattacharya, A. Krishna, and S. Tripathi},
title = {Using channel dependent packet scheduling to improve {TCP} throughput over wireless LANs},
journal = {Wireless Networks},
year = {1997},
volume = {3},
number = {1},
pages = {91-102},
month = {March},
annote = {This paper discusses the impact of burst errors and error recovery in the MAC layer for TCP.},
url = {http://www.cs.kau.se/cs/prtp/papers/bhagwat97.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, April 21, 1999 at 15:22:58 (MEST)}
}
@article{Alkhatib97,
author = {H. S. Alkhatib, C. Bailey, M. Gerla, J. McCrae},
title = {Wireless Data Networks: Reaching the Extra Mile},
journal = {IEEE Computer},
year = {1997},
volume = {30},
number = {12},
pages = {59-62},
month = {December},
annote = {This article surveys contemporary link layer technologies for wireless networks. The authors also discusses the problems that have to be solved in order to improve coverage and level of service for these techniques.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, April 21, 1999 at 16:23:44 (MEST)}
}
@article{ns2,
author = {S. McCanne and S. Floyd},
title = {{VINT Network Simulator} - ns (version 2)},
journal = {http://isi.edu/nsnam/ns/},
year = {1999},
month = {April},
url = {http://isi.edu/nsnam/ns/},
}
@article{Etsi98,
author = {{ETSI}},
title = {{Radio Link Protocol (RLP)} for data and telematic services. {GSM} 04.22},
journal = {{DRAFT pr ETS 300 946}},
year = {1998},
month = {November},
annote = {Contains the description of RLP - a HDLC derivative used for data link control over GSM channels. RLP frames are 240 bits (9.6kbit coding) or 576 bits (14.4 coding) and in an example it performs up to 6 retransmssions per lost frame. Retransmissions can be either Go-bach N or one-frame selective (optional). V.42bis data compression can also be applied. },
url = {papers/etsi_rlp.zip},
submitter = {Johan Garcia},
bibdate = {Tuesday, April 27, 1999 at 09:14:51 (MEST)}
}
@article{Koenen99,
author = {Rob Koenen},
title = {MPEG-4 Multimedia for our time},
journal = {IEEE Spectrum},
year = {1999},
volume = {36},
number = {2},
pages = {26--33},
month = {February},
annote = {Provides a good, non-technical overview of the MPEG-4 Standard.},
bibdate = {Tuesday, April 27, 1999 at 09:21:21 (MEST)},
submitter = {Johan Garcia}
}
@article{Jacobs98a,
author = {Stephen Jacobs and Huayan Wang and Alexandros Eleftheriadis and Mischa Schwartz},
title = {A Simple Analytical Model for Wireless {TCP} Utilization},
journal = {submitted to SIGCOMM, January 1998.},
annote = {Seems to make somewhat strange assumptions about the wireless channel behaiviour. Abstract: TCP performance over wireless networks can be poor because TCP assumes losses occur in the network only when there is congestion. However, wireless networks often drop packets due to random errors on the wireless link. TCP performance can be severely degraded when packets with errors are mistaken for congestion. We present a very simple, tractable, and closed-form analytical model for determining the performance of TCP over hybrid wired/wireless networks. We compare our model with previous work in this area from an analytical perspective. We also compare our model with several simulations. The results show that our model matches both the previous work and the simulations very well. We also provide a simple heuristic for determining when wireless errors will severely degrade TCP performance.},
url = {papers/Jacobs98_Wireless_TCP_analytic.ps.gz},
submitter = {Johan Garcia},
bibdate = {Tuesday, April 27, 1999 at 11:51:43 (MEST)}
}
@article{Jacobs98b,
author = {Stephen Jacobs and Alexandros Eleftheriadis},
title = {Streaming Video using Dynamic Rate Shaping and {TCP} Flow Control},
journal = {Journal of Visual Communication and Image Representation},
year = {1998},
volume = {9},
number = {3},
pages = {211-222},
month = {September},
annote = {Describes a partially reliable protocol using if-time selective retransmissions and a TCP-like congestion avoidance mechanism over UPD.
Abstract:We present a new technique for streaming real time video on today's Internet, based on dynamic rate shaping and TCP congestion control. Dynamic rate shaping is a signal processing technique that adapts the rate of compressed video (MPEG-1, MPEG-2, H.26x) to dynamically varying bandwidth constraints. This provides an interface (or filter) between the source and the network, with which the encoder's output (either live or stored) can be perfectly matched to the network's available bandwidth. We couple this adaptation capability with the use of a new semi-reliable protocol that uses the TCP congestion window to pace the delivery of data into the network, but without using other TCP algorithms that are poorly suited to real time media. Use of TCP congestion control ensures that the protocol competes fairly with all other TCP data, and that it optimally shares the available bandwidth. It also avoids the latency problems commonly associated with TCP. In addition, we describe a real application that uses this approach to stream MPEG video on the Internet. We present several experiments, performed in both a controlled environment and the wide area Internet, that were used to evaluate the effectiveness and fairness of the scheme. The results show that the proposed solution achieves superior video quality while at the same time providing fairness by sharing bandwidth equally with other non-real-time connections.},
url = {papers/Jacobs98_Stream_Video_TCP_Cong.ps.gz},
bibdate = {Tuesday, April 27, 1999 at 12:06:42 (MEST)},
submitter = {Johan Garcia}
}
@article{Calveras98,
author = {A. Calveras and J. Paradells},
title = { "TCP/IP Over Wireless Links: Performance Evaluation". },
journal = {48th Annual Vehicular Technology Conference VTC'98, Ottawa (Ontario), Canada},
year = {1998},
month = {May},
annote = {Poorly written paper that reports on experiments that show that TCPs throughput is almost cut in half by an uncompensated BER of 1/15000 for a MTU of 296 bytes. (This maps to approximately 15% packet loss, depending on their error distribution) },
url = {papers/Calveras98_TCP_wireless_perf_eval.ps.gz},
submitter = {Johan Garcia},
bibdate = {Tuesday, April 27, 1999 at 14:10:12 (MEST)}
}
@article{Ott99,
author = {J\"{o}rg Ott and Stephan Wenger and Gerd Knorr},
title = {Application of H.263+ Video Coding Modes in Lossy Packet Network Environments},
journal = {Journal of Visual Communication and Image Representation},
year = {1999},
volume = {10},
number = {1},
pages = {12-38},
month = {March},
annote = {Provides some classification of the problem area and then discusses H.263 and RTP/RTCP adn how they can be adapted for lossy nets. Section 8.1 provide figures from packet loss experiments: avg 2.5% (max 22%) for intra-german comm, avg 4.5% (max 28%) for berlin-london, and avg 10% (max 30%) for berlin-boston. All of the above for 8-120 kbit/s rates with no rate correlation of losses. Larger packets which caused IP-level fragmentation caused considerably higher values => relevant for TCP-packetization. },
url = {papers/Ott99_H263_lossy_networks.pdf},
bibdate = {Wednesday, April 28, 1999 at 09:11:09 (MEST)},
submitter = {Johan Garcia}
}
@article{Eckert98,
author = {Michael P. Eckert and Andrew P. Bradley},
title = {Perceptual quality metrics applied to still image compression },
journal = {Signal Processing (Elsevier)},
year = {1998},
volume = {70},
number = {3},
pages = {177-200},
month = {November},
annote = {Provides a thorough overview of the area. Suggests that simpler, non-perceptual, metrics such as MSE and PSNR can be used for examining differences between compression degrees using the same algorithm (our case). Perceptual metrics should be used when comparing different compression algorithms. Many references.},
url = {papers/Eckert98_Quality_metrics_images.pdf},
submitter = {Johan Garcia},
bibdate = {Wednesday, April 28, 1999 at 09:32:13 (MEST)}
}
@article{Kim98,
author = {Tae Keun Kim and Joon Ki Paik},
title = {Fast Image Restoration for Reducing Block Artifacts Based on Adaptive Constrained Optimization},
journal = {Journal of Visual Communication and Image Representation},
year = {1998},
volume = {9},
number = {3},
pages = {234-242},
month = {September},
annote = {Uses the quantized DCT coefficients to detect the inter-block edge direction and uses this for filitering to remove the intra-block boundary artifacts of DCT-based coding. Chang uses HxV position of one coefficient, these use relative strength of all coefficients. Possible to enhance Chang with this?},
url = {papers/Kim98_Fast_Image_restoration.pdf},
bibdate = {Wednesday, April 28, 1999 at 09:55:37 (MEST)},
submitter = {Johan Garcia}
}
@article{Hsu99,
author = {Yuh-Feng Hsu and Chien-Hua Hsieh and Yung-Chang Chen},
title = {Embedded SNR scalable MPEG-2 video encoder and its associated error resilience decoding procedures},
journal = {Signal Processing: Image Communication},
year = {1999},
volume = {14},
number = {5},
pages = {397-412 },
month = {March},
annote = {Interesting mainly for its error resilience discussion. Uses the Gilbert loss model for bursty bit errors and reports good resilience to bit errors.},
url = {papers/Hsu99_MPEG2_scaling_error_res.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, April 30, 1999 at 10:06:02 (MEST)}
}
@article{Senbel99,
author = {Samah Senbel and Hussein Abdel-Wahab},
title = {Scalable and robust image compression using quadtrees},
journal = {Signal Processing: Image Communication},
year = {1999},
volume = {14},
number = {5},
pages = {425-442 },
month = {March},
annote = {Interesting since it allows "higher compression" to be achieved without transcoding. Also has provisions for error resiliency. Poorer compression performance than JPEG though.},
url = {papers/Senbel99_robust_quadtrees.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, April 30, 1999 at 11:14:04 (MEST)}
}
@article{Abdat98,
author = {Mourad Abdat and Ziad Al Kachouh and Maurice G. Bellanger},
title = {Transmission error detection and concealment in {JPEG} images},
journal = {Signal Processing: Image Communication},
year = {1998},
volume = {13},
number = {1},
pages = {45-64 },
month = {July},
annote = {Contains a discussion on transfer format resilience. The concealment itself not so intersting since is designed to handle random bitlosses on the magnitude of 2x10-4 i.e. much to small.},
url = {papers/Abdat98_JPEG_error_coceal.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, April 30, 1999 at 11:37:03 (MEST)}
}
@article{Smith96,
author = {Brian C. Smith and Lawrence A. Rowe},
title = {Compressed Domain Processing of {JPEG}-encoded images},
journal = {Real-Time Imaging},
year = {1996},
volume = {2},
number = {1},
pages = {3-17 },
month = {February},
annote = {Describes an approach using tensors to do a two step DCT domain processing. First, an off-line step computes a tensor for each image, then, the real-time processing use the tensor and the image to perfrom the processing. The use of tensors overcomes the block-characteristcs for processing. Does not mention DCT domain recompression. },
url = {papers/Smith96_DCTdomain_processing.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, April 30, 1999 at 12:21:11 (MEST)}
}
@techreport{Bajaj99a,
author = {Sandeep Bajaj and Lee Breslau and Deborah Estrin and Kevin Fall and Sally Floyd and Padma Haldar and Mark Handley and Ahmed Helmy and John Heidemann and Polly Huang and Satish Kumar and Steven McCanne and Reza Rejaie and Puneet Sharma and Kannan Varadhan and Ya Xu and Haobo Yu and Daniel Zappala},
title = {Improving Simulation for Network Research},
institution = {University of Southern California},
year = {1999},
number = {99-702},
month = {March},
url = {http://www.isi.edu/~johnh/PAPERS/Bajaj99a.html},
copyrightholder = {author},
organization = {USC/Information Sciences Institute},
keywords = {vint, ns, nam, network simulation}
}
@article{Han99,
author = {R. Han and D. Messerschmitt},
title = {A progressively reliable transport protocol for interactive wireless multimedia},
journal = {Multimedia Systems, Springer-Verlag},
year = {1999},
volume = {7},
number = {2},
pages = {141-156},
annote = {The idea is the same as their earlier paper, leak pakets with bit errors to the application and allow for refinement by retransmissions. Contains a section on choosing the optimal packet size for a given header size and BER. Also contains several references worth checking out. },
url = {papers/Han99_progressively_reliable_tpprot.pdf},
bibdate = {Tuesday, May 04, 1999 at 11:24:14 (MEST)},
submitter = {Johan Garcia}
}
@article{Jianhua98,
author = {Lu Jianhua and Ming L. Liou and K. Ben Letaief and Justin C-I Chuang},
title = {Mobile image transmission using combined source and channel coding with low complexity concealment},
journal = {Signal Processing: Image Communication},
year = {1998},
volume = {12},
number = {2},
pages = {87-104},
month = {April},
annote = {Discusses the transmission of images over a raw 22kbps GSM channel. Suggests the use of FEC to combat some channel errors, block shuffling to distribute bursty errors and ARQ to recover essential header data and tables. Suggest the insertion of standard JPEG resynchronization markers for each block, at considerable overhead (>25%). Gives optimimization expression for selection of FEC parameters. Well presented but no new ideas. Using GSM builtin channel coding and Changs scheme will produce better results at comparable complexity (no software FEC...)},
url = {papers/Jianhua98_Mobile_Img_trans_err_conceal.pdf},
bibdate = {Tuesday, May 04, 1999 at 12:31:06 (MEST)},
submitter = {Johan Garcia}
}
@article{Balakrishnan95b,
author = {Hari Balakrishnan and Srinivasan Seshan and Randy H. Katz},
title = {Improving Reliable Transport and Handoff Performance in Cellular Wireless Networks },
journal = {ACM Wireless Networks},
year = {1995},
volume = {1},
number = {4},
month = {December},
annote = {This is an extended and much-revised journal version of the Mobicom 1995 paper. Significant changes include a better description of the protocol and sections on how the Daedalus low-latency multicast-based handoff protocol and the snoop protocol are integrated together. },
url = {http://www.cs.berkeley.edu/~ss/papers/winet/winet.ps.gz},
bibdate = {Tuesday, May 04, 1999 at 15:39:56 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Kanal78,
author = {Laveen N. Kanal and A. R. K. Sastry},
title = {Models for Channels with Memory and Their Application to Error Control},
journal = {Proceedings of the {IEEE}},
year = 1978 ,
volume = 66,
number = 7,
pages = {724-744},
month = {July},
bibdate = {Tuesday, May 04, 1999 at 18:39:56 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@techreport{Asplund99,
author = {Katarina Asplund and Anna Brunstrom and Johan Garcia and Sean Schneyer and Karl-Johan Grinnemo},
title = {{PRTP} -- A Partially Reliable Transport Protocol for Multimedia Applications: Background Information and Analysis},
institution = {Karlstad University},
year = 1999,
type = {Karlstad University Studies},
number = {1999:05},
annote = {This prestudy report presents extensive background information in the area of multimedia communication and an evaluation of the initial design decisions made for PRTP. The background study covers multimedia application requirements, both user induced and coding induced, as well as multimedia networking technologies. An overview of existing protocols is provided. The mechanisms of partial reliability are covered in detail. In the evaluation, the use of TCP as a design base for PRTP is assessed. In addition, some possible applications of PRTP are suggested. The design of partial reliability in PRTP is also outlined.}
}
@article{Schneyer99,
author = {Sean Schneyer and Johan Garcia and Anna Brunstrom and Katarina Asplund},
title = {{PRTP}: A Partially Reliable Transport Protocol for Multimedia Applications},
journal = {Proc. International Symposium on Intelligent Multimedia and Distance Education (ISIMADE'99)},
year = 1999,
month = {August},
annote = {This paper provides an overview of PRTP and the application testbed developed to test the protocol. Some key characteristics of partial reliability are also discussed. The overview of PRTP focuses on the partial reliability mechanism of the protocol. The design of PRTP as a receiver-based protocol and an extension to TCP is also described.}
}
@article{vint,
author = {VINT},
title = {Project Overview},
journal = {http://netweb.usc.edu/vint/project\_overview.html},
year = {1997},
month = {October},
annote = {This is the project overview for the VINT (Virtual InterNetwork Testbed). This is the umbrella project that has taken over the ns simulator and developed the ns2 version. They are also responsible for nam (network animator) and are working on nam2.},
url = {http://netweb.usc.edu/vint/project_overview.html},
bibdate = {Thursday, May 20, 1999 at 18:01:45 (MEST)},
submitter = {Sean Schneyer}
}
@article{Wetherall95,
author = {David Wetherall},
title = {{MIT} {Object Tcl}},
journal = {ftp://ftp.tns.lcs.mit.edu/pub/otcl/},
year = {1995},
month = {September},
annote = {This is the OTcl package which is used (and extended) by the ns simulator.},
url = {ftp://ftp.tns.lcs.mit.edu/pub/otcl/ },
submitter = {Sean Schneyer},
bibdate = {Saturday, May 22, 1999 at 22:51:14 (MEST)}
}
@manual{Fall99,
title = {\emph{ns} Notes and Documentation},
author = {Kevin Fall (Ed.) and Kannan Varadhan (Ed.)},
organization = {The VINT Project},
month = {April},
year = 1999,
annote = { This is the reference manual for the ns simulator. }
}
@article{Chang94,
author = {S-F Chang and A Eleftheriadis},
title = {Error Accumulation of Repetitive Image Coding},
journal = {Proc. IEEE Intl. Symposium on Circuits and Systems},
year = {1994},
month = {May},
annote = {Discusses the error accumulation for DCT and motion compensation. Shows that extra noise is added when recoding with a different quantizer.},
bibdate = {Thursday, May 27, 1999 at 17:07:44 (MEST)},
submitter = {Johan Garcia}
}
@article{IJG99,
author = { },
title = {Independent JPEG Group Software},
journal = {http://www.ijg.org/},
annote = {Industry quality JPEG library.},
bibdate = {Monday, May 31, 1999 at 11:36:43 (MEST)},
submitter = {Johan Garcia}
}
@article{Rabbit99,
author = { },
title = {RabbIT Proxy},
journal = {http://www.nada.kth.se/projects/prup98/web\_proxy/},
annote = {Homepage of the RabbIT proxy.},
bibdate = {Monday, May 31, 1999 at 12:50:31 (MEST)},
submitter = {Johan Garcia}
}
@article{Amir95,
author = {Elan Amir and Steve McCanne and Hui Zhang },
title = {An Application Level Video Gateway },
journal = {In Proc. ACM Multimedia '95, San Francisco, CA},
year = {1995},
month = {November},
annote = {Presents a video gateway that manipulates RTP and RTCP packets supporting an efficient DCT domain transcoding of motion-JPEG to H.261.},
url = {papers/Amir95_App_level_video_gateway.ps},
submitter = {Johan Garcia},
bibdate = {Monday, May 31, 1999 at 16:14:36 (MEST)}
}
@article{nam,
author = {VINT},
title = {{NAM}: Network Animator},
journal = {http://www-mash.cs.berkeley.edu/nam/},
year = {1999},
month = {June},
annote = {This is the network animator used with the ns network simulator and being maintained by the VINT project.},
url = {http://www-mash.cs.berkeley.edu/nam/},
bibdate = {Tuesday, June 01, 1999 at 16:44:28 (MEST)},
submitter = {Sean Schneyer}
}
@misc{RFC2481,
title = {{RFC} 2481: {A} Proposal to add Explicit Congestion Notification ({ECN}) to {IP}},
author = {K. Ramakrishnan and S. Floyd},
month = jan,
year = {1999},
url = {ftp://ftp.internic.net/rfc/rfc1570.txt, ftp://ftp.internic.net/rfc/rfc1994.txt, ftp://ftp.internic.net/rfc/rfc2284.txt, ftp://ftp.internic.net/rfc/rfc2481.txt, ftp://ftp.math.utah.edu/pub/rfc/rfc1570.txt, ftp://ftp.math.utah.edu/pub/rfc/rfc1994.txt, ftp://ftp.math.utah.edu/pub/rfc/rfc2284.txt, ftp://ftp.math.utah.edu/pub/rfc/rfc2481.txt},
status = {EXPERIMENTAL},
acknowledgement = ack-nhfb,
bibdate = {Sat Jan 23 08:05:58 MST 1999},
updates = {Updates RFC2284, RFC1994, RFC1570},
format = {TXT=64559 bytes},
online = {yes}
}
@article{Batra98,
author = {Pankaj Batra and Shih-Fu Chang},
title = {Effective algorithms for video transmission over wireless channels},
journal = {Signal Processing: Image Communication},
year = {1998},
volume = {12},
number = {2},
pages = {147-166 },
month = {April},
annote = {Discusses a combination of content specific FEC and ARQ techniques to be used for transmisison of video over mobile links. Centered around DCT-based techniques. Uses coding knowledge to distribute FEC (JSCC) and use selective ARQ (i.e. partial reliability) to constain loss and delay (playout buffering is used). Shows clear gains from using ARQ. },
url = {papers/Batra98_Eff_alg_video_wireless.pdf},
bibdate = {Friday, June 04, 1999 at 15:26:57 (MEST)},
submitter = {Johan Garcia}
}
@article{Alanko99,
author = {Timo Alanko and Markku Kojo and Mika Liljeberg and Kimmo Raatikainen},
title = {Mobile access to the {Internet}: a mediator-based solution},
journal = {Internet Research },
year = {1999},
volume = {9},
number = {1},
pages = {58-65},
annote = {Discusses some of the problems of running TCP over RLP. States that RLP retransmissions cause TCP timeouts and subsequent retransmissions, among other problems.},
url = {papers\Alanko99_Mobile_inet_TCP-RLP.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, July 02, 1999 at 11:29:54 (MEST)}
}
@article{Foo99,
author = {Schubert Foo and Siu Cheung Hui and See Wai Yip},
title = {Enhancing the quality of low bit-rate real-time Internet communication services},
journal = {Internet Research},
year = {1999},
volume = {9},
number = {3},
pages = {212-224},
annote = {Presents a communication system for transmitting multimedia data. Basically a collection of "old" methods put together: compression, buffering, dynamic rate control, redundant transmission of audio data for packet loss recovery, silence deletion and a virtual video play-out mechanism where old frames are used. Runs on top of RTP.},
url = {papers/foo.pdf},
bibdate = {Monday, July 19, 1999 at 17:38:45 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Ghinea98,
author = {G. Ghinea and J. P. Thomas},
title = {QoS Impact on User Perception and Understanding of Multimedia Video Clips},
journal = {ACM Multimedia'98},
year = {1998},
pages = {49-54},
annote = {Reports on an experimental study evaluating how well users pick up the information content in multimedia clips (audio+video) at varying quality. Shows that the presentation can be fairly bad without much loss in information. Primarily interesting to see how they've selected the clips etc. (done pretty informally) with our "image experiment" in mind.},
url = {papers/p49-ghinea.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Monday, July 19, 1999 at 17:56:21 (MEST)}
}
@article{Watson98,
author = {Anna Watson and M. Angela Sasse},
title = {Measuring Perceived Quality of Speech and Video in Multimedia Conferencing Application},
journal = {ACM Multimedia'98},
year = {1998},
pages = {55-60},
annote = {Considers how multimedia QoS can be assesed "subjectively" by the user. Argues that the methods suggested by ITU for subjective assessment are inadequate. Discusses how new terminology and more suitable scales can be developed. Interesting with respect to our "image experiment". },
url = {papers/p55-watson.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Monday, July 19, 1999 at 18:05:36 (MEST)}
}
@article{Rhee98,
author = {Injong Rhee},
title = {Retransmission-based Error Control for Internet Video Applications over the Internet},
journal = {IEEE International Conference on Multimedia Computing and Systems},
year = {1998},
month = {June},
annote = {Describes a retransmission-based technique for improved video quality. Retransmitted packets are used to improve the quality of reference frames and hence can be used even after their display time. Implementation and experimental results for a version based on H261 are described. Nicely written paper.},
url = {papers/icmcs-rhee.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Wednesday, August 25, 1999 at 17:10:16 (MEST)}
}
@article{Rojas-Cardenas99,
author = {L. Rojas-Cardenas and L. Dairaine and P. Senac and M. Diaz},
title = {An Adaptive Transport Service for Multimedia Streams},
journal = {IEEE International Conference on Multimedia Computing and Systems},
year = {1999},
month = {June},
annote = {Describes how a POC (partially ordered and partially reliable) service can be modelled, outlines the POC protocol architecture and shows how it has been used in designing an adaptive MPEG video server. Not to much detail on the protocol implementation.},
url = {papers/icmcs-poc.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Wednesday, August 25, 1999 at 18:00:36 (MEST)}
}
@article{Charrier99,
author = {Maryline Charrier and Diego santa Cruz and Mathias Larsson},
title = {JPEG2000: A New Standard for Still Image Compression},
journal = {IEEE International Conference on Multimedia Computing and Systems},
year = {1999},
month = {June},
annote = {Provides a brief overview of JPEG2000 and describes two demonstrations, a Java decoder implementation and progressive transmission over a limited bandwidth channel. As a sidenote states that 80% of web images are JPEG encoded, which was kind of interesting.},
url = {papers/icmcs-jpeg2000.pdf},
bibdate = {Wednesday, August 25, 1999 at 18:25:18 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Dwyer98,
author = {Dane Dwyer and Sungwon Ha and Jia-Ru Li and Vaduvur Bharghavan },
title = {An Adaptive Transport Protocol for Multimedia Communication},
journal = {IEEE International Conference on Multimedia Computing and Systems},
year = {1998},
month = {June},
annote = {Presents HPF, a transport protocol for heterogenous packet flows. HPF supports sending multiple flows of packets over a single stream. Reliable (high priority) and unrealiable (low priority) flows can be combined. In a sense this gives a partially reliable stream-oriented protocol. Congestion control is decoupled from the reliability mechanism and based on feedback from the receiver on the fraction of packets received in a window. This is also interesting with respect to PRTP. Mentions that a custom WWW proxy that runs over HPF has been designed. },
url = {papers/icmcs-dwyer.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Thursday, August 26, 1999 at 09:31:46 (MEST)}
}
@article{Shiu99,
author = {Da-Jin Shiu and Chia-Chiang Ho and Ja-Lin Wu },
title = {A DCT-Domain H.263 Based Video Combiner for Multipoint Continuous Presence VideoConferencing},
journal = {IEEE International Conference on Multimedia Computing and Systems},
year = {1999},
month = {June},
annote = {Describes how 4-6 H.263 video streams can be combined into one. The combines stream is intended for multiparty conferencing. The merge is performed in the DCT-domain which provides a (weak) connetion to our JPEG recoder.},
url = {papers/icmcs-dctvideo.pdf},
bibdate = {Thursday, August 26, 1999 at 10:25:26 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Loon97,
author = {T. S. Loon and V. Bharghavan},
title = {Alleviating the Latency and Bandwidth Problems in WWW Browsing.},
journal = {Usenix Symposium on Internet Technologies and Systems '97},
year = {1997},
month = {December},
annote = {Describes a proxy system that addresses three problems; access over low bandwith links, long and variable delays, disconnected operation. The proxy system monitors user and group access patterns and perform prefetching, filter HTTP traffic in order to reduce data traffic over bottleneck links and hoards documents in anticipation of disconnected operation. No refernce to HPF, but several references to full length version of paper. The paper I got from the research groups website may be the extended abstract for the version that appeared in the proceedings?},
url = {papers/usits_v1.ps},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Thursday, August 26, 1999 at 11:35:30 (MEST)}
}
@article{Stangel98,
author = {M. Stangel and V. Bharghavan},
title = {Improving {TCP} Performance in Mobile Computing Environments},
journal = {International Conference on Communications '98, Atlanta, GA},
year = {1998},
month = {June},
annote = {Describes a version of TCP (Mobile TCP) that is mobility aware at both the mobile host and the stationary correspondent host. Packet loss due to handoff, interface switching and congestion (default) are distinguised. The mobile host detects handoffs and interface switches and informs the correspondent host. Special connection management allows multiple addresses to be used to identify a transport end point during a connection. },
url = {papers/icc98-stangel.ps},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Thursday, August 26, 1999 at 12:27:35 (MEST)}
}
@article{Peng99,
author = {Fei Peng, Jian Ma},
title = {An Effective way for Enhancement of {TCP} Performance in Wireless and Mobile Networks},
journal = {Internet Draft},
year = {1999},
pages = {11},
month = {August},
annote = {This paper describes a novel method to enhance TCP performance over wireless and mobile networks. The method, called Wireless-ECN (Explicit Congestion Notification), lets the nodes in a network signal congestion by setting the WECN-bit (bit 8 in the reserved field of a TCP-header).},
url = {draft-fpeng-wecn-00.txt},
bibdate = {Monday, September 13, 1999 at 17:08:26 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Sculzrinne98,
author = {H. Schulzrinne, A. Rao, R. Lanphier},
title = {Real Time Streaming Protocol},
journal = {Internet Draft},
pages = {74},
month = {February},
annote = {RTSP, Real Time Streaming Protocol, administers video and/or audio streams via a HTTP like protokol. RTSP makes use of one or several UDP/TCP/RTP sessions in order to accomplish a media presentation.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 13, 1999 at 18:31:06 (MEST)}
}
@article{Greg99,
author = {Greg Roelofs},
title = {PNG Gaining Acceptance},
journal = {Webreview.com},
year = {1999},
pages = {5},
month = {August},
annote = {PNG, Portable Network Graphics, is a compressed format for storing bitmapped (raster) images. Like GIF, PNG uses a lossless compression method. The main features of PNG are: 1) support for palette images (GIF), 2) support for alpha transparency, which allows different portions of an image to have varying levels of transparency, 3) support for a two-dimensional interlacing scheme, close in appearance to JPEG.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 13, 1999 at 19:19:00 (MEST)}
}
@article{Derrick99,
author = {Derrick Story},
title = {JPEG2000 - More Than New Millennium Buzz},
journal = {Webreview.com},
year = {1999},
pages = {4},
month = {August},
annote = {This article gives a brief overview of JPEG2000. JPEG2000 differs from JPEG in that it uses Wavelet technology for image compression instead of DCT (Dicrete Cosine Transform), making it possible to give images a much smoother look without having to resort an abundance of fourier coefficients. It also uses a default colospace, sRGB, which enables a more accurate representation of colors irrespective of target platform. JPEG2000 also contains a vast improvement in metadata management, e.g. you can specify the digital camera used to record the image.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 13, 1999 at 19:32:28 (MEST)}
}
@article{Janus99,
author = {Janus Boye},
title = {SVG Brings Fast Vector Graphics to the Web},
journal = {Webreview.com},
year = {1999},
pages = {5},
month = {August},
annote = {The SVG (Scalable Vector Graphics) standard is currently a working draft at the W3C. Contrary to graphics formats like GIF and JPEG, SVG are vector based, which makes it immune to pixel constraints. Some of the nice features of SVG are: high-resolution printing, high-performance zooming and panning inside of graphics without reloading, animation, filter, kerning, masking, and scripting. Since SVG is based on XML, it is entirely text-based, which allows search engines to index SVG graphics, and users to search for text in them.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 13, 1999 at 19:40:14 (MEST)}
}
@article{Nosratinia99,
author = {A. Nosratinia},
title = {Embedded Post-Processing for Enhancement of Compressed Images},
journal = {Data Compression Conference 99, Snowbird, Utah},
month = {March},
annote = {Describes what seem to be a useable post procressing apporach for reducing the blockyness of images by reapplying DCT at different pixel offsets in relation to the original block. Could be used for only postprocess block reconstructed, considerably reducing the needed computing time.},
url = {papers/Nosratinia99_PostrProcImg.pdf},
bibdate = {Tuesday, September 14, 1999 at 08:27:09 (MEST)},
submitter = {Johan Garcia}
}
@article{Perkins99,
author = {Charles E. Perkins},
title = {Mobile networking in the Internet },
journal = {Mobile Networks and Applications},
year = {1999},
volume = {3},
number = {4},
pages = {319-334},
annote = {Describes problems encountered when protocols are developed for mobility. The paper follows the classical layered model. Gives an overview; no details.},
url = {papers/p319-perkins.pdf},
bibdate = {Tuesday, September 21, 1999 at 18:20:31 (MEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Conrad98,
author = {P. Conrad and P. Amer and M. Taube and G. Sezen and S. Iren and A. Caro},
title = {Testing environment for innovative transport protocols },
journal = {Proc. MILCOM '98},
year = {1998},
month = {October},
annote = {A short paper (5 pages) which describes the development of the Universal Transport Library (UTL). The library provides a single API to different transport protocols. This makes it easier to test applications with different transport protocols and to develop prototypes for new transport protocols.},
url = {papers/milcom98-conrad.ps},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Wednesday, September 22, 1999 at 14:46:09 (MEST)}
}
@article{Lucas97a,
author = {Matthew Lucas and Dallas Wrege and Bert Dempsey and Alfred Weaver},
title = {Statistical Characterization of Wide-Area IP Traffic},
journal = {Sixth International Conference on Computer Communications and Networks (IC3N'97)},
year = {1997},
month = {September},
annote = {The statisical characterisation is based on a week long packet trace of nearly a billion IP packets. The results could provide a basis for background traffic modelling. The findings are: packet sizes are short-term correlated, the probability distribution of packet arrivals is log-normal, and packet size and correlation structure is independent of time of day and network load. Hard to read. Requires in-depth knowledge of statistics.},
url = {papers/ic3n-2col.pdf},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Wednesday, September 22, 1999 at 16:01:57 (MEST)}
}
@article{Kumar98,
author = {Anurag Kumar},
title = {Comparative Performance Analysis of Versions of {TCP} in a Local Network with a Lossy Link},
journal = {IEEE/ACM Transactions on Networking},
year = {1998},
volume = {6},
number = {4},
pages = {485-498},
month = {August},
annote = {A study of the throughput of various versions of TCP in the presence of random losses on a wireless link. It is shown, among other things, that setting the duplicate ACK threshold for fast retransmission to 1 can double the throughput (in the range 5% - 20% loss)},
submitter = {Katarina Asplund},
bibdate = {Monday, September 27, 1999 at 11:52:45 (MEST)}
}
@article{Clark98,
author = {David D. Clark and Wenjia Fang},
title = {Explicit Allocation of Best-Effort Packet Delivery Service},
journal = {IEEE/ACM Transactions on Networking},
year = {1998},
volume = {6},
number = {4},
pages = {362-373},
month = {August},
annote = {This paper presents the "allocated-capacity" framework for providing different levels of best-effort service in times of congestion. The basic idea is that all packets that enter the network are tagged as either "in" or "out", and routers then drop "out" packets with a higher probability than "in" packets.},
bibdate = {Monday, September 27, 1999 at 12:14:31 (MEST)},
submitter = {Katarina Asplund}
}
@article{Dovrolis99,
author = {C. Dovrolis and D. Stiliadis and P. Ramanathan},
title = {Proportional Differentiated Services: Delay Differentiation and Packet Scheduling},
journal = {SIGCOMM 99},
year = {1999},
annote = {The paper proposes a proportional differentiation between service classes in order to achieve both a predictable and a controllable differentiation. Predictable here means that higher classes always should get better, or at least no worse, service than lower classes. Controllable means that network operators should be able to adjust the quality spacing between classes. },
url = {http://www.acm.org/sigcomm/sigcomm99/papers/session3-3.html},
bibdate = {Monday, September 27, 1999 at 12:41:52 (MEST)},
submitter = {Katarina Asplund}
}
@article{Stoica99,
author = {Ion Stoica and Hui Zhang},
title = {Providing Guaranteed Services Without Per Flow Management},
journal = {SIGCOMM 99},
year = {1999},
annote = {The paper proposes techniques for providing guaranteed service where only edge routers maintain QoS state. The authors claim that their approach is much more scalable and robust than the stateful Integrated Services approach.},
url = {http://www.acm.org/sigcomm/sigcomm99/papers/session3-1.html},
bibdate = {Monday, September 27, 1999 at 14:34:43 (MEST)},
submitter = {Katarina Asplund}
}
@article{Rhee98a,
author = {Injong Rhee},
title = {Error Control Techniques for Interactive Low-bit Rate Video Transmission over the Internet},
journal = {SIGCOMM98},
year = {1998},
annote = {This paper presents an error recovery scheme for interactive video based on retransmissions and FEC. Layered coding is used, where the essential signals is protected by FEC. In addition, lost packets from reference frames are retransmitted, and if they arrive after their playout times, they are used to reduce error propagation to the dependent frames. },
url = {http://www.acm.org/sigcomm/sigcomm98/tp/abs_24.html},
bibdate = {Monday, September 27, 1999 at 15:29:04 (MEST)},
submitter = {Katarina Asplund}
}
@article{Rejaie99,
author = {R. Rejaie and M. Handley},
title = {Quality Adaptation for Congestion Controlled Video Playback over the Internet},
journal = {SIGCOMM 99},
year = {1999},
annote = {This paper presents a congestion control scheme for streaming video. Layers of the video stream is added and dropped to perform coarse-grain adaptation, and a TCP-friendly congestion control mechanism is used to react to congestion on very short timescales.},
url = {http://www.acm.org/sigcomm/sigcomm99/papers/session5-3.html},
submitter = {Katarina Asplund},
bibdate = {Monday, September 27, 1999 at 15:41:13 (MEST)}
}
@article{Breslau98,
author = {L. Breslau and S. Schenker},
title = {Best-Effort versus Reservations: A simple Comparative Analysis},
journal = {SIGCOMM 98},
year = {1998},
annote = {This paper discusses if the Internet should retain its best-effort only architecture, or if it should adopt one that is reservation-capable. Using a simple analytical model, pros and cons are presented and discussed.},
url = {http://www.acm.org/sigcomm/sigcomm98/tp/abs_01.html},
submitter = {Katarina Asplund},
bibdate = {Monday, September 27, 1999 at 15:53:14 (MEST)}
}
@article{Barford99,
author = {P. Barford and M. E. Crovella},
title = {Measuring Web Performance in the Wide Area},
journal = {Performance Evaluation Review},
year = {1999},
month = {August},
annote = {This paper focuses on studying WWW performance from an integrated standpoint, i.e. to take both web server performance and network characteristics into account. Relevant contents is a small discussion on TCPs packet loss performance degradation, aspects on measuring network traffic and some references that should be checked out (Paxsons analytical model mm).},
url = {papers/Barford99_WebPerformance.ps},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 05, 1999 at 09:29:14 (MEST)}
}
@inproceedings{Ratnam98,
author = {K. Ratnam and I. Matta},
title = {{WTCP}: An Efficient Transmission Control Protocol for Networks with Wireless Links. },
booktitle = {Proceedings of the IEEE Symposium on Computers and Communications (ISCC)},
address = {Athens, Greece},
year = {1998},
month = jun,
annote = {The paper proposes a scheme based on local retransmissions by the wireless basestation. The basestation buffers TCP segments and performs independent flow and error control over the wireless link. It does retains TCPs end-to-end semantics by not issuing ack to the sender until it has received an ack from the mobile receiver. No modification to either the sender or the receiver is necessary. It hides the time taken to perform the wireless retransmssions from the senders RTT calculation by faking the timestamp of the returned ack. The scheme looks interesting, but the results are only simulation based, and they assume a somewhat low wireless RTT (order 10ms). NS-2 code could probably be fetched from here. },
url = {papers/Ratnam98_WTCP_BasestnRetransmit.ps},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 05, 1999 at 10:20:02 (MEST)}
}
@article{Ratnam97,
author = {K. Ratnam and I. Matta.},
title = {{WTCP}: An Efficient Transmission Control Protocol for Networks with Wireless Links. },
journal = {Tecnical Report NU-CCS-97-11 },
year = {1997},
annote = {A covering the same as the paper above, but with some more detail.},
url = {papers/Ratnam97_WTCP_Techreport.ps},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 05, 1999 at 10:43:25 (MEST)}
}
@article{Zenel97a,
author = {Bruce Zenel and Dan Duchamp},
title = {A General Purpose Proxy Filtering Mechanism Applied to the Mobile Environment },
journal = {Mobicom'97},
year = {1997},
month = {October},
annote = {This paper describes a two level proxy setup with a low level proxy for network/transport layer protocols (IP,ICMP,UDP,TCP) and a high level proxy for application layer protocols (HTTP, MPEG, SMTP). They have implemented some filters, among others a HTTP compression based on Zlib, just as us. They report improvments over slow links. No image recoding filter implemented though. },
url = {papers/Zenel97a_ProxyMobile},
bibdate = {Wednesday, October 06, 1999 at 16:03:38 (MEST)},
submitter = {Johan Garcia}
}
@article{Brooks95,
author = {Charles Brooks and Murray S. Mazer and Scott Meeks and Jim Miller},
title = {Application-Specific Proxy Servers as HTTP Stream Transducers.},
journal = {Proc. 4th Intl WWW Conference, Boston MA},
year = {1995},
month = {December},
annote = {Describes OreOs, a general proxy framework with two "wafers" that communicate with the client and server side respectively and a "filling" that performs the filtering/transformation. },
url = {http://www.w3.org/Conferences/WWW4/Papers/56/},
bibdate = {Thursday, October 07, 1999 at 09:32:06 (MEST)},
submitter = {Johan Garcia}
}
@article{Fox96a,
author = {Armando Fox and Steven D. Gribble and Eric A. Brewer and Elan Amir },
title = {Adapting to Network and Client Variability via On-Demand Dynamic Distillation},
journal = {Proc. Seventh Intl. Conf. on Arch. Support for Prog. Lang. and Oper. Sys. (ASPLOS-VII), Cambridge, MA },
year = {1996},
month = {October},
annote = {This paper is an extension/refinement of Fox's earlier paper on distillation and provides some design principles and a short discussion on their GloMop modular proxy. },
url = {papers/Fox96_DynamicDestillation.pdf},
submitter = {Johan Garcia},
bibdate = {Thursday, October 07, 1999 at 09:50:32 (MEST)}
}
@article{Bharadvaj98,
author = {Harini Bharadvaj and Anupam Joshi and Sansanee Auephanwiriyakul },
title = {An Active Transcoding Proxy to Support Mobile Web Access},
journal = {Proceedings of the 17th IEEE Symposium on Reliable Distributed Systems},
year = {1998},
annote = {The paper presents a proxy system called MOWSER used to enhance mobile web browsing. Contains a webbrowser and perl scripts for managing user preferences. They support HTTP 1.1 content negotiation if the server happens to have several versions of an image. HTML, Image and video trascoding is performed, but not with much finesse (except maybe for the video).},
url = {papers/Bharadvaj98_TranscodingProxyMobile.pdf},
submitter = {Johan Garcia},
bibdate = {Thursday, October 07, 1999 at 16:08:00 (MEST)}
}
@misc{rfc2582,
author="S. Floyd and T. Henderson",
title={The {NewReno} Modification to {TCP}'s Fast Recovery Algorithm},
series="Request for Comments",
number="2582",
howpublished="RFC 2582 (Experimental)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1999,
month=apr,
note="Obsoleted by RFC 3782",
url="http://www.ietf.org/rfc/rfc2582.txt",
}
@article{Balakrishnan98,
author = {Hari Balakrishnan and Venkata Padmanabhan and Srini Seshan and Mark Stemm and Randy H. Katz},
title = {{TCP} Behavior of a Busy Internet Server: Analysis and Improvements},
journal = {Proc. IEEE Infocom},
year = {1998},
month = {March},
annote = {About the interaction between TCP and HTTP. Analyses the TCP behavior of a busy Web server (real-world traffic trace). Individual and parallel TCP connections are examined. Results: The loss recovery techniques are not effective. Parallel connections are overly aggressive users of network. Modifications to the server-side TCP are proposed as a solution: Enhanced or rigth edge loss recovery (can be combined with SACK) which improves the effectiveness of TCP fast retransmission when window sizes are small. TCP-Int for parallel connections: an integrated approach to congestion control and loss recovery across parallel conncetions.},
url = {papers/infocom98.ps},
bibdate = {Monday, October 11, 1999 at 09:31:41 (MEST)},
submitter = {Annika Wennstrom}
}
@article{Granbohm99,
author = {H. Granbohm and J. Wiklund},
title = {{GPRS} - General packet radio service},
journal = {Ericsson Review},
year = {1999},
volume = {2},
pages = {82-88},
annote = {By adding GPRS to the GSM network, operators can offer efficient wireless access to external IP-based networks, such as Internet and corporate intranets. End-users can remain connected indefinitely to the external network and have transfer rates of up to 115 kbit/s. This article briefly explains the changes to the GSM net imposed by GPRS. It should be noted that the article has a bias towards Ericsson's product AXB 250 which accommodates both SGSN (Serving GPRS Support Node) as well as GGSN (Gateway GPRS Support Node) functionality.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, October 11, 1999 at 15:41:03 (MEST)}
}
@article{Rysavy98,
author = {P. Rysavy},
title = {PAPER: General Packet Radio Service},
journal = {GSM Data Today online journal},
year = {1998},
pages = {7},
month = {September},
annote = {Gives a brief introduction to GPRS; What it is; What the main incentives are. At the end of the article, the author gives a very succinct introduction to the more technical matters underlying GPRS.},
bibdate = {Monday, October 11, 1999 at 19:05:34 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Trillium99,
author = {Trillium Digital Systems, Inc.},
title = {General Packet Radio Service (GPRS) White Paper},
journal = {www.trillium.com},
year = {1999},
pages = {8},
month = {May},
annote = {Gives a brief introduction to the foundations of GPRS. More specifically, the most important new nodes are introduced and succinctly described. A very short description of how routing works in GPRS is given.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, October 11, 1999 at 19:10:23 (MEST)}
}
@article{Campbell99,
author = {Andrew T. Campbell and Javier Gomez and András G. Valkó},
title = {An Overview of Cellular IP},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {606-611},
month = {September},
annote = {Discusses a third alternative for handling mobility besides Mobile IP and 3G mobile networks. Cellular IP is designed for small up to MAN size cellular networks. A gateway router is located in the intersection of the Internet and the ceular network, and it periodically broadcasts beacon packets that flood the network. The base stations use this packet to build a route back to the host. Mobile hosts that enter a base stations area get recorded in the soft state routing cache that the base station holds for its mobile hosts. The routing cache of the intermediate base stations to the gateway also gets updated as a packet from the mobile host traverses the network to the gateway.Idle mobile hosts may send route update packages to the gateway to uphold the routing cache entries or may let the routing cache entries expire and use paging messages to inform the network of their location. Unlocated mobile hosts are serched for by broadcasting in the entire access network. Handoffs are managed by semisoft handoffs that ensure that packets are routed to the new basestation before leaving the old one. By appropiately delaying the stream to the new basestation, a transparent handoff can be achieved.},
submitter = {Johan Garcia},
bibdate = {Monday, October 11, 1999 at 23:53:32 (MEST)}
}
@article{Furuskar99,
author = {Anders Furusk\"{a}r and David Bl{\aa}sj\"{o} and Stefan Eriksson and Magnus Frodigh and Stefan J\"{a}verbring and H{\aa}kan Olofsson},
title = {System Performance of the {EDGE} Concept for Enhanced Data Rates is {GSM} and {TDMA/136}},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {752-756},
month = {September},
annote = { The paper focuses on the system performance of the EDGE concept and gives an overview of the EDGE system. EDGE shares its basical radio interface characteristics such as 200 kHz channel spacing and 8 time slots of GSM. The modulation is however different where EDGE uses 8PSK instead of GMSK, which increase the gross data rate to 554 kbps/carier (which includes training sequence and other overhead). Mentions that link adaption can be performed by either Link Adaption (i.e. changing the modulation and FEC) or by Incremental Redundancy (i.e. hybrid II/III selective repeat ARQ). A WWW Traffic model distribution including user behaiviour and transfer sizes is also given. Theit conclusion is that EDGE can provide user data rates exceeding 384 kbps corresponding to a threefold improvment over standard GPRS.},
bibdate = {Monday, October 11, 1999 at 23:55:59 (MEST)},
submitter = {Johan Garcia}
}
@article{Prakash99,
author = {Ravi Prakash and Meghana Saharabudhe},
title = {Modifications to {TCP} for Improved Performance and Reliable end-to-end Communiction in Wireless Networks},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {938-943},
month = {September},
annote = {Paper is poorly titled, it only deals with ad-hoc wireless networks. The main idea is to use routing protocol information to send Route Failure Notification messages to the sender via ICMP when a route fail due to node mobility. The sender then freezes its TCP timers until it receives a Route Reestablishent Notification},
submitter = {Johan Garcia},
bibdate = {Monday, October 11, 1999 at 23:57:32 (MEST)}
}
@article{Anjum99,
author = {Farooq Anjum and L. Tassiulas},
title = {An Analytical Model for the various {TCP} Algorithms Operating over a Wireless Channel},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {943-947},
month = {September},
annote = {This paper describes an analytical model that deals with the data transfer phase only. Karn's algorithm, loss of fast retransmitted packets and ACK losses are simplified away. The analytical model is based on the concepts of cycles containg mini-cycles which generate packet trains. The analytial model is claimed to give results near those obtained by simuation.},
bibdate = {Monday, October 11, 1999 at 23:58:47 (MEST)},
submitter = {Johan Garcia}
}
@article{Tran-Gia99,
author = {P Tran-Gia and K Leibnitz},
title = {Teletraffic Models and Planning in Wireless IP networks},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {598-602},
month = {September},
annote = {Interesting mainly because it gives some details on the IS-95 CDMA systems RLP protocol (IS-707). It is based on 20ms 171 bit frames with SR-ARQ. It is also said that CDMA power control will keep the Frame Error Rate at around 1%.},
bibdate = {Tuesday, October 12, 1999 at 00:00:21 (MEST)},
submitter = {Johan Garcia}
}
@article{Yang99,
author = {N Yang},
title = {The Third Generation Wireless Network Using CDMA Air Interface},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {649-653},
month = {September},
annote = {Provides an overview of the CDMA2000 and WCDMA proposals and their relations to previous technology, focusing on the air interface. },
bibdate = {Tuesday, October 12, 1999 at 00:01:33 (MEST)},
submitter = {Johan Garcia}
}
@article{Umehira99,
author = {M Umehira and M Nakura and M Umeuchi and J Murayama and T Murai and H Hara},
title = {Wireless and IP integrated system architectures for broadband mobile multimedia services},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {593-597},
month = {September},
annote = {Describes three possible IP backbone network architectures supporting mobility. (1) Gateway architecture, for example GPRS. (2) Overlay nework architecture, for example Mobile ATM. (3) Integrated architecture, for example mobile IP. },
submitter = {Johan Garcia},
bibdate = {Tuesday, October 12, 1999 at 00:02:59 (MEST)}
}
@article{Qiu99,
author = {X Qiu and J Chuang and K Chawla and J Whitehead},
title = {Performance Comparison of Link Adaption and Incremental Redundancy in Wireless Data Networks},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {771-775},
month = {September},
annote = {Provides a discussions of FEC (part of Link Adaption) contra ARQ (here called Incremental Redundancy) from an EDGE link layer viewpoint. background. Since they do not consider delay performance, they come to the not so surprising conclusion that ARQ provides better throughput. However, they do not discuss the optimal tradeoff between FEC and ARQ},
bibdate = {Tuesday, October 12, 1999 at 00:04:23 (MEST)},
submitter = {Johan Garcia}
}
@article{Melis99,
author = {B Melis and M Mastroforti and G Romano},
title = {Link Level Performance of the W-CDMA Radio Interface for UMTS},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {913-917},
month = {September},
annote = { Introduction provides some details on some services that should be provided by UMTS (384 kbit/s Circuit switched constrained delay and 480 kbit/s packet switched unconstrained delay). Good first part, the rest is too much radio channel simulation details for our purposes. },
submitter = {Johan Garcia},
bibdate = {Tuesday, October 12, 1999 at 00:05:39 (MEST)}
}
@article{Murad99,
author = {A Murad and T Fuja},
title = {Robust Transmission of Variable-Length Encoded Sources},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {968-971},
month = {September},
annote = {This paper proposes a decoder structure that is based on combining the Finite state machine graphs of the surce, the source encoder and the channel coder. By doing this, the residual redundancy not removed by the imperfect source enoder can be used for maximum a posteriori decoding. In addition to this scheme to limit the error propagation for the VLC is suggested. It is based on suppying the dedoder with side information about the number of symbols and the number of bits in the block to be protected. This is used by the decoder in a list Viterbi decoder with trellis pruning, i.e the decoder maintains a list of the most probable paths, but prunes paths that cannot lead to the correct number of symbols. },
submitter = {Johan Garcia},
bibdate = {Tuesday, October 12, 1999 at 00:07:00 (MEST)}
}
@inproceedings{Ridge99,
author = {J Ridge and F W Ware and J D Gibson},
title = {Image refinement for lossy channels with relaxed latency constraints},
booktitle = {Proceedings IEEE Wireless Communications and Networks Conference 1999 (WCNC'99)},
year = {1999},
pages = {993-997},
month = {September},
annote = {This paper describes an enhancment to a previously developed 'smoothed description coder' (SDC) , a coder which at the encoder produces two streams of equal importance. In this case it takes every other coefficient in one stream and the other coefficients in the other stream. These coefficent runs are then shifted between every block. The decoder then interpolates from the available information. Reminds of Changs scheme, but some differences exist. Uses smarter DC-coefficient interpolation, uses the pair of blocks u-d,l-r that give the least difference. This paper describes two enhancements to SDC, namely Permuting block transmission order. This shuffles block around according to some permutation order (i.e pseudo-random). There goes my idea... Also proposes the use of selective refinement of erred blocks i.e partial reliability. Proposes an importance metric to be used to decide which erred blocks to be retransmitted. This metric is based on retransmitting blocks which cause a neighbouring block to lose its neigbour, and not retransmit low-energy blocks. Reports good results. Perhaps could SDC be improved with Changs AC coefficient interpolation? },
volume = {2},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 12, 1999 at 00:08:39 (MEST)}
}
@article{Cho99,
author = {S Cho},
title = {Adaptive error control for Hybrid (Satellite-Terrestrial) Networks},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {2},
pages = {1013-1017},
month = {September},
annote = {This paper proposes the use of a Reed-Solomon FEC and landline ARQ as a combination for achieving reliability over a satellite link. The idea is to adapt the FEC parameters according to a function based on channel estimation and feedback data. This is shown by simulation to perform better than similar non-adapting hybrid FEC and ARQ schemes. Latency is not discussed. },
submitter = {Johan Garcia},
bibdate = {Tuesday, October 12, 1999 at 00:09:30 (MEST)}
}
@inproceedings{Meyer99,
author = {M. Meyer},
title = {{TCP} Performance over {GPRS}},
booktitle = {Proceedings IEEE Wireless Communications and Networks Conference 1999 (WCNC'99)},
year = {1999},
pages = {1248-1252},
month = {September},
annote = {This paper gives a brief overview of the GPRS link layer and describes simulations that show that TCP can work well with GPRS. This is partly due to the fact that four different FEC coding schemes (CS1-4) is used by GPRS. Switching between the CSs is done adaptively according to channel conditions, and this controls the amount of RLC block errors and consequently the RLC retransmission delay. The effect of this delay is mitigated by the fact that the buffering of the TCP send window adds buffering delay, thus producing more space for delay varatio due to RLC retransmissions. The results were obtained by simulation for one user, but should hold for many users (?). },
bibdate = {Tuesday, October 12, 1999 at 00:11:18 (MEST)},
volume = {3},
submitter = {Johan Garcia}
}
@article{Balakrishnan97a,
author = {Hari Balakrishnan and Srinivasan Seshan and Mark Stemm and Randy H. Katz},
title = {Analyzing Stability in Wide-Area Network Performance},
journal = {Proc. ACM SIGMETRICS Conference on Measurement \& Modeling of Computer Systems (SIGMETRICS 97)},
year = {1997},
month = {June},
annote = {Describes statistical models for the Internet. The models are developed and analysed from a packet-level traces. Despite the heterogeneity of the Internet the troughput for Web transfers can be expressed as a random variable with a log-normal distribution. The probability distribution of throughput is almost the same for hosts close to each other. Throughput to individual host is stable for several minutes. This implies that there is promise in techniques that cache and share network characteristics amongst nearby host.},
url = {papers/sigmetrics97.ps.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Tuesday, October 12, 1999 at 08:25:55 (MEST)}
}
@article{Ericsson99,
author = {{Ericsson AB}},
title = {WebOnAir software product information},
journal = {http://mobileinternet.ericsson.se/},
annote = {Reference Link for Ericsson WebOnAir. Go to software, webonair.},
bibdate = {Thursday, October 14, 1999 at 13:10:44 (MEST)},
submitter = {Johan Garcia}
}
@article{Spectrum99,
author = {{Spectrum Information Technologies}},
title = {Fastlane},
journal = {http://stage.acunet.net/spectrum/fastlaneisp.html},
annote = {A reference link to the Fastlane recoding proxy.},
bibdate = {Thursday, October 14, 1999 at 14:43:38 (MEST)},
submitter = {Johan Garcia}
}
@article{Dugad99,
author = {Rakesh Dugad and Narendra Ahuja },
title = {A Fast Scheme for Altering Resolution in the Compressed Domain},
journal = {Proceedings of the Computer Vision and Pattern Recognition, Fort Collins, Colorado},
year = {1999},
month = {June},
annote = {This paper presents an algorithm for fast downsampling of images in the DCT domain. Using their method a later upsampling achieve considerably better quality than other schemes, and the downsampling is computationally efficient.},
url = {papers/Dugad99_FastDCTdownsampling.pdf},
bibdate = {Thursday, October 14, 1999 at 16:52:13 (MEST)},
submitter = {Johan Garcia}
}
@article{Bhaskaran96,
author = {V. Bhaskaran},
title = {Mediaprocessing In The Compressed Domain},
journal = {Proceedings of COMPCON '96},
year = {1996},
annote = {Abstract: The emergence of compression standards such as JPEG, MPEG, H.320 and H.324 has enabled many consumer and business multimedia applications wherein the multimedia content is disseminated in their compressed form. Many applications however require processing of the multimedia content prior to presentation. Traditional approaches for processing rely on first decompressing the bitstreams and then applying the desired processing function. In this paper we describe recent developments that have led to processing techniques that can be directly implemented on an intermediate representation of the compressed domain information. By avoiding the computationally expensive decompression and recompression processes needed in traditional spatial/temporal domain based techniques, mediaprocessing in the compressed domain achieves 2-4 fold speedup for typical processing functions such as image downscaling, bitrate conversions, interframe to intra- frame conversions. },
bibdate = {Friday, October 15, 1999 at 15:21:09 (MEST)},
submitter = {Johan Garcia}
}
@article{Queiroz97,
author = {Ricardo L. de Queiroz },
title = {Processing JPEG-Compressed Images},
journal = {Proceedings of the 1997 International Conference on Image Processing (ICIP '97) },
year = {1997},
annote = {Abstract: We present techniques that allow the processing of an image in the "JPEG-compressed" domain. The goal is to reduce memory requirements while increasing speed by avoiding decompression and space domain operations. An effort is made to implement the minimum number of JPEG basic operations. Techniques are presented for scaling, previewing, rotating, mirroring, cropping, recompressing, and segmenting JPEG-compressed data.},
bibdate = {Friday, October 15, 1999 at 15:22:48 (MEST)},
submitter = {Johan Garcia}
}
@article{Kohler99,
author = {E Kohler and F Kaashoek and D R Montgomery},
title = {A Readable {TCP} in the Prolac Protocol Language},
journal = {Computer Communication Review},
year = {1999},
volume = {29},
number = {4},
pages = {15-25},
month = {October},
annote = {(ACM) The paper introduces the Prolac Protocol language that can be used to describe a protocol at a higher level of abstraction than C code. This representation is then used to genererate C code (like yacc). The C code is then compiled and inserted as module into the Linux kernel. The authors have implemented TCP with Prolac, and their measurements on Prolac generated TCP show that the additional overhead of using prolac is small (<20%). The authors stresses the point that a Prolac representation is easily extendable, making this useful as a tool for us. Maybe a partially reliable UDP could be easily built with this? Their homepage is at MIT},
url = {papers/Kohler99_ProlacTCP.pdf},
bibdate = {Wednesday, October 20, 1999 at 23:59:57 (MEST)},
submitter = {Johan Garcia}
}
@article{Tan99,
author = {Wai-tian Tan and Avideh Zakhor},
title = {Real-Time Internet Video Using Error Resilient Scalable Compression and {TCP}-Friendly Transport Protocol },
journal = {IEEE Transactions on Multimedia},
year = {1999},
volume = {1},
number = {2},
pages = {172-186},
month = {June},
annote = {This is an extended version of their conference paper. Contains a disussion of their 3D suband coding system and A TCP friendly transport protocol. This protocol lowers its transmission speed when congestion is detected, but not as abruptly as TCP. It is shown to cooperate gracefully with standard TCP. The use of FEC is for protecting the video streams is discussed. Reports from actual wide-area experiments is included. Nicely written.},
url = {papers/Tan99_TCPFriendly_ErrorResilient_Video.pdf},
bibdate = {Friday, October 22, 1999 at 10:00:00 (MEST)},
submitter = {Johan Garcia}
}
@article{Youn99,
author = {Jeongnam Youn and Ming-Ting Sun and Chia-Wen Lin },
title = {Motion Vector Refinement for High-Performance Transcoding},
journal = {IEEE Transactions on Multimedia},
year = {1999},
volume = {1},
number = {1},
pages = {30-40},
annote = {The paper is not so interesting in itself, but it contains several references (9-15) to papers discussing DCT coefficint rescaling / requantization. },
url = {papers/Youn99_video_transcoding.pdf},
bibdate = {Friday, October 22, 1999 at 10:26:17 (MEST)},
submitter = {Johan Garcia}
}
@article{Chang99,
author = {R.-S. Chang and C.-D. Wang},
title = {Improved WWW Multimedia Transmission in {HTTP/TCP} over {ATM} networks},
journal = {IEEE Transactions on Multimedia},
year = {1999},
volume = {1},
number = {3},
pages = {278-290},
month = {September},
annote = {This paper proposes the use of a (composite) partially reliable TCP to be used over ATM networks. The idea to use the data type field in the header to indicate that the packet is lossable (i.e. image, video or audio). This information is put there by the server, thus requiring mdifications to both the server an client stacks. The lossable indication is also appended to the end of the packet, to be used in case the ATM cell containing the TCP header is lost. The loss of one ATM cell will hence not trigger a TCP resend of the entire packet. No discussion on the agressiveness is provided. Some examples of how standand GIF and JPEG images may look after being transported over such a link is also provided, although this section lacks a lot of detail .},
url = {papers/Chang99_WWW_TCP_ATM.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, October 22, 1999 at 10:59:41 (MEST)}
}
@article{Fladenmuller99,
author = {Anne Fladenmuller, Ranil De Silva},
title = {The effect of Mobile {IP} handoffs on the performance of {TCP}},
journal = {Mobile Networks and Applications (MONET)},
year = {1999},
volume = {4},
number = {2},
pages = {131 - 135},
month = {May},
annote = {About the effects of mobile routing handoffs on TCP. Data can be lost when a handoff occurs. TCP interprets data loss as a congestion problem. A slow handoff has negative effects on TCP performance: long delays before retransmission (due to the exponential backoff algorithm) and throughput is reduced (due to the slow start algorithm). In order to avoid slow start Mobile IP or TCP must be modified. There are however no means to prevent timeouts and slow start from occurring. The authors expect the problem to become more significant in the future; modifications to both Mobile IP and TCP will be required.},
url = {papers/p131-fladenmuller.pdf},
submitter = {Annika Wennstrom},
bibdate = {Tuesday, November 02, 1999 at 09:25:03 (MET)}
}
@article{Jacobs98,
author = {S. Jacobs and A. Eleftheriadis},
title = {Streaming Video using Dynamic Rate Shaping and {TCP} Congestion control},
journal = {http://www.ee.columbia.edu/~sej/publications/papers/jvcir.ps.gz},
year = {1998},
month = {January},
annote = {The paper presents a technique for streaming real time video on today's Internet in a "TCP-friendly" way. The idea is to filter the encoded video signal so it is adapted to the network's available bandwidth. This adaptation capability is coupled with the use of a semi-reliable protocol that uses the TCP congestion window to pace the delivery of data into the network.},
submitter = {Katarina Asplund},
bibdate = {Tuesday, November 02, 1999 at 14:00:29 (MET)}
}
@article{Padhye98a,
author = {J. Padhye and J. Kurose and D. Towsley and R. Koodli},
title = {A Model Based {TCP}-friendly Rate Control Protocol},
journal = {UMass-CMPSCI Technical Report TR 98-04},
year = {1998},
month = {October},
annote = {This paper presents a protocol that is meant to be a first step towards developing a comprehensive (and TCP-friendly) protocol for congestion control for time-sensitive multimedia data streams. The model proposed in [Padhye98] is used as a base for the protocol.},
url = {ftp://gaia.cs.umass.edu/pub/Padhye98-tcp-friendly-TR.ps.gz},
bibdate = {Tuesday, November 02, 1999 at 16:09:06 (MET)},
submitter = {Katarina Asplund}
}
@article{Hamdi99,
author = {M. Hamdi and O. Verscheure and J-P. Hubaux and I. Dalgic and P. Wang},
title = {Voice Service Interworking for PSTN and IP Networks},
journal = {IEEE Communications},
year = {1999},
volume = {37},
number = {5},
pages = {104-111},
month = {May},
annote = {An overview of the main technical problems to be addressed for the provision of interoperable services between IP telephony and the PSTN.},
bibdate = {Tuesday, November 02, 1999 at 16:20:10 (MET)},
submitter = {Katarina Asplund}
}
@article{Schneyer99a,
author = {Sean K. Schneyer},
title = {The Effects of {PRTP} on the Congestion Control Mechanisms of {TCP}},
journal = {Master's Thesis 1999:05, Karlstad University},
year = {1999},
month = {June},
annote = {This thesis focuses on the interaction of PRTP with TCP. Specifically, this thesis presents the results of a simulation study conducted to determine the consequences of the changes made to TCP's congestion control mechanisms in order to implement PRTP. In addition, a significant amount of background material is presented in order to allow the reader to fully appreciate the results.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, November 08, 1999 at 15:13:44 (MET)}
}
@article{Liew99,
author = {J.S.A Liew and C.W. Ang and K.F. Ho},
title = {Improving {TCP} Performance Over Multi-Slot {GSM}},
journal = {Proceedings of ICC'99},
year = {1999},
month = {June},
annote = {This paper describes experiments performed with a mobile testbed using eight mobile handsets multiplexed to get a channel similar to that of multi-timeslot HSCD and to some extent GPRS. The snoop protocol has been implemented in this testbed (they ran in transparent mode?), and the results show an improvment over TCP of about four times for BER of about 10-5. (But the goodput is still way too low, ca 0.1) . The paper contains numerous inconsistencies (Fig 5 and the text results directly above does not match), omissions of necessary details, uncertainties (IMUX-induced delay not discussed), strange calculations (see (1), why use asynchronous start-stop bit over the constreined link, surely an RS-232 interface cant be constraining the bandwidth?) The work was done in Singapore in supported by Ericsson Radio AB in Sweden.},
url = {papers/Liew99_TCP_Multislot_GSM.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, November 23, 1999 at 11:11:08 (MET)}
}
@article{Landgre99,
author = {Jens Landgre and Katrin Nilsson},
title = {Evaluation of Proxy vs Plug-in and Experimental Implementations for {PRTP}},
journal = {Bachelor's Project 1999:15, Karlstad University},
year = {1999},
month = {June},
annote = {This report gives a background on proxies and plugins and analyze their applicability for an image recoding application. The conclusion is that a proxy is the most suitable solution for the image recoding problem associated with the use of PRTP on the transport layer. Further, the document surveys the capabilities and limitations of a number of proxies. The conclusion of this survey is that the most suitable solution would be to use a proxy framework called WBI. However, this solutions still has some drawbacks and the report proposes the development of a new proxy. In addition to the new proxy (SEWP - Simple Extendable Web Proxy) an implementation based on WBI was developed. An overview of the SEWP and WBI implementations are given.}
}
@article{Persson99,
author = {Hannes Persson and Robin Staxhammar},
title = {Anv\"{a}ndning av datakompression och felr\"{a}ttning i en komponent f\"{o}r hantering av dataf\"{o}rluster \"{o}ver en partiellt p{\aa}litlig kanal},
journal = {Bachelor's Project 1999:18, Karlstad University},
year = {1999},
month = {June},
annote = { This report describes the development of a component that uses data compression and forward error correction to recover from text data losses over a partially reliable channel. The authors' work has included testing and analyzing different compression and forward error correction algorithms, in the process of selecting one of each kind. The two selected algorithms have then been adapted and integrated into a working component.},
submitter = {Katarina Asplund},
bibdate = {Monday, November 22, 1999}
}
@article{Brunstrom99,
author = {Anna Brunstrom},
title = {{Analysis and Implementation of a Partially Reliable Transport Protocol for Multimedia Applications: Project Specification}},
journal = {Project Specification, University of Karlstad},
year = {1998},
month = {Dec}
}
@article{Brunstrom99a,
author = {Anna Brunstrom},
title = {Analysis and Implementation of a Partially Reliable Transport Protocol for Multimedia Applications: Project Report, Year 1},
journal = {Project Report, University of Karlstad},
year = {1999},
month = {Dec}
}
@article{Parsa99,
author = {C. Parsa and J.J. Garcia-Luna-Aceves},
title = {Improving {TCP} Congestion Control Over Internets with Heterogeneous Transmission Media},
journal = {Proc. IEEE ICNP 99: 7th International Conference on Network Protocols},
year = {1999},
month = {November},
annote = {TCP Santa Cruz is better suited for today's Internet than TCP Reno and Tahoe. Simulation experiments show that TCP Santa Cruz achieves higher throughput, smaller delays, and smaller delay variances than TCP Reno and Tahoe. Congestion control: Congestion is detected earlier than in Reno and Tahoe and the direction of congestion is identified (forward or reverse link). Error recovery: Lost packets are retransmitted faster and the retransmission of correctly received packets is prevented. The paper also describes how the improvements to the congestion control and error recovery mechanisms are implemented.},
url = {papers/chris.icnp99.ps.gz},
submitter = {Annika Wennstrom},
bibdate = {Thursday, December 02, 1999 at 15:08:03 (MET)}
}
@article{Iren99,
author = {Sami Iren and Paul D. Amer and Philip T. Conrad},
title = {The Transport Layer: Tutorial and Survey},
journal = {ACM Computing Surveys},
year = {1999},
pages = {51},
month = {June},
annote = {This paper is a tutorial on the transport layer. The authors differentiates between the service provided by the transport layer and the details of how a transport sender and a transport receiver cooperate to provide that service, i.e. the protocol. In the last section of the paper, eleven transport protocols are presented in terms of the services they offer and how (i.e. the protocol).},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, December 02, 1999 at 20:28:27 (MET)}
}
@article{Ludwig99a,
author = {Reiner Ludwig and Bela Rathonyi},
title = {Link Layer Enhancements for {TCP/IP} over {GSM}},
journal = {IEEE INFOCOM},
year = {1999},
annote = {The paper provide a good introduction to data support in GSM. It reports measurements on the round trip latency excluding transmission delay of about 410 ms for GSM. The paper also suggests the use of Quickstart-PPP. which will allow the subcomponents of PPP (LCP, PAP/CHAP and IPCP) to be transmitted in paralell istead of sequentially, yielding improved setup time for high latency links. They also disuss the possibility of the Link layer to snoop into the protocol identifcation field in IP and using either I-mode or UI-mode of RLP. UI-mode is an unreliable RLP mode that exists but which isnt currently implemented bu most current GSM equipment. They call this a 'flow-adaptive' link layer. Well written paper.},
url = {papers/Ludwig99a_Link_enhancments-TCP-GSM.pdf},
bibdate = {Friday, December 03, 1999 at 08:41:00 (MET)},
submitter = {Johan Garcia}
}
@article{Ludwig99b,
author = {Reiner Ludwig and Bela Rathonyi and Almudena Konrad and Kimberly Oden and Anthony Joseph},
title = {Multi-layer Tracing of {TCP} over a Reliable Wireless Link },
journal = {ACM SIGMETRICS },
year = {1999},
annote = {This paper provides an extensive overview of TCP works over GSM. They use a trace-based system, i.e. they collect raw traces of RLP and TCP and then postprocessed these. They ported a RLP implementation to UNIX to modify it to output trace info, and used tcpdump and tcpstat. The results are that practically no competetive error correction is performed by RLP and TCP, they cooperate well. Problems were RLP resets that occur at very low signal strength, reset is activated after 6 RLP retransmissions, which is too low. Another problem is too large queues for the low bitrate, leading to too high retransmission timeout times. Well written and strongly suggested reading. },
url = {papers/Ludwig99a_Tracing_TCP_GSM.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, December 03, 1999 at 08:51:53 (MET)}
}
@article{Ludwig99,
author = {Reiner Ludwig and Almudena Konrad and Anthony D. Joseph},
title = {Optimizing the End-to-End Performance of Reliable Flows over Wireless Links},
journal = {Fifth Annual International Conference on Mobile Computing and Networks (MobiCom '99), Seattle},
year = {1999},
month = {August},
annote = {This paper provides some details on the error distribution on the RLP level. The authors suggest that improved througput can be achieved by increasing the RLP frame size. They also argue against the use of pure end-to-end retransmissions only, since this will inevitably be more innefficient. },
url = {papers/Ludwig99_end-end_perf_over_wireless.pdf },
bibdate = {Friday, December 03, 1999 at 08:55:45 (MET)},
submitter = {Johan Garcia}
}
@article{Allman99,
author = {Mark Allman, Aaron Falk},
title = {On the Effective Evaluation of {TCP}},
journal = {ACM Computer Communication Review},
year = {1999},
volume = {5},
number = {29},
month = {October},
annote = {The paper lists questions that TCP researchers should try to answer. Recommendations are given on how to structure TCP testing: choice of TCP implementation, implementation details to look for, how to evaluate TCP (simulation, emulation or live Internet tests), collecting and analyzing data, etc.},
url = {papers/tcp-evaluation.ps.gz},
bibdate = {Friday, December 03, 1999 at 08:56:20 (MET)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Savage99,
author = {Stefan Savage and Neal Cardwell and David Wetherall and Tom Anderson},
title = {{TCP} Congestion Contorl with a Misbehaving Receiver},
journal = {Computer Communication Review},
year = {1999},
volume = {29},
number = {5},
month = {October},
annote = {Teh paper describes three attacks that a misbehaiving receiver can perform in order to convince a sender to send at a to hig rate, thus gaining an unfairly large portion of the available bandwidth. The attacks are: - ACK Division, Where the receiver sends many acks for subsets of each received segment, increacsing the congestion window with each divised ACK.
- DupACK spoofing, Sends a large number of DupACKs, even though a segment has not been lost. Each DupACK increases the window.
- Optimistic ACKing. ACKs segments that has not been received. This causes a faster slowstart, but creates problems if the segment is actually lost.
Each attack has been implemented and tested against several operating systems. Possibly DupACKing could be useful for speeding up slowstart over high delay, low bandwidth links? },
url = {papers/Savage99_TCP_misbehaving_rec.pdf},
bibdate = {Tuesday, December 07, 1999 at 10:25:52 (MET)},
submitter = {Johan Garcia}
}
@article{Wallace91,
author = {G. K. Wallace},
title = {The {JPEG} Still Picture Compression Standard},
journal = {Communications of the ACM, CACM},
year = {1991},
volume = {34},
number = {4},
pages = {30--44},
annote = { A short and very well written overview of the JPEG image compression standard},
url = {papers/Wallace91_JPEG_overview.ps.gz},
publisher = {ACM Press},
submitter = {Johan Garcia},
address = {New York, NY},
bibdate = {Monday, December 13, 1999 at 10:25:52 (MET)}
}
@book{Pennebaker93,
author = {William B. Pennebaker and Joan L. Mitchell},
title = {{JPEG} Still Image Data Compression Standard},
publisher = {Van Nostrand Reinhold},
year = {1993},
address = {New York, NY, USA},
annote = { A book that gives an in-depth description fo the JPEG standard, and also includes part 1 & 2 of the standard.},
bibdate = {Monday, December 13, 1999 at 11:25:52 (MET)},
submitter = {Johan Garcia}
}
@article{Chaskar99,
author = {H. M. Chaskar and T. V. Lakshman and U. Madhow},
title = {{TCP} Over Wireless with Link Level Error Control: Analysis and Design Methodology},
journal = {IEEE/ACM Transactions on Networking},
year = {1999},
volume = {7},
number = {5},
pages = {605 - 615},
month = {October},
annote = {This paper presents an approximate analysis, validated by computer simulations, for TCP performance over wireless links. The authors have shown that if TCP sees a probability of end-to-end loss above the inverse square of the bandwidth-delay product, then its throughput will deteriorate significantly. To keep the level of loss seen by TCP below this threshold, the authors suggest link-layer retransmissions and properly sized buffers},
submitter = {Katarina Asplund},
bibdate = {Monday, December 20, 1999 at 11:35:55 (MET)}
}
@article{Bolot98a,
author = {J-C. Bolot and T. Turletti},
title = {Experience with Control Mechanisms for Packet Video in the Internet},
journal = {Computer Communication Review},
year = {1998},
month = {January},
annote = {The authors give an overview of their experience with rate and error control mechanisms for vidoconferencing software over the past several years.},
bibdate = {Monday, December 20, 1999 at 12:24:17 (MET)},
submitter = {Katarina Asplund}
}
@article{Feng99,
author = {W-C. Feng and D. D. Kandlur and D. Saha and K. G. Shin},
title = {Adaptive Packet Marking for Maintaining End-to-End Throughput in a Differentiated-Services Internet},
journal = {IEEE/ACM Transactions on Networking},
year = {1999},
volume = {7},
number = {5},
pages = {685 - 697},
month = {October},
annote = {This paper examines the use of adaptive priority marking for providing soft bandwidth guarantees in a differentiated services Internet. In this scheme, only two traffic types are supported, priority and best-effort. Example: A TCP connection has a target bandwidth of 4Mb/s. If the observed throughput is greater or equal to the target bandwidth, all packets are marked as best-effort. If the observed throughput is less than the target bandwidth, packets are marked as priority packets with some probability. The probability increases as the quota observed throughput/target throughput gets smaller.},
submitter = {Katarina Asplund},
bibdate = {Monday, December 20, 1999 at 12:52:01 (MET)}
}
@article{Vivo99,
author = {Marco de Vivo and Eddy Carrasco and Germinal Isern and Gabriela O. de Vivio},
title = {A Review of Port Scanning Techniques},
journal = {Computer Communication Review},
year = {1999},
volume = {29},
number = {2},
month = {April},
annote = {This security paper describes a variety of port scanning techniques such as SYN, indirect, stealth, decoy, coordinated, ident, proxy and UDP scanning. Ping scanning, security scanners and Stack fingerprinting is also mentioned.},
url = {papers/vivo99_portscanning.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, December 21, 1999 at 08:25:38 (MET)}
}
@article{Alanko94,
author = {Timo Alanko and Markku Kojo and Heimo Laamanen and Mika Liljeberg and Marko Moilanen and Kimmo Raatikainen},
title = {Measured Performance of Data Transmission Over Cellular Telephone Networks},
journal = {ACM Computer Communication Review},
year = {1994},
volume = {24},
number = {5},
annote = {Presents performance measurements of data transmission over a GSM and a NMT network. Measurements for three different scenarios are presented: establishment of wireless dial-up connection, request-reply communication and bulk data transfer.},
url = {papers/Alanko94.ps.gz},
bibdate = {Wednesday, February 02, 2000 at 18:44:14 (MET)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Rojas99,
author = {L. Rojas Cardenas and L. Dairaine and P. Sénac and M. Diaz},
title = {Error Recovery Mechanisms Based on Retransmissions for Video Coded with Motion Compensation Techniques},
journal = {, Packet Video Workshop},
year = {1999},
pages = {11},
annote = {This paper shows that error recovery mechanisms based on retransmissions can be used to improve the performance of transporting interactive video. More specifically, they argue that a partially ordered and partially reliable transport protocol will leverage the performance of transporting MPEG I coded video sequences with respect to two aspects. Firstly, it minimizes the number of damaged pictures. Secondly, it improves the continuity of the picture stream (less jitter). The mean inter-packet time for sending a MPEG I video sequence using TCP was 0.06 s, while POC, the partially ordered and partially reliable transport protocol they used, only had an inter-packet time of 0,02 s. The POC transport protocol comprises three modules: - a flow management module for handling congestion and provide for a graceful degradation of QoS, - an order management module which handles the order of arriving picture elements at the receiving side (e.g. GOBs and slices). The order is governed by a modified Petri Net (Time Stream Petri Net).},
url = {Karl-Johan has a paper copy},
bibdate = {Wednesday, February 09, 2000 at 09:36:22 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Rojas99a,
author = {L. Rojas and E. Chaput and L. Dairaine and P. Senac and M. Diaz},
title = {Transport of video over partial order connections},
journal = {Computer Networks},
year = {1999},
volume = {31},
number = {7},
pages = {709-725},
month = {April},
annote = {This paper is almost identical to the paper "Error Recovery Mechanisms Based on Retransmissions for Video Coded with Motion Compensation Techniques. They argue that neither a connection oriented service nor a connectionless dito is appropriate for transporting video. Instead, a compromise is needed, POC (Partial Order and partial reliable Connection). This is demonstrated by an experminet where an MPEG video stream is transported using UPD, TCP, and POC. Using UDP on a connection experiencing 5.56% packet loss resulted in a GOP (Group Of Pictures) with 15 pictures lost. Using the same connection for POC resulted in only 2 packets lost. When it comes to continuity, POC is on the average a factor of three better than TCP.},
url = {Karl-Johan has a paper copy},
bibdate = {Wednesday, February 09, 2000 at 09:49:04 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Chassot96,
author = {C. Chassot and M. Diaz and A. Lozes},
title = {From the partial order connection concept to partial order multimedia transport connections},
journal = {Journal for High Speed Networks},
year = {1996},
volume = {5},
number = {2},
pages = {181-191},
month = {January},
annote = {This article is probably one of the earliest written articles about POC (Partial Order Connection (sic)). It introduces the partially ordered and partially reliable service as a generalization of the connection less and connection oriented protocols. It also introduces Time Stream Petri Net (TSDN) as an appropriate formal model for describing the order and loss constraints inherent to an application. One interesting aspect of POC introduced in this article is the way of partitioning a connection into several logical connections each providing a different quality of service. The reliability can be measured per logical connection, or per group of conmnections.},
url = {Karl-Johan has a paper copy},
bibdate = {Wednesday, February 09, 2000 at 12:20:37 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Garcia00,
author = {Johan Garcia and Anna Brunstrom},
title = {Efficient Image Transfer for Wireless Networks},
journal = {2nd International Conference on Advanced Communication Technology (ICACT2000), Muju, South Korea},
year = {2000},
pages = {305-310},
month = feb,
annote = {This report describes an approach to speed up recoding of baseline JPEG images. Recoding is commonly done in proxy systems in order to increase the degree of compression before transmitting data over a slow link such as a mobile phone connection. The described approach performs the recoding in the intermediate DCT coefficient format instead of decoding the image to a pixel representation before applying the increased compression. Measurements show that other recoders require between 50 and 300\% more time to recode than the proposed efficient recoder. For typical Web images and an efficient but pixel-based recoder the additional time required is approximately 100\% more than the time required for the proposed recoder. Memory requirements were also measured, and the results show that the suggested recoder is memory efficient for typical images.},
url = {papers/icact2000final.ps.gz},
submitter = {Johan Garcia},
bibdate = {Thursday, February 24, 2000 at 13:52:03 (MET)}
}
@misc{RFC2757,
author = {G. Montenegro and S. Dawkins and M. Kojo and V. Magret and N. Vaidya},
title = {{RFC 2757}: Long Thin Networks},
year = {2000},
month = {January},
annote = {Contains a nice overview of the different techniques that can be used to enhance TCPs performance over low bandwidth, large RTT networks (LTNs). Techniques discussed include TCP modifications such as larger initial window, SACK, Ack spacing and Split-TCP techniques (Performance enhancing proxies).},
url = {ftp://ftp.math.utah.edu/pub/rfc/rfc2757.txt},
bibdate = {Sunday, March 05, 2000 at 14:49:33 (MET)},
submitter = {Johan Garcia}
}
@article{Budagavi00,
author = {Madhukar Budagavi and Wendi Rabiner Heinzelman and Jennifer Webb and Raj Talluri},
title = {Wireless MPEG-4 Video Communication on DSP Chips},
journal = {IEEE Signal Processing Magazine},
year = {2000},
volume = {17},
number = {1},
pages = {36-53},
month = {January},
annote = {The contains a voluminous but readable high-level overview of MPEG-4 simple profile and its error resiliency tools such as Resynchronization Markers, Data Partitioning, Reversible Variable length codes, Header extension code, Adaptive Intra Refresh and error concealment. Implementation of these in DSPs is also discussed. Further, the H.223 multiplexing standard and its channel error protection adaption layer is discussed. Experiments using unequal error protection run over a GSM channel simulator (not using RLP) is presented, though no GSM details are given. The conclusions are that unequal error protection is slightly better (1dB PSNR) than equal EP. They suggest retransmissions to ensure correct delivery of the first frame, so in sense partial reliability is used. },
url = {Mag in library},
submitter = {Johan Garcia},
bibdate = {Wednesday, March 08, 2000 at 11:13:49 (MET)}
}
@article{Barakat00,
author = {Chadi Barakat and Eitan Altman and Walid Dabbous},
title = {On {TCP} Performance in a Heterogenous Network: A Survey},
journal = {IEEE Communications Mag.},
year = {2000},
volume = {38},
number = {1},
pages = {40-46},
month = {January},
annote = {A relatively short highlevel survey of problems that can occur over connections with non-typical characterisctics. The problem areas considered are paths with large Bandwidth-delay products , large round-trip time, noncongestion losses and assymetric bandwidth. Contains an ok overview of the problems, but touches a lot of things only briefly and provide no really new insights.},
submitter = {Johan Garcia},
bibdate = {Wednesday, March 08, 2000 at 22:50:06 (MET)}
}
@article{Kim00,
author = {Il-Min Kim and Hyung-Myung Kim},
title = {Efficient Power Management Schemes for Image Transmission in CDMA},
journal = {2nd International Conference on Advanced Communicaton Technlogy (ICACT2000)},
year = {2000},
pages = {101-106},
month = {February},
annote = {This paper presents two power managment schemes that can be used to optimize the performance of video transfer over CDMA transmission. The idea is to control the transmission power so that the GEP (Group-Of-Blocks Error Prob) is constant independent of the number of bits in a GOB. Regular power control keeps the BER nearly constant, but for video this will lead to larger GOBs aving larger GEP. This is clearly undesirable since larger GOBs induce larger distorsion when lost. The paper also argues that the power control should take into account if CBR or VBR encoding is performed. With CBR the distorsion influence of a GOB is not decided by its bitsize only, but also on the instantaneous video activity and the traffic shaping function used. The paper proposes a hardware addition that can calculate the distorsion influence of each GOB and adjust the transmission power and hence the error probability accordingly. This paper is interesting since it shows the gains of letting higher layers control some characteristics of lower. Perhaps a partially reliable transport protocol should be able to influence the power control?},
url = {Proceeding (Johan)},
bibdate = {Wednesday, March 08, 2000 at 23:07:27 (MET)},
submitter = {Johan Garcia}
}
@article{Diaz94,
author = {M. Diaz and A. Lopez and C. Chassot and P. Amer},
title = {Partial order connections: a new concept for high speed and multimedia services and protocols},
journal = {Annales des telecoms},
year = {1994},
volume = {49},
number = {5-6},
annote = {This paper introduces the concept partial reliability as an extension of connection-oriented and connection-less protocols. Partial reliability is seen as the "inner part" of a RO-graph (Reliability-Order graph). Metrics are defined, which makes it possible to qunatify and compare different partial order services. The two metrics introduced are: uncertainty and entropy. Both these metrics originates from the concept of the number of linear extensions. The smaller the number of acceptable linear extensions is the larger is the uncertainty and the smaller is the entropy. As a matter of fact, the sum of the uncertainty and the entropy is shown to be constant.},
url = {Karl-Johan Grinnemo has a copy of this article},
bibdate = {Thursday, March 09, 2000 at 08:30:10 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Chassot95,
author = {C. Chassot and M. Fournier and M. Diaz and A. Lozes},
title = {Service Definition of a Multimedia Partial Order Connection},
journal = {COST 237 Workshop on Multimedia Teleservices, Copenhagen, Denmark},
year = {1995},
month = {November},
annote = {This paper differs from most of the other papers published by POC World. It gives an insight in how a POC (Partial Order Connecton) works in practice. The paper ends with an in-depth discussion of how the service API has been realized.},
url = {Karl-Johan Grinnemo has a copy of this paper.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, March 09, 2000 at 11:38:43 (MET)}
}
@article{Diaz95,
author = {M. Diaz and K. Drira and A. Lozes and C. Chassot},
title = {On the Definition and Representation of the Quality of Service for Multimedia Systems},
journal = {6th IFIP 6th IFIP International Conference on High Performance Net.working (HPN'95)},
year = {1995},
pages = {12},
month = {September},
annote = {The main contribution of this paper is the formalization of the concept QoS (Quality of Service). QoS is seen as a subset of E x E, E being the set of transmitted objects. A service is said to uphold a certain QoS if it is ordering, relaibility, and timing conformant with this QoS.},
url = {Karl-Johan Grinnemo has a copy of this paper.},
bibdate = {Thursday, March 09, 2000 at 12:01:25 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Fournier97,
author = {M. Fournier and C. Chassot and M. Diaz and A. Lozes},
title = {Performance Evaluations of Partial Order Connections},
journal = {Seventh IFIP Conference of High Performance Networking (HPN '97)},
year = {1997},
pages = {14},
month = {May},
annote = {This paper presents the simulation results obtained when simulating POC in OPNET. The simulation focuses on POC's impact on sending/receiving buffers and transit delay. Concerning order, it is, not surprisingly shown, that the less the order constraints are, the less buffers are needed and the lower the transit delay is. When the network loss rate is less than the user required reliability, reliability at the transport level is not useful since the QoS is uphold by the network itself. Of particular interest is the transit delay improvements gained when total reliability is not required. },
url = {Karl-Johan Grinnemo has a copy of this paper},
bibdate = {Sunday, March 12, 2000 at 11:43:51 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Cardenas98,
author = {L. Cardenas and P. Senac and L. Dairaine and M. Diaz},
title = {Temporal Partial Order and Partial Reliability Service for Distributed Multimedia Applications},
journal = {MMM'98 (MultiMedia Modelling), Lausanne},
year = {1998},
pages = {11},
month = {October},
annote = {Apart from discussing the POC service in terms of ordering and reliability compliance, TPOC, Temporal POC, is introduced, which is POC with timing considerations added. The performance and applicability of TPOC for multimedia applications is demonstrated by sending an MPEG video sequence using TPOC.},
url = {Karl-Johan Grinnemo has a copy of this paper},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, March 12, 2000 at 12:38:37 (MET)}
}
@article{Amer93,
author = {P. Amer and T. Connolly and C. Chassot and M. Diaz},
title = {Partial Order Transport Service for Multimedia Applications: Reliable Service},
journal = {2nd High Performance Distributed Computing Conference, Spokane, Washington},
year = {1993},
month = {July},
annote = {This paper introduces the POC (Partial Order Connection) protocol. A metric based on the number of linear extensions of a partial order in the presence of no lost objects is proposed to quantify different partial ordered services. In the section "Implementation considerations" gives the authors a nice overview of how they communicate the Petri Net, holding the partial order, from the sender to the receiver. This paper only considers a reliable partial order service. In a paper written in connection to this paper, "Partial Order Transport Service for Multimedia Applications: Unreliable Service", an unreliable version of POC is presented.},
url = {Karl-Johan Grinnemo has a copy of this paper.},
bibdate = {Monday, March 13, 2000 at 10:07:27 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Amer93b,
author = {P. Amer and T. Connolly and C. Chassot and M. Diaz},
title = {Partial Order Transport Service for Multimedia Applications: Unrelaible Service},
journal = {International Networking Conference (INET'93), San Fransisco, CA},
year = {1993},
month = {August},
annote = {This paper discusses un unreliable version of POC. A metric based on the number of linear extensions accepted by the partial ordered service is presented. This metric is thought to make it possible comparing different partially ordered services. The POC research group has specified POC in Estelle, an ISO international standard for formally describe protocols. Four reliability classes are introduced: BART-NL, BART-L, NBART-L, BART-NL, making it possible to tag each object sent with an apporpriate reliability.},
url = {Karl-Johan Grinnemo has a copy of this paper.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, March 13, 2000 at 10:41:55 (MET)}
}
@misc{rfc2474,
author="K. Nichols and S. Blake and F. Baker and D. Black",
title={Definition of the Differentiated Services Field ({DS} Field) in the {IPv4} and {IPv6} Headers},
series="Request for Comments",
number="2474",
howpublished="RFC 2474 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1998,
month=dec,
url="http://www.ietf.org/rfc/rfc2474.txt",
}
@misc{rfc2475,
author="S. Blake and D. Black and M. Carlson and E. Davies and Z. Wang and W. Weiss",
title = {An Architecture for Differentiated Services},
series="Request for Comments",
number="2475",
howpublished="RFC 2475 (Informational)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1998,
month=dec,
url="http://www.ietf.org/rfc/rfc2475.txt",
}
@article{Lan99,
author = {Tse-Hua Lan and Ahmed H. Tewfik},
title = {JPEG transCompressor and video networks},
journal = {Proc. of ACM Multimedia '99, Orlando, FL.},
year = {1999},
pages = {139 - 142 },
month = {October},
annote = {This paper proposes the use of MultiGrid Embedding (MGE) to pack the DCT coefficients. (kind like EZW) This is proposed to be used to create a embedded, scalable representation that also has slightly better compression performance than regular JPEG. Seems to state that progressive JPEG isnt scalable. },
url = {papers/Lan99_JPEGTransCompressor.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, March 17, 2000 at 10:53:58 (MET)}
}
@article{Karlsson98,
author = {P. Karlsson and {\AA}. Arvidsson},
title = {On {TCP/IP} Traffic Modeling},
journal = {Proc. Fourteenth Nordic Teletraffic Seminar},
year = {1998},
pages = {pp. 156-166},
annote = {Using a simulation model of TCP/IP over ATM, the authors show that models dealing only with a cell or packet arrival process tend to highly overestimate the required capacity in terms of bandwidth and buffer space. This implies that models for TCP/IP traffic must take into account the behavior of the protocols involved. In the conducted experiments, the stream of arrivals of new TCP connections were represented by a Poisson process. Each new connection was a file transfer session of files having various types of distributions, e.g. Pareto and Negative exponential. Between two nodes, A and B, connected by and ATM link, all traffic had to pass. The access links for senders connecting to node A and receivers at node B have unlimited bandwidth and no lateny, while the link between node A and B is both bandwidth constrained and has a latency.},
url = {On Ericsson's intranet. Karl-Johan has a copy of this article.},
bibdate = {Monday, March 20, 2000 at 10:14:15 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Arvidsson98,
author = {{\AA}. Arvidsson, P. Karlsson},
title = {On Traffic Models for {TCP/IP}},
journal = {Teletraffic Engineering in a Competitive World, Oxford, Mississippi, USA},
year = {1998},
pages = {457-466},
annote = {Two approaches to simulate TCP/IP traffic are: Fitting the parameters of a stochastic process to the characteristics of packet arrivals (link-level model), or building a model of user actions (user-level model). The authors of this paper argues that the latter approach gives a far more accurate description of TCP/IP traffic. Their argumentation is enforced by a real-world experiment which supports their conclusions. The ramifications of their conclusions are, among others: - Traditional link-level simulation has a tendency to highly underestimate the performance of TCP/IP traffic and overestimate the cosumption of buffers. - The large queuing problems anticipated from earlier works based on link-level models do not occur with user-level models. },
url = {Ericsson's intranet. Karl-Johan has a copy of this paper.},
bibdate = {Monday, March 20, 2000 at 10:16:43 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Claypool99,
author = {Mark Claypool and Jonathan Tanner},
title = {The Effects of Jitter on the Perceptual Quality of Video},
journal = {Proc. ACM Multimedia '99(Part 2)},
year = {1999},
pages = {115-118},
month = {October},
annote = {This paper briefly presents experiments aimed assesing the perception on video containing jitter. They have conducted experiments with 41 users that whas shown five different clips with each clip having five versions: original, low (1x trace) &high (3x trace) jitter, and low(8%) & high(22%) packet loss. The results mainly show that both low jitter or low packet loss gives low opinion scores, which are not much lowered when going to high jitter or losses. The study also shows that the temporal activity plays a minor role inte sensibility to jitter or losses, but their main results are that jitter is almost as important as packet loss. More details are available in a tech report.},
url = {papers/Claypool99_Jitter_Quality_video.pdf},
submitter = {Johan Garcia},
bibdate = {Monday, March 20, 2000 at 21:15:13 (MET)}
}
@article{Boyce98,
author = {Jill Boyce and Robert Gaglianello},
title = {Packet Loss Effects on MPEG Video Sent Over the Public Internet},
journal = {Proc. of ACM Multimedia '98},
year = {1998},
pages = {181-190},
month = {September},
annote = {This papers main contribution is the loss measurements they have made when performing video traffic transfer between Holmdel,NJ and New York, Austin & London. They sent video streams at 1Mbps and 384 Kbps using RTP/UDP. They made 100+ replications in order to capture the difference in behaiviour between different times of day. They used a somewhat strange setup by pacing the sender to send at the specified bitrate by delaying packets, a thing that would normally not be done. Loss rates for the 384 kbps stream was 3-9% and for the 1Mbps 5-13% on the average, with large short-term variations. They also report on the burstiness of packet losses and their results show that a majority of losses had short burst lengths. They also found a significant amount of out-of-order delivery ranging from 2 to 15 %. For two cases there were more packet out-of-order than lost. },
url = {papers/Boyce98_Packet_loss_MPEG.pdf},
submitter = {Johan Garcia},
bibdate = {Monday, March 20, 2000 at 23:49:05 (MET)}
}
@article{Bennett99,
author = {J. Bennett and C. Partridge and N. Shectman},
title = {Packet Reordering is Not Pathological Network Behavior},
journal = {IEEE/ACM Trans. on Networking},
year = {1999},
volume = {7},
pages = {789-798},
month = {December},
annote = {The authors have found that parallelism in Internet components and links is causing packet reordering under normal operation, and not are due to incorrect or malfunctioning network components. Furthermore, reordering appears to be substantially higher than previously reported. Reordering is a function of three properties: the existence of parallel links between nodes, configuration (hardware and software), and load on the nodes. It is also shown that reordering can cause serious performance problems, especially for TCP.},
bibdate = {Tuesday, March 21, 2000 at 18:49:57 (MET)},
submitter = {Katarina Asplund}
}
@article{Ahmed99,
author = {U. Ahmed and J. H. Salim},
title = {Performance Evaluation of Explicit Congestion Notification (ECN) in IP Networks},
journal = {http://www7.nortel.com:8080/CTL/ecnperf.pdf},
year = {1999},
month = {December},
annote = {This paper presents a performance study of the ECN mechanism in the TCP/IP protocol using a Linux implementation. The behavior of ECN for both bulk and transactional transfers are studied. The experiments show that there is improvement in throughput over standard TCP Reno in the case of bulk transfers and substantial improvement for transactional transfers.},
url = {http://www7.nortel.com:8080/CTL/ecnperf.pdf},
submitter = {Katarina Asplund},
bibdate = {Tuesday, March 21, 2000 at 21:50:54 (MET)}
}
@article{Karandikar00,
author = {S. Karandikar and S. Kalyanaraman and P. Bagal and B. Packer},
title = {{TCP} Rate Control},
journal = {ACM Computer Communications Review},
year = {2000},
month = {January},
annote = {This paper presents TCP Rate Control, a technique for transparently augmenting end-to-end TCP performance by controlling the sending rate of a TCP source. The "rate" of a TCP source is determined by its window size, the round trip time and the rate of acknowledgements. TCP rate control controls these aspects by modifying the receiver window in acks and by modulating the ack rate. A comparative study with RED and TCP-ECN is performed, which shows that TCP rate control avoids adverse performance effects due to packet losses, such as reduced goodput due to retransmission and timeout delays, or large spread in user goodputs.},
url = {http://www.packeteer.com/technology/PDF/tcprate.pdf},
bibdate = {Tuesday, March 21, 2000 at 22:42:08 (MET)},
submitter = {Katarina Asplund}
}
@article{Gringeri99,
author = {Steven Gringeri and Roman Egorov and Khaled Shuaib and Arianne Lewis and Bert Basch},
title = {Robust compression and transmission of MPEG-4 Video},
journal = {Proc ACM Multimedia '99, Orlando, FL},
year = {1999},
pages = {113-120},
month = {October},
annote = {This paper describes shortly describes the error resilience techniques incorporated into MPEG-4. It reports on performance experiments made in order to evaluate the cost and performance of these measures. The overhead is shown to be around 5 to 10 % for packet video sizes of 400 and 800 bits. One interesting result is that the PSNR is about 4% better for Frame Loss (i.e. one VOP/frame=ALF) than for fixed size packets. Also reports on the use of BCH FEC codes used to improve BER to minimisesubjective quality loss. Optimization FEC / residual BER is said to be future work.},
url = {papers/Gringeri99_Robust_MPEG4.pdf},
bibdate = {Wednesday, March 22, 2000 at 08:25:36 (MET)},
submitter = {Johan Garcia}
}
@misc{RFC2216,
author = {S. Shenker and J. Wroclawski},
title = {{RFC 2216}: Network Element Service Specification Template },
year = {1997},
month = {September},
annote = {This document defines a framework for specifying services provided by network elements, and available to applications, in an internetwork which offers multiple qualities of service.},
url = {http://www.ietf.org/rfc/rfc2216.txt},
submitter = {Katarina Asplund},
bibdate = {Monday, March 27, 2000 at 12:42:26 (MEST)}
}
@misc{RFC2215,
author = {S. Shenker and J. Wroclawski},
title = {{RFC 2215}: General Characterization Parameters for Integrated Service Network Elements},
year = {1997},
month = {September},
annote = {This memo defines a set of general control and characterization parameters for network elements supporting the IETF integrated services QoS control framework. General parameters are those with common, shared definitions across all QoS control services.},
url = {http://www.ietf.org/rfc/rfc2215.txt},
submitter = {Katarina Asplund},
bibdate = {Monday, March 27, 2000 at 12:45:47 (MEST)}
}
@misc{RFC2211,
title = {{RFC 2211: S}pecification of the Controlled-Load Network Element Service},
author = {J. Wroclawski},
month = {September},
year = {1997},
annote = {This memo specifies the network element behavior required to deliver Controlled-Load service in the Internet. Controlled-load service provides the client data flow with a quality of service closely approximating the QoS that same flow would receive from an unloaded network element, but uses capacity (admission) control to assure that this service is received even when the network element is overloaded.},
url = {http://www.ietf.org/rfc/rfc2211.txt},
submitter = {Katarina Asplund},
bibdate = {Monday, March 27, 2000 at 14:45:24 (MEST)}
}
@misc{RFC2212,
title = {{RFC 2212: S}pecification of Guaranteed Quality of Service},
author = {S. Shenker and C. Partridge and R. Guerin},
month = {September},
year = {1997},
annote = {This memo describes the network element behavior required to deliver a guaranteed service (guaranteed delay and bandwidth) in the Internet. Guaranteed service provides firm (mathematically provable) bounds on end-to-end datagram queueing delays. },
url = {http://www.ietf.org/rfc/rfc2212.txt},
bibdate = {Monday, March 27, 2000 at 14:48:51 (MEST)},
submitter = {Katarina Asplund}
}
@article{Balakrishnan99,
author = {Hari Balakrishnan and Randy H. Katz and Venkata N. Padmanbhan},
title = {The effects of asymmetry on {TCP} performance},
journal = {Mobile Networks and Applications},
year = {1999},
volume = {4},
number = {3},
pages = {219-241},
month = {October},
annote = {The effects of network asymmetry on end-to-end TCP performance is studied. They analyze TCP performance in such networks where the throughput achieved is not solely a function of the link and traffic characteristics in the direction of data transfer (the forward direction), but depends significantly on the reverse direction as well. The paper focuses on bandwidth and latency asymmetries, and proposes and evaluates several techniques to improve end-to-end performance. These include techniques to decrease the rate of acknowledgments on the constrained reverse channel (ack congestion control and ack filtering), techniques to reduce source burstiness when acknowledgments are infrequent (TCP sender adaptation), and algorithms at the reverse bottleneck router to schedule data and acks differently from FIFO (acks-first scheduling).},
url = {Journal in CS library},
bibdate = {Monday, March 27, 2000 at 17:40:11 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@misc{rfc2581,
author="M. Allman and V. Paxson and W. Stevens",
title={{TCP} Congestion Control},
series="Request for Comments",
number="2581",
howpublished="RFC 2581 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1999,
month=apr,
url="http://www.ietf.org/rfc/rfc2581.txt",
}
@misc{RFC2309,
author = {B. Braden et al},
title = {{RFC 2309}: Recommendations on Queue Management and Congestion Avoidance in the Internet},
year = {1998},
pages = {16},
month = {April},
annote = {This RFC recommends the use of active queue management techniques in routers in the Internet. The active queue management technique, RED (Random Early Detection), is proposed as the, for the present, best queueing technique to be used. In their concluding remarks, the authors urge the research community to increase their research efforts on unresponsive flows and congestion notification.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, April 03, 2000 at 10:56:27 (MEST)}
}
@article{Konrad99,
author = {A. Konrad},
title = {{Performance Study of the {GSM} Circuit-switched Data Channel}},
journal = {Research Project},
year = {1999},
month = {December},
annote = {Provides almost the same information as Ludwig99 and Ludwig99b. More detail on RLP.},
url = {papers/masters.ps.gz},
bibdate = {Mon Apr 10 11:39:05 CEST 2000},
submitter = {Annika Wennstrom}
}
@article{ETSI98a,
author = {ETSI},
title = {{Digital cellular telecommunications system (Phase 2+); Radio Link Protocol (RLP) for data and telematic services on the Mobile Station - Base Station System (MS - BSS) interface and the Base Station System - Mobile-services Switching Centre (BSS - MSC) interface (GSM 04.22 version 7.0.1 Release 1998)}},
journal = {ETSI},
year = {1999},
url = {papers/rlp_spec.pdf.gz},
bibdate = {Mon Apr 10 11:39:05 CEST 2000},
submitter = {Annika Wennstrom}
}
@inproceedings{Noble97,
author = {Noble, B. and Satyanarayanan, M. and Nguyen, G. and Katz, R.},
title = {Trace-Based Mobile Network Emulation},
booktitle = {Proceedings of ACM SIGCOMM},
year = {1997},
month = {September},
annote = {Thorough performance evaluation of a mobile computing system requires realistic and reproducible network characteristics. The authors have developed trace-based emulation as an extension to the NetBSD kernel. Trace-based emulation is described and evaluated (Wave-LAN). The methodology consists of trace collection, distillation of suitable parameters, and modulation (each host in the network replays the trace). The advantage is that a synthetic networking environment is provided. All traffic sent and received by a host is affected, not only a specific application.},
url = {papers/sigcomm97.ps.gz},
bibdate = {Mon Apr 10 15:03:04 CEST 2000},
submitter = {Annika Wennstrom}
}
@inproceedings{Bai99,
author = {Yong Bai and Andy T. Ogielski and Gang Wu},
title = {{{TCP} over {IS-707}}},
booktitle={Proceedings of the IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)},
year = {1999},
month = {September},
annote = {The performance of TCP Reno over IS-707 is examined through simulation. IS-707 is the radio link layer used for spread spectrum digital cellular systems. The link layer provides partial error recovery through a small number of retransmissions (how many?), then further retransmissions are left to higher layers. The number of idle frames sent after data is sent determine how often the link layer receiver checks its buffer for missing frames. By controlling the number of idle frames TCP retransmissions are avoided and TCP goodput is increased. The paper also describes how fading is modeled.},
url = {papers/pimrc99.pdf.gz},
bibdate = {Mon Apr 10 15:56:20 CEST 2000},
submitter = {Annika Wennstrom}
}
@article{Estrin99,
author = {Deborah Estrin and Mark Handley and John Heidemann and Steven McCanne and Ya Xu and Haobo Yu},
title = {{Network Visualization with the VINT Network Animator Nam}},
journal = {Technical Report 99-703, University of Southern California},
year = {1999},
month = {March},
annote = {The network animator, nam, provides packet-level animation and protocol-specific graphs. Input data can be taken from network simulators or live networks (some processing seems to be required first).},
url = {papers/Estrin99d.ps.gz},
submitter = {Annika Wennstrom}
}
@article{Paxson95,
author = {Paxson, V. and Floyd, S.},
title = {{Wide-Area Traffic: The Failure of Poisson Modeling}},
journal = {IEEE/ACM Transactions on Networking, Vol. 3 No. 3, pp. 226-244},
year = {1995},
month = {June},
annote = {Traces of wide-area TCP traffic is analyzed. Session, connection, and packet arrivals are analyzed for different applications such as FTP, TELNET, WWW. Wide-area traffic is much burstier than than Poisson models predict. Poisson-based modeling is only appropriate for modeling user session arrivals. The authors suggest further studies of self-similarity. Distributions are explained in the appendix. Suggested reading. (Skip the details)},
url = {papers/WAN-poisson.ps.Z},
submitter = {Annika Wennstrom},
bibdate = {Tue Apr 11 10:35:06 CEST 2000}
}
@article{Paxson97,
author = {Paxson, V. and Floyd, S.},
title = {{Why We Don't Know How To Simulate The Internet}},
journal = {Proceedings of the 1997 Winter Simulation Conference},
year = {1997},
month = {December},
annote = {It is hard to simulate Internet traffic since the Internet is heterogeneous and rapidly changing. Meaningful simulations must be both tractable and realistic. Trace-driven source-level simulation is recommended instead of the often used packet-level simulation. Source-level simulation (trace or abstract model driven) includes source behavior such as adaptive congestion control. Promising candidates of invariants are listed (aspect of behavior that holds in many environments). This paper and Paxson95 are both possible to understand without detailed knowledge statistics. Together they give an overview of traffic modeling.},
url = {papers/wsc97.ps.gz},
bibdate = {Tue Apr 11 11:01:55 CEST 2000},
submitter = {Annika Wennstrom}
}
@article{Chiu89,
author = {D. Chiu and R. Jain},
title = {Analysis of the Increase and Decrease Algorithms for Congestion Avoidance in Computer Networks},
journal = {Computer Networks and ISDN Systems},
year = {1989},
volume = {17},
pages = {14},
annote = {In this paper Dr. Chiu and Dr. Jain discusses congestion avoidance from a control theoretic viewpoint. The simplest feedback system, the binary feedback system, is analyzed, and criterias for selecting the parameters are mathematcally derived. It is also shown that a simple additive increase and multiplicative decrease algorithm satisfies sufficient conditions for convergence to an efficient and fair state regardless of the starting point of the network. I found this paper very interesting especially their formulation of such performance aspects as "efficiency", "fairness", "distributedness", and "convergence". At the end of the discussion they briefly treat non-linear control mechanisms. Their conslusion is though that non-linear systems are very sensitive to system parameters, and should only be used when linear-systems are found to be insufficient.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, April 12, 2000 at 08:54:17 (MEST)}
}
@article{Jain84,
author = {R. Jain and D. Chiu and W. Hawe},
title = {A Quantitative Measure of Fairness and Discrimination for Resource Allocation in Shared Computer Systems},
journal = {Technical Report, Digital Equipment Corporation},
year = {1984},
number = {DEC-TR-301},
annote = {One aspect we often strive for in network system is "fairness". It could be a "fair" allocation of bandwidth, "fair" throughputs or response times. In this paper, a fairness index is derived having some very nice properties: - It is population size independent - It is scale and metric independent - It is bounded between 0 and 1 - It is a continuous function. The authors exemplify the usefulness of the fairness index by applying it to problems in various fields, for example, computer networks and social science. Even though the abundance of mathematical expressions make this paper a little bit hard to penetrate, I found their discussion on how to quantify fairness very profitable, and could most certainly be used when specifying other performance properties.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, April 12, 2000 at 09:37:57 (MEST)}
}
@article{Estrin96,
author = {J. Estrin and S. Casner},
title = {Multimedia Over {IP}: Specs Show the Way},
journal = {Data Communications},
year = {1996},
pages = {5},
month = {August},
annote = {This article gives an easy-to-grasp introduction to RTP (Real Time Transport Protocol) and its accompaning protocol RTCP (Real Time Transport Control Protocol). At the end of the article, the authors gives an example of how an RTP session work by discussion an audio conference. I found this example being this articles major contribution.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, April 18, 2000 at 17:00:37 (MEST)}
}
@article{Caceres91,
author = {R. Caceres and P. Danzig and S. Jamin and D. Mitzel},
title = {Charactersitics of Wide-Area {TCP/IP} Conversations},
journal = {Computer Communication Review},
year = {1991},
volume = {21},
number = {4},
pages = {pp. 101 - 112},
month = {September},
annote = {This paper is the result of an attempt to characterize the traffic on the Internet. Instead of confining themselves to network and transport layers, they have studied the characteristics of application-level traffic, both interactive, e.g. telnet, rlogin, x11, and bulk transfers, e.g. ftp, smtp, nntp. The traffic data was collected at two university campuses, University of Southern California and University of Californa at Berkeley, and one industrial research laboratory, Bellcore. The conclusions from their measurements where: - 80% of the time a classic bulk transfer application, e.g. ftp, transfer less than 10 kb/session. - Traffic generated by bulk applications, e.g. ftp, smtp, nntp, is strongly bidirectional. - Interactive applications routinely generate 10 times more data in one direction than the other, using packet sizes ranging from 1 byte to 512 bytes. - Interactive packet interarrivals closely match a constant plus exponential distribution. },
bibdate = {Tuesday, April 18, 2000 at 17:20:20 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Baker96,
author = {F. Baker},
title = {Real-Time Services for Router Nets},
journal = {Data Communications},
year = {1996},
pages = {pp. 85 - 90},
month = {March},
annote = {It is far from easy to fully comprehent how the Integrated Services Architecture (ISA) works just by wading through some of the RFCs. In this article, Fred Baker, Senior Software Engineer at Cisco Systems Inc. and also the chair of the IETF, tries to explain ISA from a practitioners viewpoint. The article starts off by elaborating on some of the obstacles we face when trying to run multimedia traffic over the Internet. Two types of multimedia traffic are defined: elastic and inelastic traffic. Elastic traffic characterizes itself by having the ability to adapt its behavior to ambient network conditions, which its not the case for inelastic traffic. In the rest of the article, the author discusses the impacts ISA have on the current Internet infrastructure, e.g. routers. The article concludes with a section on missing pieces in ISA, e.g. the fact that ISA almost entirely is targeted towards audio and video, leaving out other applications that might benefit from guaranteed-bandwidth channels.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, April 18, 2000 at 18:00:34 (MEST)}
}
@article{Stephenson98,
author = {A. Stephenson},
title = {Diffserv and MPLS: A Quality Choice},
journal = {Network Magazine},
year = {1998},
month = {November},
annote = {Two emerging technologies in the endeavour to provide guaranteed and predictable QoS in IP networks are Differentiated Services (DiffServ) and MultiProtocol Label Switching). DiffServ evolved from Integrated Services (IS) but scales better than its predecessor since it does not employ RSVP. Unlike IS, DiffServ offers QoS for classes of connections, not single ones. MPLS, on the other hand, has its roots in such technologies as IP switching and tag switching. IP packets is tunneled through the link setup by MPLS. In this article, an overview is given over DiffServ and MPLS. The text is succinct, but still written in an easy-to-read manner. It is recommended to anyone who, like me, wants to learn more about these technologies but do not want to invest hours wading through all drafts and RFCs.},
bibdate = {Thursday, April 20, 2000 at 08:14:50 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Passmore98,
author = {D. Passmore},
title = {Guaranteed Performance},
journal = {Business Communications Review},
year = {1998},
month = {October},
annote = {Guaranteed performance is one of the foundations of multimedia communication. In this article, the problem of achieving guaranteed performance is discussed. The conclusion is that the prize is very high, sometimes maybe too high. Firstly, you need som way of performing priorization of traffic flows, then you have to control the traffic entry into the network, for example by using RED (Random Early Detection). Controlled latency can be achieved through WFQ (Weighted Fair Queueing). Eventually, you need some way to reserve bandwidth beforehand, for example by sing RSVP (ReSource reserVation Protocol). The combination of WFQ, RED, and RSVP are sufficient to guarantee low network latency. This is as a matter of fact equivalent to the IETF's IntServ definition. The downside to this arrangement is the well-known fact that RSVP doesn't scale well to large networks. I found this article very thought-provoking. It surely gives a meaning to the set phrase: "There's no such thing as a free lunch!".},
bibdate = {Thursday, April 20, 2000 at 08:38:05 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Heimlich90,
author = {S. Heimlich},
title = {Traffic Characterization of the NSFNET National Backbone},
journal = {ACM conference on Measurement and modeling of computer systems, Boulder, CO USA},
year = {1990},
pages = {257-258},
month = {May},
annote = {This paper summarizes the result of an investigation of the traffic on the NSFNET backbone. A more in-depth analysis of this investigation can be found in this paper's full-text version. In this short-text version, the FTP traffic has been analyzed. Traditionally, models of packet arrival in communication networks have assumed either Poisson or compound Poisson arrival patterns. A previous investigation of a LAN at MIT found that packet arrival followed neither of these models. Instead, traffic followed a more general model, commonly named "packet train", which describes network traffic as a collection of packet streams. Train characteristics on NSFNET was not found to be as striking as those found on the MIT LAN. However, for FTP, the train length for all traffic throughout the day was relatively constant with a mean of 9.2 packets and a standard deviation of 0.23 packets.},
url = {http://www.acm.org/pubs/citations/proceedings/metrics/98457/p257-heimlich/},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, April 25, 2000 at 08:50:18 (MEST)}
}
@article{Lucantoni94,
author = {D. Lucantoni and M. Neuts and A. Reibman},
title = {Methods for Performance Evaluation of VBR Video Traffic Models},
journal = {IEEE/ACM Transactions on Networking},
year = {1994},
volume = {2},
number = {2},
pages = {176-180},
annote = {In this paper a model is proposed that represent a single video source as a Markov renewal process whose states represent different bit rates. In order to compare how well various models predict traffic characteristics, two novel goodnes-of-fit metrics are proposed: a leaky bucket contour plot and a quantization histogram. The Markov renewal process was compared to a model named "Discrete autoregressive model" and outperformed this model both with respect to capture burstiness and quantization effects.},
url = {http://www.acm.org/pubs/citations/journals/ton/1994-2-2/p176-lucantoni/},
bibdate = {Tuesday, April 25, 2000 at 08:54:27 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Durst96,
author = {R. Durst and G. Miller and E. Travis},
title = {{TCP} Extensions for Space Communications},
journal = {Proceedings of the second annual international conference on Mobile computing and networking, White Plains, NY USA},
year = {1996},
pages = {15},
month = {November},
annote = {The space communication environment imposes other constraints to TCP than ordinarily found in fixed terrestial environments. This implies that in order to use TCP in such an envronment some modifications and extensions to TCP have to be done. In this paper, the authors proposes a new TCP-based protocol, SPCS-TP (Space Communications Protocol Standards-Transport Protocol), that is specifically designed for space communication. More specifically, this protocol addresses the following properties of space communication: 1) Error-prone links. Contrary to TCP, SPCS-TP uses two mechanisms to identify the source of loss. TCP's default assumption is that all loss is caused by congestion. SPCS-TP's default behaviour is a parameter that can be set by a network manager or an application on a per route basis. The second mechanism for identifying the source of loss is explicit signaling. The destination host or other network element, e.g. routers and groundstations, may send explicit signals to the sending TCP regarding the source of the packet loss. 2) Limited link capacity. SPCS-TP uses a loss-tolerant TCP header compression scheme that operates end-to-end at the transport layer, and a more efficient acknowledgement scheme than both cumulative-ack, SACK, and NACK, the SNACK-scheme (Selective Negative Acknowledgement Option). 3) Assymetric channels, i.e. more link capacity in one direction than the other. SPCS-TP removes the requirement that TCP acknowledges at least every other segment received. SPCS-TP also disregards sending immediate acknowledges when out-of-sequence packets have been queued. Instead, the SPCS-TP receiver delays acknowledgements for a configurable period of time that is related to the RTT. Also, SPCS-TP uses TCP Vegas variant of slow start which doubles the congestion window every other RTT as opposed to every RTT like standard TCP. SPCS-TP enters congestion avoidance based on TCP Vegas' specified technique for sensing the point at which the throughput increas tapers. Real tests and experiments with a testbed shows great improvements in using SPCS-TP as compared to TCP when having error-prone and highly assymetric connections.},
url = {http://www.acm.org/pubs/citations/proceedings/comm/236387/p15-durst/},
bibdate = {Tuesday, April 25, 2000 at 08:59:21 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Xylomenos99,
author = {George Xylomenos and George C. Polyzos},
title = {Internet Protocol Performance over Networks with Wireless Links},
journal = {IEEE Network},
year = {1999},
volume = {13},
number = {4},
pages = {55-63},
annote = {Contains a discussion of the problems of using TCP over wirelinks, in this work signified by wireless LANs and cellular telephony. Contains some info on RLP characteristics for GSM and CDMA systems. General description of wireless characteristics and results are quite thin and the results reported seems to be mainly derived by simple mathematical calculations and not taking retransmission and congestion avoidance behaiviour into account. A packet error rate of 47% is reported to give a TCP throughput of 52%, which seems way too large and the number is of questionable practical use.},
url = {papers/Xylomenos99_Link_Wireless_TCP.pdf},
bibdate = {Tuesday, April 25, 2000 at 15:59:48 (MEST)},
submitter = {Johan Garcia}
}
@article{Haas97,
author = {Z. Haas and P. Agrawal},
title = {Mobile-{TCP}: An Asymmetric Transport Protocol Design for Mobile Systems},
journal = {ICC'97, Montreal, Canada},
year = {1997},
month = {June},
url = {papers/icc97.ps.gz},
bibdate = {Saturday, April 29, 2000 at 20:37:23 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@inproceedings{Biaz99,
author = {S. Biaz and N. Vaidya},
title = {Discriminating Congestion Losses from Wireless Losses using Inter-Arrival Times at the Receiver},
booktitle = {Proceedings of the IEEE Symposium on Application-Specific Systems and Software Engineering and Technology (ASSET)},
address={Richardson, TX, USA},
year = {1999},
month = mar,
annote = {Tries to distinguish congestion losses from losses over the wireless channel based on inter-arrival times. Assuming that the wireless link is the bottleneck link, there should always be a packet ready for transmission over the link. If a packet loss is detected but packets clock out at an even pace, then the loss is contributed to congestion. If a loss is detected and the interarrival time is larger than the time to send two packets, then the loss is contributed to the wireless channel. The performance of the classification as well as the performance of TCP-Reno with the scheme is evaluated using ns-2. When the amount of congestion losses is low and the bandwidth-delay product large improvements of up to 20\% are reported. Performance is close to ideal (ie TCP reno with perfect knowledege of what caused a loss). Interesting in that it could be combined with PRTP-ECN. How would it perform over transparent mode GSM?},
url = {papers/asset99.ps.gz},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Saturday, April 29, 2000 at 20:41:57 (MEST)}
}
@article{Vaidya99,
author = {N. Vaidya and M. Mehta and C. Perkins and G. Montenegro},
title = {Delayed Duplicate Acknowledgements: A {TCP}-Unaware Approach to Improve Performance of {TCP} over Wireless},
journal = {Technical Report, Computer Science Dept., Texas A\&M University},
year = {1999},
month = {February},
annote = {Delayed duplicate acknowledgements tries to mimic snoop, but without the need for a TCP-aware link layer. Local retransmissions are performed over the wireless link. To prevent interference with TCP retransmissions, the TCP receiver delayes the third duplicate acknowledgement (to prevent triggering fast retransmit). The performance of the scheme is evaluated through simulations using ns-2. Results show that it can increase performance when errors occur over the wireless link and more than three packets can get reordered by the link layer. Also shows that it does not degrade performance much when congestion losses occur.},
url = {papers/dupacks99.ps.gz},
bibdate = {Saturday, April 29, 2000 at 20:46:14 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@inproceedings{Bakshi97,
author = {B. Bakshi and P. Krishna and N. H. Vaidya and D. K. Pradhan},
title = {Improving Performance of {TCP} over Wireless Networks},
booktitle = {Proceedings of the 17th International Conference on Distributed Computing Systems},
address={Baltimore, USA},
year = {1997},
month = may,
annote = {Evaluates two performance improvements through simulations (using ns-2, including an extension for wireless link characteristics based on a 2-state Markov model). Selecting the right packet size and using explicit bad state notifications (EBSN). For most error conditions, the optimal packet size differs from the link MTU (128 bytes in their simulations) and the default IP datagram size. EBSNs are sent by the base station to the sender after every unsuccesful attempt to transmit packets over the wireless link. When the sender receives an EBSN it retstarts its retransmission timer. Under the right conditions, performance can greatly improve.},
url = {papers/icdcs97.ps.gz},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Saturday, April 29, 2000 at 20:51:51 (MEST)}
}
@article{Floyd99a,
author = {S. Floyd and K. Fall},
title = {Promoting the Use of End-to-End Congestion Control in the {Internet}},
journal = {IEEE/ACM Transactions on Networking},
year = {1999},
volume = {7},
number = {4},
pages = {458-472},
month = {August},
url = {papers/collapse.may99.ps.gz},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Saturday, April 29, 2000 at 20:56:52 (MEST)}
}
@inproceedings{Balakrishnan98a,
author = {Hari Balakrishnan and Randy Katz},
title = { Explicit Loss Notification and Wireless Web Performance},
booktitle = {Proceedings Globecom Internet Mini-Conference},
address={Sydney, Australia},
year = {1998},
month = {November},
url = {http://wind.lcs.mit.edu/papers/globecom98/index.html},
bibdate = {Saturday, April 29, 2000 at 21:02:22 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Xylomenos99b,
author = {George Xylomenos and George C. Polyzos},
title = {Link Layer Support for Quality of Service on Wireless Internet Links},
journal = {IEEE Personal Communications},
year = {1999},
volume = {6},
number = {5},
pages = {52-60},
annote = {This paper describes some simulation experiments performed using ns-2. The setup is with one or two slow (14.4Kbps, 100ms) links at the ends. Snoop, Karn's RLP and a SACK scheme is compared to raw link service. The TCP performance is found to be improved by all three link layers, but snoop does not work well over two wireless links. UDP performance is also simulated with XOR FEC, Kran's RLP and Out-Of-Sequnce RLP. The results indicate a considerably lower delay and delay variability for OOS rlp and FEC, as expected. The paper continue by describing the Multi service link layer approach, that in effect is the same as previously discussed by Ludwig, i.e. to use different link layer protocols for different applications. A higher degree of abstraction is provided through the use of packet classifier, a number of service modules and a packet scheduler. Simulations made with simultaneous TCP and UDP flows show that sucha multi service link layer can keep up the performance gain achieved by the application suitable link layer protocol.},
url = {papers/Xylomenos99b_Links_supp_Wireless_QoS.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, May 02, 2000 at 15:45:20 (MEST)}
}
@book{Mouly92,
author = {Michel Mouly and Marie-Bernadette Pautet},
title = {The {GSM} System for Mobile Communications},
publisher = {Cell \& Sys},
year = {1992},
bibdate = {Tuesday, May 02, 2000 at 18:36:41 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@inproceedings{Mohr99,
author = {Alexander E. Mohr and Eve A. Riskin and Richard E. Ladner},
title = {Graceful Degradation over Packet Erasure Channels through Forward Error Correction},
booktitle = {Proceedings Data Compression Conference (DCC)},
year = {1999},
month = {March},
annote = {Nicely written paper that describes a method of segmenting progressive image data streams into packets and providing FEC unequally according to the relative importance of the progressive data. The method uses a channel loss estimator that produces a PMF of losses of n length. The paper describes this related to SPIHT wavelet coding, but the general principle is valid for any progressive coding. Very good visual results are obtained for 0,2 bpp total bitrate and 40% losses. Details regarding how the estimator was setup for these result are not given, so the estimator may have been optimized for each specific loss rate? Also 48 byte ATM packets were used, larger packets would probably be harder to handle.},
url = {papers/Mohr_Wavelet_FEC_graceful.ps.gz},
bibdate = {Tuesday, May 02, 2000 at 23:29:34 (MEST)},
submitter = {Johan Garcia}
}
@inproceedings{Gupta00,
author = {R. Gupta and M. Chen and S. McCanne and J. Walrand},
title = {A Receiver-Driven Transport Protocol for the Web},
booktitle = {Proceedings INFORMS '2000 Telecommunications Conference},
year = {2000},
annote = {In this paper a novel protocol, WebTP, is introduced. WebTP is a receiver-driven, request/response protocol primarily intended to be the transport protocol for HTTP transfers. WebTP tries in as much as possdible to adhere to the mechanisms used in TCP. For example, the flow control scheme follows the TCP model of an initial slow-start phase followed by a congestion avoidance phase. During congestion avoidance, WebTP employs a hybrid window-based and rate-based scheme. The receiver calculates a smoothed estimate of the round-trip-time and communicates this to the sender, which paces its transmission of packets in accordance to this round-trip-time estimate. Since the responsibility for reliable transmission of packets in WebTP lies completely with the receiver, it needs to have knowledge about the objects (e.g. the components of the Web page) it requests. The receiver accomplishes this by using a queue called the Expected Objects Queue which is a list of all the objects it has asked for but not received yet. Simulations show that WebTP achieves efficient utilization of a link, at the same time being fair to competing TCP flows.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, May 03, 2000 at 09:07:01 (MEST)}
}
@article{Ludwig00,
author = {Reiner Ludwig and Randy H. Katz},
title = {The {Eifel} Algorithm: Making {TCP} Robust Against Spurious Retransmissions},
journal = {ACM Computer Communication Review},
year = {2000},
volume = {30},
number = {1},
pages = {30-37},
month = jan,
annote = {This paper describes the eifel algorithm which aims to improve TCPs performance by eliminating the ack-ambiguity after spurious retransmissions. If a packet just has been delayed for a long time and not lost a timeout-triggered retransmission can occur. The sender then interprets the original ack as an ack for the retransmission and enters slow-start and also gets a number of dupacks when the retransmitted packets occur at the receiver. In this paper the timestamp option is used for removing the ambiguity, though other mechanisms could be used. To be noted is that this paper suggests a solution to problem that according to their earlier paper very rarely occur. The results was obtained in an emulated environment. },
url = {papers/Ludwig00_Eifel_TCP_robust_Spur_Retransm.pdf},
bibdate = {Wednesday, May 03, 2000 at 09:08:18 (MEST)},
submitter = {Johan Garcia}
}
@article{Ladid00,
author = {L. Ladid},
title = {IPv6 - The new-generation Internet},
journal = {Ericsson Review},
year = {2000},
number = {1},
pages = {6 - 13},
annote = {This article presents arguments for adopting IPv6 in the very near future. IPv4 has several drawbacks which makes it inadequate for the future Internet. For example, IPv4 can support as many as four billion unique addresses. But the actual allocation of space has locked up nearly 75% of these addresses; In the early days, institutions like MIT and AT&T were each allocated Class-A IP addresses. In recent years, IPv4 has begun to use a technique, CIDR, which makes it possible to allocate a variable portion of a 32-bit IPv4 address to a network or subnetwork. A drawback with CIDR is that it does not guarantee an efficient and scalable address hieararchy. The lack of uniformity in the current hierarchical address system, coupled with the need for rationing IPv4 addresses, greatly complicates Internet addressing and routing. Moreover, the renumbering of IPv4 sites, e.g. when a user changes from one ISP to another, is unneccessary complicated, and thus more expensive compared to IPv6. IPv6 will, apart from a huge address space, give us automatic configuration. IPv6 nodes can automatically configure themselves with DHCPv6. It will give us an improved privacy and security system as compared to IPv4. When NAT is employed in an IPv4 network, every packet needs to be converted which inhibits the use of IPsec. IPv6 will give improved control over QoS thorugh the use of TC flag and the flow label. Other areas in which IPv6 excel is mobility support, multicast.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, May 08, 2000 at 09:10:46 (MEST)}
}
@article{Pehrson00,
author = {S. Pehrson},
title = {{WAP} - The catalyst of the mobile {Internet}},
journal = {Ericsson Review},
year = {2000},
number = {1},
pages = {14 - 19},
annote = {WAP, the Wireless Application Protocol, is a suite of specifications that cover a microbrowser, a scripting language (WMLScript), integration with telephony applications through WTA (Wireless Telephony Applications), a mark-up language, WML, and a layered stack. In this article, the author argues for the widespread use of WAP. The three major incentives to use WAP are: - Will be compliant with existing 2G-wireless systems, and future 3G-wireless systems. - In its future incarnations, WML, will be compatible with XML/XHTML. - The WAP inititive is backed up by more than 250 actors. Event those actors having proprietary systems of their own have promised to convert to WAP, for example, Microsoft, NTT (I-mode), and the proponents of SIM-toolkit.},
bibdate = {Monday, May 08, 2000 at 09:13:16 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Paxson99,
author = {Vern Paxson},
title = {End-to-End Internet Packet Dynamics},
journal = {IEEE/ACM Transactions on Networking},
year = {1999},
volume = {7},
number = {3},
pages = {277-292},
month = {June},
annote = {This paper uses two set of traces from 94 and 95 to characterize the amount of packet loss, packet delay and out-of-order delivery that occurs. 100 Kbyte of TCP data was sent and traced at both the receiver and sender. The paper says that larger windows were used for the second set, but gives no figures. The method of measuring bottleneck bandwidth with packet pairs is discussed, and Packet Bunch Mode (PBM) is used to overcome problems of packet pair such as multi-channel links. The loss ratio average is reported to be 3% and 5% for each of the sets, with many connections lossless. The lossiest half of the connections lose on average 6% and 9%, respectively. The forward and reverse loss ration is relatively independent although in the same order of magnitude. Two mechanisms work here: reverse losses should be less since less data is sent, but higher since TCP does not adapt to ack loss, only data loss. Losses are typically bursty with a conditional probability of 20-50%. Redundant retransmissions, i.e where the data had/would arrive at the receiver where classified into three types: 1 unavoidable, due to large ack loss. 2 coarse feedback due to the go-back-n method used. 3 bad RTO. A majority of the redundant retransmissions were of type 2 for both sets, and hence SACK is argued for.},
url = {papers/Paxson99_end_end_packet_dyn.ps.gz},
submitter = {Johan Garcia},
bibdate = {Monday, May 08, 2000 at 17:26:42 (MEST)}
}
@article{Zhang00,
author = {Y. Zhang and V. Paxson and S. Shenker},
title = {The Stationarity of Internet Path Properties: Routing, Loss, and Throughput},
journal = {ACIRI Technical Report},
year = {2000},
month = {May},
annote = {This paper describes data that were measured between a network of 31 measurement hosts across the Internet, mainly in the US. The stationarity, i.e if previous behaiviour could be used for prediction was of interest. UDP loss rates were low, with 84 percent of the traces having loss rates lower than 1%, and the other traces a median loss of 5,1%. Only 0.3% was out-of-order, but the packets were sent with 50 msec interpacket times. Contains a lot of statistics reasoning.},
url = {papers/Zhang00_stationarity_Inet_Paths.ps.gz},
submitter = {Johan Garcia},
bibdate = {Monday, May 08, 2000 at 18:04:18 (MEST)}
}
@article{Shenker00,
author = {S. Shenker and L. Breslau},
title = {Two Issues in Reservation Establishment},
journal = {Proceedings of the conference on Applications, technologies, architectures, and protocols for computer communication },
year = {2000},
pages = {14-26},
month = {August},
annote = {Two questions that needs to be answered before introducing real-time services are: - How do reservation establishment protocols enable applications to receive their desired end-to-end service? - How can a reservation establishment mechanism enable applications to achieve the end-to-end service they desire in the face of a heterogeneous network? In this paper, two of the members of the Integrated Services Internet Project, tries to answer these questions. They advocate a solution that employs a novel approach to resource reservation, one-pass reservation with advertisement (OPWA). OPWA combines the flexibility and scalability of a one-pass reservation protocol with the support for guaraneed service given by a two-pass reservation protocol. The heterogeneity of Internet is accommodated by a replacement services. When a particular service is not offered end-to-end, it can be replaced at one or more routers by an alternate service.},
url = {papers/p14-shenker.pdf.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, May 09, 2000 at 08:43:40 (MEST)}
}
@article{Popescu99,
author = {Catalin T. Popescu and A. Udaya Shankar},
title = {Empirical TCP Profiles and Application},
journal = {7th Intl. Conf. on Network Protocols (ICNP'99)},
year = {1999},
month = {October},
annote = {This paper describes the use of TCP Profiles. A profiles is a function describing the instantaneous throughput surface in terms of instantaneous RTT and instantaneous loss ratio. The profile function is empirically derived by senderbased measurement of connections to a number of hosts so that a spread in RTT and loss ratio is achieved. The profiles are markedly different for different implementations of TCP Reno. NetBSD 1.2 for example has up to three times the througput of NT4SP3 for low loss ratios, but NT is better for higher losses. ns TCP profile is different from any implemented. Using tce proflies they introduce a evaluation method that is faster than ns simulations for large topologies, but does not capture the phase effects.},
url = {papers/Popescu99_Empiri_TCP_Profiles.ps.gz},
bibdate = {Tuesday, May 09, 2000 at 08:54:01 (MEST)},
submitter = {Johan Garcia}
}
@article{Sikdar00,
author = {B. Sikdar and S. Kalyanaraman and K. Vastola},
title = {Transfer Times and Steady State Throughputs of TCP connections},
journal = {submitted for publication},
year = {2000},
annote = {This paper develops a mathematical model for TCP throughput. Starting with a model for flows with no losses the mode is extended for single loss, multiple loss. It is then extended to cover cases with connections that are restrained by the receiver window. The model is verified by ns simulations which swow a tight match, and measurements also show a reasonable match. An empirical model is also presented. Some implications are also discussed, such as the effect of packet size and Receiver window on throughput. Also, a questionable argumentation says that non-persistent is better than persistent connections for web-traffic. This paper should be a good base for developing a mathematical model of PRTP.},
url = {papers/Sikdar00_TCP_math_model.ps.gz},
bibdate = {Tuesday, May 09, 2000 at 10:34:23 (MEST)},
submitter = {Johan Garcia}
}
@article{Paxson97a,
author = {Vern Paxson},
title = {Automated Packet Trace Analysis of TCP Implementations},
journal = {SIGCOMM '97},
year = {1997},
month = {September},
annote = {The paper describes tcpanaly, a tool for automatically analyzing a TCP implementation's behavior by inspecting packet traces produced by tcpdump. Tcpanaly can determine whether a given trace appears consistent with a given implementation and, if so, exactly why the TCP chose to transmit each packet at the time it did. The observed behavior of various TCP implementations is also discussed in the paper (among them Linux 1.0 and Linux 2.1.36). },
submitter = {Katarina Asplund},
bibdate = {Tuesday, May 09, 2000 at 19:11:54 (MEST)}
}
@article{Allman99b,
author = {Mark Allman and Vern Paxson},
title = {On Estimating Ent-to-End Network Path properties},
journal = {SIGCOMM '99},
year = {1999},
month = {September},
annote = {The paper examines, among other things, the problem with determining the setting of the retransmission timer (RTO. For this purpose, a large collection of TCP packet traces are analyzed. Of the different RTO estimator parameters, the one with the greatest effect is the lower bound placed on RTO, followed by the clock granularity, while other parameters had little effect. },
submitter = {Katarina Asplund},
bibdate = {Tuesday, May 09, 2000 at 19:25:33 (MEST)}
}
@article{Allman99a,
author = {Mark Allman},
title = {On the Generation and Use of {TCP} Acknowledgements},
journal = {ACM Computer Communications Review},
year = {1998},
month = {October},
annote = {This paper investigates several alternate ways to generate and utilize TCP acknowledgements that mitigate the negative impact that delayed ACKs can have on performance.The mechanisms that are examined include using delayed ACKs only when the sender is not in slow start, allowing senders to increase the amount of data being injected in the network based on the number of bytes acknowledged, and acking each segment instead of every other. },
bibdate = {Tuesday, May 09, 2000 at 22:14:57 (MEST)},
submitter = {Katarina Asplund}
}
@article{Bettstetter99,
author = {C. Bettstetter and H-J. V\"{o}gel and J. Ebersp\"{a}cher },
title = {{GSM} Phase 2+ General Packet Radio Service {GPRS}: Architecture, Protocols, and Air Interface },
journal = {IEEE Communication Surveys},
year = {1999},
volume = {2},
number = {3},
annote = {This paper provides a nice technical overview of GPRS and some relevant details. The overall system architecture is described, and the services classes are described. It seems that rather high delays are anticipated where the best delay class has a mean delay of < 0.5s and 95% delay of <1.5s for 128 byte packets. The session managment and Air interface is the described. A protocol stack overview is given and it seems that they have three reliability enhancing layers: FEC at the Physical Link Layer, ARQ at the Radio Link Control and ARQ at the Logical Link Control (and then ARQ at TCP). Interworking with IP networks are also touched in the article. },
url = {papers/Betstetter99_GPRS_arch_proto_aif.pdf},
bibdate = {Wednesday, May 10, 2000 at 08:25:33 (MEST)},
submitter = {Johan Garcia}
}
@article{Muller99,
author = {N. Muller},
title = {Improving Multimedia Performance Over TCP/IP Networks},
journal = {Ericsson intranet},
year = {1999},
month = {June},
annote = {The Internet and consequently, corporate intranets and VPNs, were not designed for multimedia traffic. The native TCP/IP suite of prototcols was designed to convey data traffic in the form of email and files over relatively low-speed links. The challenge for content developers and network managers is to improve the performance of high-bit-rate multimedia applications over relatively slow TCP/IP networks. In this article, problems with multimedia traffic over TCP/IP networks are discussed as well as solutions to these problems. Several characteristics of TCP/IP networks may cause problems for multimedia traffic. These include the protocols themselves, latency, bandwidth capacity, and response time. One way to alleviate the scarcity of bandwidth is using some form of active queueing technique, e.g. priority queueing, custom queueing, or weighted fair queueing. Solutions to the rest of the problems with the current design of TCP/IP networks are also discussed.},
bibdate = {Wednesday, May 10, 2000 at 11:32:30 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Floyd94,
author = {S. Floyd},
title = {{TCP} and Explicit Congestion Notification},
journal = {ACM Computer Communication Review},
year = {1994},
volume = {24},
number = {5},
pages = {10-23},
month = {October},
annote = {This is one of the first papers discussing Explicit Congestion Notification, ECN. ECN only prescribes a way to unambiguously signal congestion to the sender. To be of any use it has to be combined with a mechanism to detect the congestion in the first place. Two mechanisms are presented in this paper. In the DECbit scheme, the queue sizes at the gateways are monitored. When their sizes exceed one, they start setting the ECN bit in the packet header of arriving packets. The other scheme is RED, Random Early Detection, where the probability of setting the ECN bit increases as the buffer queue builds up. Simulations of ECN in LAN and WAN environments show that ECN can improve both delay and throughput. It seems as if the improvements are more obvious in LANs. ECN also alleviates the effects of coarse clock granularities.},
url = {papers/tcp_ecn.4.ps.Z},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, May 30, 2000 at 08:08:30 (MEST)}
}
@article{Floyd93,
author = {S. Floyd and V. Jacobson},
title = {Random Early Detection Gateways for Congestion Avoidance},
journal = {IEEE/ACM Transactions on Networking},
year = {1993},
volume = {1},
number = {4},
pages = {397-413},
annote = {This paper presents Random Early Detection, RED, gateways for congestion avoidance in packet-switched networks. The gateway detects incipient congestion by computing the average queue size. The gateway could notify connections of congestion either by dropping packets arriving at the gateway or by setting a bit in packet headers. When the average queue size exceeds a preset threshold, the gateway drops or marks each arriving packet with a certain probability, where the probability is a function of the average queue size. RED gateways keep the average queue size low while allowing occasional bursts of packets in the queue. During congestion, the probability that the gateway notifies a particular connection to reduce its window is roughly proportional to that connection's share of the bandwidth through the gateway. Furthermore, the RED gateway has no bias against bursty traffic, as compared to Drop Tail or Random Drop gateways, and avoids the global synchronization of many connections decreasing their window at the same time.},
url = {papers/p397-floyd.pdf},
bibdate = {Tuesday, May 30, 2000 at 08:17:07 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Li00,
author = {B. Li and M. Hamdi and D. Jiang and X. Cao and Y. Hou},
title = {QoS-Enabled Voice Support in the Next-Generation Internet: Issues, Existing Approaches and Challenges},
journal = {IEEE Communications Magazine},
year = {2000},
volume = {38},
number = {4},
pages = {54-61},
month = {April},
annote = {This article begins with a survey of the existing technologies in supporting voice over IP networks, especially the components in the IETF Internet telephony architecture (SIP, SDP, RTP), and the H.323 recommendations. It then discusses the Intserv and Diffserv service models for QoS support in the Internet and concludes with a presentation of Cisco's and Lucent's solutions to offering IP telephony.},
submitter = {Katarina Asplund},
bibdate = {Tuesday, May 30, 2000 at 14:45:00 (MEST)}
}
@article{Li00a,
author = {Jin Li, Hong-Hui Sun},
title = {A Virtual Media (Vmedia) JPEG2000 Interactive Image Browser},
journal = {ACM Multimedia 2000 Los Angeles},
year = {2000},
annote = {A Vmedia JPEG2000 interactive image browser is presented. Two key technologies are used: implements the ROI (region of interest) functionality of JPEG2000 and the use of virtual media (Vmedia) protocol to manage and deliver the ROI bitstream in an efficient manner. The ROI is defined by the user on a low bandwidth client device (i.e PDA). Only the ROI portion of the compressed bitstream is streamed to the client.},
url = {papers/p501-li.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:46:58 (CET)}
}
@article{Rao00,
author = {H. Rao and Y. Lin and S. Cho},
title = {iGSM: VoIP Service for Mobile Networks},
journal = {IEEE Communications Magazine},
year = {2000},
volume = {38},
number = {4},
pages = {62-69},
month = {April},
annote = {This article proposes iGSM, a VoIP value-added service for mobile network. A GSM subscriber ordering the iGSM service uses the standard GSM services when he/she is in the GSM network. When the person moves to the IP network, a H.323 terminal (IP phone or PC) can be utilized to receive call delivery to the mobile station number. The GSM roaming mechanism determines whether the subscriber is in the GSM or IP network. The implementation of iGSM does not require modifications to the GSM network, just a translation between GSM MAP and H.323. },
submitter = {Katarina Asplund},
bibdate = {Tuesday, May 30, 2000 at 15:24:11 (MEST)}
}
@article{Liao00,
author = {W. Liao and J. Liu},
title = {VoIP Mobility in IP/Cellular Network Internetworking},
journal = {IEEE Communications Magazine},
year = {2000},
volume = {38},
number = {4},
pages = {70-75},
month = {April},
annote = {This article discusses the problems associated with the interworking between IP networks and cellular networks through H.323 gateways. In the current version of H.323, the signals have to be transmitted through the PSTN to reach the H.323 system, making transmission very inefficient. The authors propose a simple solution to this problem, where the existing call transfer supplementary service is used to provide VoIP mobility in the H.323 IP telephony networks. },
bibdate = {Tuesday, May 30, 2000 at 15:41:35 (MEST)},
submitter = {Katarina Asplund}
}
@article{Hassan00,
author = {M. Hassan and A. Nayandoro and M. Atiquzzaman},
title = {Internet Telephony: Services, Technical challenges, and Products},
journal = {IEEE Communications Magazine},
year = {2000},
volume = {38},
number = {4},
pages = {96-103},
month = {April},
annote = {This article discusses new services that can be expected from Internet telephony, the technical challenges and solutions, and some emerging products and targeted market segments. },
bibdate = {Tuesday, May 30, 2000 at 16:12:40 (MEST)},
submitter = {Katarina Asplund}
}
@article{Aracil99,
author = {J. Aracil and D. Morato and M. Izal},
title = {Analysis of Internet Services in IP over ATM Networks},
journal = {IEEE Communications Magazine},
year = {1999},
volume = {37},
number = {12},
pages = {92-97},
month = {December},
annote = {This article presents a trace-driven analysis of IP over ATM services. The traces show for example that 75% of the traffic in bytes is Web data, and that the average size of a Web transfer is 6.5kB with a 99% percentil of 70kB. It is also shown that 95% of the packet bursts consists of 1 or 2 packets. Interesting for its statistics of real Internet traffic.},
bibdate = {Tuesday, May 30, 2000 at 17:38:32 (MEST)},
submitter = {Katarina Asplund}
}
@article{Eichler00,
author = {G. Eichler and H. Hussman and G. Mamais and I. Venieris and C. Prehofer and S. Salsano},
title = {Implementing Integrated and Differentiated Services for the Internet with ATM Networks: A Practical Approach},
journal = {IEEE Communications Magazine},
year = {2000},
volume = {38},
number = {1},
pages = {132- 141},
month = {January},
annote = {This article presents a network architecture that supports QoS for Internet applications. The architecture supports both Intserv and Diffserv, where Diffserv is used in the core ATM network and Intserv in the access network. },
submitter = {Katarina Asplund},
bibdate = {Tuesday, May 30, 2000 at 17:49:54 (MEST)}
}
@article{Rejaie99a,
author = {R. Rejaie and M. handley and D.estrin},
title = {RAP: An End-to-end Rate-based Congestion Control Mechanism for Realtime Streams in the Internet},
journal = {IEEE Infocom'99},
year = {1999},
month = {March},
annote = {This paper presents the design and evaluation of the RAP protocol. RAP is an end-to-end rate-based congestion control mechanism that is suited for unicast playback of realtime streams as well as other semi-reliable Internet applications. The goals of RAP are to be well behaved and TCP-friendly. Simulations show that RAP is TCP-friendly as long as TCP's congestion control is dominated by the additive-increase / multiplicative-decrease algorithm.},
bibdate = {Tuesday, May 30, 2000 at 18:31:26 (MEST)},
submitter = {Katarina Asplund}
}
@article{Sinha99,
author = {Prasun Sinha and Thyagarajan Nandagopal and Narayanan Venkitaraman and Raghupathy Sivakumar and Vaduvur Bharghavan},
title = {{WTCP}: A Reliable Transport Protocol for Wireless Wide-Area Networks},
journal = {Wireless Networks},
year = {2002},
volume = {8},
number = {2},
pages = {301-316},
month = {March},
annote = {This paper presents WTCP, a reliable transport protocol that addresses rate control and reliability over commercial WWAN networks such as CDPD. WTCP is rate-based, uses only end-to-end mechanisms, performs rate control at the receiver, and uses inter-packet delays as the primary metric for rate control. WTCP also distinguishes the cause of packet loss and adjusts the transmission rate accordingly. Simulation results indicate that WTCP can improve performance by 20%-200%, compared to TCP. (JG updated ref 020709)},
url = {papers/Sinha02_WTCP.pdf},
submitter = {Katarina Asplund},
bibdate = {Tuesday, May 30, 2000 at 18:48:04 (MEST)}
}
@article{Sinha02,
author = {Prasun Sinha and Thyagarajan Nandagopal and Narayanan Venkitaraman and Raghupathy Sivakumar and Vaduvur Bharghavan},
title = {{WTCP}: A Reliable Transport Protocol for Wireless Wide-Area Networks},
journal = {Wireless Networks},
year = {2002},
volume = {8},
number = {2},
pages = {301-316},
month = {March},
annote = {This paper presents WTCP, a reliable transport protocol that addresses rate control and reliability over commercial WWAN networks such as CDPD. WTCP is rate-based, uses only end-to-end mechanisms, performs rate control at the receiver, and uses inter-packet delays as the primary metric for rate control. WTCP also distinguishes the cause of packet loss and adjusts the transmission rate accordingly. Simulation results indicate that WTCP can improve performance by 20%-200%, compared to TCP. (JG updated ref 020709, kept Sinha99 for backward compatibility)},
url = {papers/Sinha02_WTCP.pdf},
submitter = {Katarina Asplund},
bibdate = {Tuesday, May 30, 2000 at 18:48:04 (MEST)}
}
@article{Kim99,
author = {T. Kim and S. Lu and V. Bharghavan},
title = {Loss Proportional Decrease based Congestion Control in the Future Internet},
journal = {TIMELY Research Report},
year = {1999},
month = {July},
annote = {This paper presents a new congestion control algorithm based on the Linear Increase / Loss Proportional Decrease (LIPD) paradigm, and compare it with the predominant Linear Increase / Multiplicative Decrease (LIMD) algorithm. In LIPD, the reduction of the transmission rate of a connection upon loss feedback is proportional to the fraction of lost packets. This algorithm assumes that routers can provide fair packet marking/dropping with early congestion detection/notification.},
bibdate = {Tuesday, May 30, 2000 at 19:16:45 (MEST)},
submitter = {Katarina Asplund}
}
@article{Cisco,
author = {Cisco},
title = {Congestion Management Overview},
journal = {Cisco Press},
annote = {In this excerpt, taken from one of Cisco's routing manuals, several common active queuueing techniques are presented. More specifically, the following queueing techniques are discussed: Weighted Fair Queueing (WFQ) and variants of WFQ, i.e. Flow-based WFQ (WFQ), Distributed WFQ (DWFQ), Class-based WFQ (CBWFQ), and Low Latency Queueing, Custom Queueing (CQ), and Priority Queueing(PQ). WFQ classifies traffic into different flows based on some characteristics, i.e. source and destination addresses, protocol. Each flow are given a weight which determines the transmit order of queued packets. There are several advantages to using WFQ, i.e., WFQ alleviates the round-trip delay variability, making TCP congestion control and slow start mechanisms more accurate, WFQ shares the bandwith fairly between the flows, while at the same time give low-volume traffic precendence over high-volume traffic. The "normal" version of WFQ is also called Flow-based WFQ. DWFQ differs from WFQ in that you have several output queues, not just one. In CBWFQ, you define traffic classes based on match criteria including protocols, input interfaces etc. Packets satisfying the match criteria for a class constitute the traffic for that class. A queue is reserved for each class. LLQ brings strict priority queueing to CBWFQ, i.e. making it possible for delay-sensitive data such as voice to be dequeued and sent first. In CQ, you specify a certain number of bytes to forward from a queue each time the queue is serviced. The queues are serviced in a round-robin fashion. Finally, PQ makes it possible to give some types of flows absolute priority over others.},
url = {papers/qcconman.pdf},
bibdate = {Wednesday, May 31, 2000 at 07:28:23 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Mogul92,
author = {J. Mogul},
title = {Observing TCP Dynamics in Real Networks},
journal = {Proc. SIGCOMM '92 Symposium on Communications Architectures and Protocols},
year = {1992},
pages = {305-317},
month = {August},
annote = {The behavior of TCP in simple situations is well-understood, but when multiple connections share a set of network resources the protocol can exhibit more complex phenomena. One such phenomena is "ACK-compression", which occur when ACKs are being delayed. Since a TCP sender's self-clocking depends on the arrival of ACKs at the same spacing with which the receiver generated them, "ACK-compression" could mislead a TCP sender into sending more data than the network can accept. This could in the long run lead to congestion and loss of efficiency. In this paper, it is shown how "ACK-compression" could be observed in "real life" by using traces of a busy link in the Internet. It is also shown that "ACK-compression" is highly correlated with packet losses. The result of this paper indicates that in the future an automatic detection of "ACK-compression" could be a key to an efficient utilization of the Internet. },
url = {papers/WRL-TR-92.2.ps.gz},
bibdate = {Wednesday, May 31, 2000 at 07:37:22 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Smith99,
author = {P. Smith and B. Jamoussi and D. Fedyk},
title = {MPLS: A Progress Report},
journal = {Network Magazine},
year = {1999},
month = {November},
annote = {This article presents the main ideas and incentives behind MPLS, MultiProtocol Label Switching. Seven reasons to leave circuit-switching in favor of MPLS are given. The primary ones being MPLS' ability to transport a multitude of other protocols, minimizing the changes of routing hardware, and allowing flow control over a wide range of availbale paths to a destination.},
bibdate = {Wednesday, May 31, 2000 at 08:18:41 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Rosen99,
author = {E. Rosen and A. Viswanathan and R. Callon},
title = {Multiprotocol Label Switching Architecture},
journal = {Internet Draft},
year = {1999},
month = {August},
annote = {In TCP/IP-based networks each router makes an independent forwarding decision for a packet. This can be a quite time consuming process, especially when compared to a circuit switched network. In MPLS, the routing decision is done just once, at the ingress router. The ingress router gives each packet a label which determines its routing path. At subsequent hops, no further routing decisions are taken. Rather, the label is used as an index into a table which specifies the next hop, and a new label. The old label is replaced with the new label, and the packet is forwarded to its next hop. This draft gives an in-depth description of MPLS, especially how labels are managed, and how labeling can be done using the DLCI (Data Link Connection Identifier) of Frame Relay and VPI/VCI (Virtual Path Identifier/Virtual Circuit Identifier) of ATM.},
url = {papers/draft-ietf-mpls-arch-06.txt},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, June 13, 2000 at 10:08:15 (MEST)}
}
@article{Wang93,
author = {Z. Wang and J. Crowcroft},
title = {Analysis of Burstiness and Jitter in Real-Time Communications},
journal = {Conference proceedings on Communications architectures, protocols and applications, September 13 - 17, 1993, San Francisco, CA USA},
year = {1993},
pages = {13-19},
month = {September},
annote = {This paper examines the relationship between burstiness and jitter. Burstiness is defined as the difference between the actual arrival time and the expected arrival time, while jitter is viewed as the difference between a packet's relative distance to the first packet at the source and at the destination. Based on the assumption that a synchronization unit (A synchronization unit is defined as a group of packets that share a common fixed delay at the playback buffer.) can be characterized by two parameters, the highest rate at which the server is never idle and the peak rate of the incoming traffic, it is shown that jitter occurs as an effect of differences in the burstiness of the input and output traffic at the client application.},
url = {papers/p13-wang.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, June 18, 2000 at 13:37:40 (MEST)}
}
@article{Wang93b,
author = {Z. Wang and J. Crowcroft},
title = {Analysis of Burstiness and Jitter in Multimedia Communications},
journal = {IEEE GLOBECOM'93, Texas, USA,},
year = {1993},
month = {December},
annote = {This paper is only a reworked version of the paper "Analysis of Burstiness and Jitter in Real-Time Communications". Multimedia communication differs from traditional data communication in that it is often very sensitive to the delay jitter. More precisely, the tail of the delay distribution. Motivated by the fact that the performance of multimedia communication is dominated by the packets with worst case delay, the approach taken in this paper is to use a non-probabilistic worst case analysis. Multimedia traffic is often partitioned into so called synchronization units, a group of packets sharing a common fixed delay offset. By modeling the traffic in a synchronization unit with two parameters, the highest rate at which the server is never idle and the peak rate of the incoming traffic, it could be derived a relationship between burstiness and jitter. Furthermore, upper bounds for burstiness and jitter could be formulated.},
url = {papers/jb-g.ps.Z},
bibdate = {Sunday, June 18, 2000 at 13:51:00 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Mohan99,
author = {R. Mohan and J. Smith and C.-S. Li},
title = {Adapting Multimedia Internet Content For Universal Access},
journal = {IEEE Transactions on Multimedia},
year = {1999},
volume = {1},
number = {1},
pages = {104-114},
month = {March},
url = {papers/Mohan98_Adapting_multimedia_proxies.pdf},
submitter = {Johan Garcia},
bibdate = {Monday, June 19, 2000 at 15:42:44 (MEST)}
}
@article{Chandra99,
author = {Surendar Chandra and Carla Schlatter Ellis},
title = {JPEG Compression Metric as a Quality Aware Image Transcoding},
journal = {Second Usenix Symposium on Internet Technologies and Systems (USITS '99)},
year = {1999},
month = {October},
bibdate = {Monday, June 19, 2000 at 16:19:08 (MEST)},
submitter = {Johan Garcia}
}
@article{Han98,
author = {Richard Han and Pravin Bhagwat and Richard LaMaire and Todd Mummert and Veronique Perret and Jim Rubas},
title = {Dynamic adaptation in an image transcoding proxy for mobile WWW browsing},
journal = {IEEE Personal Communication},
year = {1998},
volume = {5},
number = {6},
pages = {8-17},
month = {December},
submitter = {Johan Garcia},
bibdate = {Monday, June 19, 2000 at 16:24:43 (MEST)}
}
@article{Watson93,
author = {Andrew B. Watson},
title = {Visual optimization of DCT quantization matrices for individual images },
journal = {Proc. SPIE, 1913},
year = {1993},
pages = {2020-216},
bibdate = {Monday, June 19, 2000 at 18:32:50 (MEST)},
submitter = {Johan Garcia}
}
@article{Brakmo96,
author = {L. Brakmo and L. Peterson},
title = {Experiences with Network Simulation},
journal = {Proceedings of the ACM SIGMETRICS conference on Measurement & modeling of computer systems},
year = {1996},
month = {May},
annote = {In this paper, the x-Sim network simulator is described. The x-Sim simulator is based on an emulation of an OS kernel, the x-kernel. Simulators like ns2, REAL, and Netsim only contains code that mimics the main characteristics of a protocol. In contrast to this, x-Sim uses complete implementations of the protocols. This has the advantage that even smaller differences between various implementations of TCP/IP are taken into account. The usefulness of x-Sim is demonstrated in some case studies. The paper concludes with some general guidelines for network simulation.},
url = {papers/p80-brakmo.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, June 21, 2000 at 08:38:35 (MEST)}
}
@article{Keshav93,
author = {S. Keshav},
title = {A Control-Theoretic Approach to Flow Control},
journal = {Proceedings of the conference on Communications architecture & protocols},
year = {1993},
pages = {3-15},
month = {September},
annote = {This paper uses a control-theoretic approach to regulate a data flow. First, a Kalman state estimator is used. This approach is however found to be impractical, and a novel estimation scheme is shown to be more promising. This scheme is based on fuzzy logic.},
url = {papers/p3-keshav.pdf},
bibdate = {Monday, June 26, 2000 at 10:52:04 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Brunstrom00,
author = {Anna Brunstrom and Katarina Asplund and Johan Garcia},
title = {Enhancing {TCP} Performance by Allowing Controlled Loss},
booktitle = {Proceedings of the SSGRR 2000 Computer \& eBusiness Conference},
address = {L'Aquila, Italy},
year = {2000},
month = aug,
annote = {This paper describes the design and implementation of PRTP. Some initial experimental results that compare the steady-state performance of PRTP with the performance of TCP are presented. The applicability of PRTP for wireless environments and the issue of congestion control are also discussed. As an example application for PRTP, an image recoding proxy system is presented. In addition, related work in the areas of partial reliability and TCP adaptation for wireless environments is discussed.},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Thursday, June 29, 2000 at 16:18:25 (MEST)}
}
@misc{Nistnet00,
author = "{NIST Internetworking Technology Group}",
title = {{NISTNet} network emulation package},
note = {Available from \url{http://snad.ncsl.nist.gov/nistnet/}. Visited 2011-12-28.},
}
@inproceedings{Garcia00a,
author = {Johan Garcia and Anna Brunstrom},
title = {A Robust {JPEG} Coder for a Partially Reliable Transport Service},
booktitle = {Proceedings 7th International Workshop on Interactive Distributed Multimedia Systems and Telecommunication Services (IDMS)},
address={Enschede, The Netherlands},
year = {2000},
publisher = {Springer},
month = oct,
annote = { A partially reliable service provides applications with the possibility of a flexible tradeoff between reliability and delay/throughput. Appropriately designed coders are, however, required to fully utilize a partially reliable service.In this paper we present a JPEG image coder tailored to suit the behavior of a partially reliable byte stream service. With regular JPEG, data loss typically results in severely distorted images. The robust recoder employs three major modifications to standard JPEG in order to adapt to the partially reliable transport: (1) extended resynchronization markers in order to be able to resynchronize effectively, (2) block interleaving in order to spread out the loss of a packet across the image and (3) error concealment in order to minimize the perceived quality loss. The modifications incorporate both new inventions, such as random window interleaving, as well as variations of previously §known techniques.},
bibdate = {Thursday, June 29, 2000 at 16:40:15 (MEST)},
submitter = {Anna Brunstrom}
}
@article{Amer99,
author = {P. Amer and S. Iren and G. Sezen and P. Conrad and M. Taube and A. Caro},
title = {Network-conscious GIF image transmission over the Internet},
journal = {Computer Networks},
year = {1999},
volume = {31},
number = {7},
pages = {693-708},
month = {April},
submitter = {Katarina Asplund},
bibdate = {Friday, June 30, 2000 at 09:00:02 (MEST)}
}
@article{Watson00,
author = {Andrew B. Watson et al},
title = {DCTune 2.0},
journal = {http://vision.arc.nasa.gov/dctune/},
year = {2000},
month = {June},
annote = {The DCTune JPEG perceptually optimizing coder. Can also report perceptual error.},
submitter = {Johan Garcia},
bibdate = {Friday, June 30, 2000 at 17:38:39 (MEST)}
}
@article{Biaz98,
author = {S. Biaz and N. H. Vaidya},
title = {Distinguishing Congestion Losses from Wireless Transmission Losses: A Negative Result},
journal = { Seventh International Conference on Computer Communications and Networks (IC3N), New Orleans},
year = {1998},
month = {October},
annote = {Tries to distinguish congestion losses from wireless transmission losses by using statistica available at the TCP sender. Three different schemes are evaluated: one based on TCP-Vegas, one based on the normalized throughput gradient and one based on the normalized delay gradient. The three loss predictors are evaluated through simulations (using ns). Neither one works to well.},
url = {papers/ic3n98.ps.gz},
bibdate = {Monday, July 03, 2000 at 15:30:42 (MEST)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Paxson94,
author = {V. Paxson},
title = {Empirically Derived Analytic Models of Wide-Area TCP Connections},
journal = {IEEE/ACM Transactions on Networking},
year = {1994},
volume = {2},
number = {4},
pages = {316-336},
annote = {In this paper, three million TCP connections at seven different sites are analyzed in an endeavour to derive analytical, as well as, empirical models of the TCP traffic. An in-depth discussion of the rationale behind the statistical methods used are given. The major findings of the investigation was: - Random variables associated with TCP traffic can be described by both analytical and empirical models. In some cases, the analytical models even gave a better prediction of the traffic than the empirical ones. - Both types of models, the empirical and analytical, showed major discrepancies in the upper 1% tails of the distributions. - Bulk-transfer traffic is best modeled by a log-normal distribution (the distribution you get when taking the logarithm of a normal stochastic variable). - Bulk-transfer traffic is highly assymetric.},
url = {papers/p316-paxson.pdf},
bibdate = {Thursday, July 06, 2000 at 09:44:14 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Bakre97,
author = {Ajay V. Bakre and B.R. Badrinath},
title = {Implementation and Performance Evaluation of {Indirect TCP}},
journal = {IEEE Transactions on Computers},
year = {1997},
volume = {46},
number = {3},
month = {March},
annote = {The paper describes the implementation and a performance evaluation of I-TCP. TCP connections are divided at the base station. I-TCP is standard TCP extended with support for mobility (on top of a mobile IP). When a mobile host moves to another cell buffered data and the state of the split connections are transmitted to the new base station. The performance was evaluated in a WaveLAN. I-TCP performs better than regular TCP. Losses over the wireless link are recovered faster although I-TCP has the same error recovery mechanisms as standard TCP. One reason is that the round trip time for the connection between the base station and the mobile host is shorter than the round trip time would be for an end-to-end connection. },
url = {papers/t0260.pdf.gz},
bibdate = {Thursday, July 06, 2000 at 13:31:35 (MEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Mehta98,
author = {Miten Mehta and N. H. Vaidya},
title = {Delayed Duplicate Acknowledgements: A Proposal to Improve Performance of {TCP} on Wireless Links},
journal = {Tecnical Report (Texas A\&M University)},
year = {1998},
month = {February},
annote = {In the delayed dupacks approach the mobile host delays the third and subsequent dupacks. This prevents the sender (fixed host) from taking congestion control action while the base station retransmits data over the wireless link. A disadvantage is that the dupacks are delayed also in case of congestion. Hence, error recovery is delayed. The delayed dupacks approach is improved by the use of ELNR. The mobile host delays dupacks unless it receives an ELNR which indicates that the network may be congested. The base station transmits an ELNR to the mobile host if it receives a segment out of sequence. Upon receipt of an ELNR the mobile host stops to delay dupacks. When the base station has received the missing packets, the base station transmits an explicit delayed dupack activation notification (EDDAN) to the mobile host. The EDDAN indicates that the network is no longer congested, and subsequent dupacks should be delayed until a new ELNR arrives.},
url = {papers/mehta.ps.gz},
bibdate = {Thursday, July 06, 2000 at 13:50:30 (MEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Biaz97,
author = {Saad Biaz and Miten Mehta and Steve West and Nitin H. Vaidya},
title = {{TCP} over Wireless Networks Using Multiple Acknowledgements},
journal = {Technical Report 97-001, Computer Science, Texas A\&M University },
year = {1997},
month = {January},
annote = {The report describes a partial acknowledgment protocol that can be used to discriminate congestion losses from congestion losses when the mobile host is the receiver. In addition to the acknowledgement from the receiver, a partial acknowledgement is transmitted from the base station over the fixed network. Provided that no data or segments are lost, the sender receives two acknowledgements for each segment: a partial acknowledgement from the base station and a complete acknowledgement from the mobile host. If only the partial acknowledgement arrives, then the sender can conclude that data must have been lost over the wireless network and no congestion control action is required. A missing partial acknowledgement, on the other hand, indicates that the network is congested. },
url = {papers/Biaz-97-001.ps.gz},
bibdate = {Thursday, July 06, 2000 at 13:57:29 (MEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Cobb95,
author = {Jorge A. Cobb and Prathima Agrawal},
title = {Congestion or Corruption? {A} Strategy for Efficient Wireless {TCP} Sessions},
journal = {IEEE Symposium on Computers and Communications, Alexandria, Egypt},
year = {1995},
pages = {262-268},
annote = {The partial acknowledgement protocol proposed in the paper considers the following cases: fixed host to mobile host, mobile host to fixed host, and mobile host to mobile host. Two new acknowledgements are introduced: last hop acknowledgement and first hop acknowledgment. These acknowledgements are transmitted from the base station to the sender. If a segment arrives on a fixed link, then a last hop acknowledgment is transmitted. On the other hand, if a segment arrives on a wireless link, then a first hop acknowledgement is transmitted. A base station transmits a last hop acknowledgement to a fixed host sender for each segment it receives. A last hop acknowledgement indicates that data could be transmitted successfully through the fixed network. If the sender that received the last hop acknowledgment also receives an acknowledgement from the mobile host, then the sender can conclude that no loss occurred. On the other hand, if no acknowledgement arrives from the mobile host, then that is an indication of loss over the wireless link. The sender does not, as in standard TCP, reduce its transmission rate because of loss over the wireless link. The lack of last hop acknowledgements is an indication of loss in the fixed network and the sender reduces its transmission rate. },
url = {papers/Cobb-lhack.ps.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thursday, July 06, 2000 at 14:24:29 (MEST)}
}
@article{Shenoy00,
author = {Prashant J. Shenoy and Harrick M. Vin:},
title = {Failure recovery algorithms for multimedia servers},
journal = {Multimedia Systems},
year = {2000},
volume = {8},
number = {1},
pages = {1-19},
month = {January},
annote = {This paper describes a system which interleaves video onto several servers in order to provide good error concealment if one disk fails. Suggests 8-way mean for reconstructing the JPEG DC value. Both coefficient and block interleaving is used. Also discusses MPEG recovery and suitable RAID strategies. Contains a more formal interleaving description.},
url = {papers/Shenoy00_Failure_recovery_mm_servers.pdf},
bibdate = {Tuesday, July 11, 2000 at 18:19:56 (MEST)},
submitter = {Johan Garcia}
}
@article{Ledesma00,
author = {S. Ledesma and D. Liu},
title = {Synthesis of Fractional Gaussian Noise Using Linear Approximation for Generating Self-Similar Network Traffic},
journal = {Computer Communication Review},
year = {2000},
volume = {30},
number = {2},
pages = {4-17},
month = {April},
annote = {Since the publication of the seminal paper in the study of self-similar data traffic "On the Self-similar Nature of Ethernet Traffic" there has been a flood of interest in generating self-similar functions. The most frequently used models are: Fractional Gaussian Noise (FGN), Fractional Brownian Motion (FBM), and fractional autoregressive integrated moving average (F-ARIMA). This paper deals with an optimization of a method, developed by Vern Paxson, for generating FGN traffic. Often when you want to generate values from a stochastic process, you start with its frequency spectrum and make an inverse Fourier transform to get the values. This process entails an infinite series summation, something that has to be approximated. In this paper, a method is presented, approximating this infinite series summation by a linear approximation. Simulation studies shows a decrease in runtime by almost 50%.},
bibdate = {Tuesday, August 01, 2000 at 11:48:56 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Medina00,
author = {A. Medina and I. Matta and J. Byers},
title = {On the Origin of Power Laws in Internet Topologies},
journal = {Computer Communications Review},
year = {2000},
volume = {30},
number = {2},
pages = {18-28},
month = {April},
annote = {When performing simulations and performance evaluations on novel technologies in the Internet it is of outmost importance to do an adequate modeling. This is not as trivial as one might first think. Commonly used methods as the Random Method, the Regular Method, the Hierarchical Method, and the Transit-Stub Methods have turned out to lack some important properties making them less suitable for predictions. Recent studies have shown that Internet topologies exhibit power laws. These laws can be used to validate the accuracy of a given tool in generating Internet topologies. In this paper, BRITE, a parametrized topology generator, is presented. This generator is used to reason about the factors underlying the power laws. Two factors turn out to be of paramount importance to the generation of topologies: preferential connectivity and incremental growth. Preferential connectivity dictates the tendency of a new node to connect to those existing nodes that have higher outdegrees, while incremental growth dictates that new nodes join the internet in an incremental fashion (this in contrast to the Random Method where all nodes are generated at the same time).},
bibdate = {Wednesday, August 02, 2000 at 11:06:18 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Willinger98,
author = {W. Willinger and V. Paxson},
title = {Where Mathematics meets the Internet},
journal = {Notices of the American Mathematical Society},
year = {1998},
volume = {45},
number = {8},
pages = {961-970},
month = {August},
annote = {This paper looks at ways to use mathematics when doing performance evaluations on data communication networks. Already from the inception, mathematics played an important role in the development of the telephone networks (PSTNs) of today. Not so for the Internet. Why? The authors of this paper points out one important reason: The static nature of traditional PSTNs made the calculations tractable. It was even possible to find "laws" which seemed to hold for all telephone networks. The success story of the Poisson process has not been possible to repeat on data communication networks. Instead, it has been shown that traffic on a data communication network has a fractal behavior, lending itself to heavy-tailed distributions like the Pareto. This makes modeling data communication networks far more complicated than PSTNs. The authors conclusion is that mathematics has a role in the development of data communication networks too.},
url = {papers/internet-math-AMS98.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, August 07, 2000 at 08:16:46 (MEST)}
}
@article{Padhye98,
author = {J. Padhye and V. Firoiu and D. Towsley and J. Krusoe},
title = {Modeling {TCP} Throughput: A Simple Model and its Empirical Validation},
journal = {Proceedings of ACM SIGCOMM},
year = {1998},
pages = {303-314},
month = aug,
annote = {In this paper, a simple analytic model of TCP Reno's throughput is presented. The model takes into account timeouts but not fast retransmits. Comparisons with real-world traces, shows a good match between the model and the observed bahavior. Parts of the work done in this paper is used in the paper: "Transfer Times and Steady State Throughput of TCP Connections".},
url = {papers/p303-padhye.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, August 07, 2000 at 08:30:08 (MEST)}
}
@article{Breslau00,
author = {L. Breslau et al},
title = {{Advances in Network Simulation}},
journal = {IEEE Computer},
year = {2000},
volume = {33},
number = {5},
pages = {59-67},
month = {May},
annote = {This article describes the VINT (Virtual InterNetwork Testbed) project and the ns2 network simulator.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, August 15, 2000 at 10:21:02 (MEST)}
}
@article{Sreenan00,
author = {C.J. Sreenan and J.-C. Chen and P. Agrawal and B. Narendran},
title = {Delay Reduction Techniques for Playout Buffering},
journal = {IEEE Transactions on Multimedia},
year = {2000},
volume = {2},
number = {2},
pages = {88-100},
month = {June},
annote = {Interesting mainly because it discusses aging of previous data and presents three variants to do this. Also nice since it references Anna.},
bibdate = {Friday, August 18, 2000 at 14:18:54 (MEST)},
submitter = {Johan Garcia}
}
@article{Stallings98,
author = {W. Stallings},
title = {{High-Speed Networks} - {TCP}/{IP} and {ATM Design Principles}},
journal = {Prentice Hall},
year = {1998},
bibdate = {Wednesday, August 23, 2000 at 10:41:15 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Asplund00,
author = {K. Asplund and J. Garcia and A. Brunstrom and Sean Schneyer},
title = {{Decreasing Transfer Delay Through Partial Reliability}},
booktitle = {Proceedings PROMS'00 },
year = {2000},
month = {October},
annote = {This paper describes the design and implementation of PRTP and evaluates the performance of PRTP in steady state. The example application that has been developed to test the protocol is also discussed briefly. Our initial experimental results suggest that PRTP can indeed decrease transfer delay in networks in which packet loss occurs. Especially for applications that can tolerate rather high loss rates (5-10\%), and where the network does not have a higher average loss rate than the loss tolerance, the performance improvement is considerable.},
bibdate = {Wednesday, August 23, 2000 at 16:40:08 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Garcia00b,
author = {Johan Garcia},
title = {Integrated Testing - Test Setup},
journal = {Working Report, Karlstad University},
year = {2000},
month = oct,
annote = {Provides test setup information for the integrated testing},
url = {papers/Garcia00b_IntegratedTesting.ps.gz},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 10, 2000 at 10:41:17 (MEST)}
}
@article{Kleinrock00,
author = {L. Kleinrock},
title = {On Some Principles of Nomadic Computing and Multi-Access Communications},
journal = {IEEE Communications},
year = {2000},
volume = {38},
number = {7},
pages = {46-50},
month = {July},
annote = {This article identifies some of the problems that comes with nomadic communication, especially when it is wireless. Examples of topics discussed are: - access control schemes, - routing, The author elaborates on the trade-off between the response time and throughput, i.e. at low throughput we tend to get a good response time and vice versa.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, October 19, 2000 at 07:54:16 (MEST)}
}
@article{Grinnemo00,
author = {K-J Grinnemo},
title = {Plan for Simulating the Test Cases Studied in the Integrated Testing Activity},
journal = {Working report KaU},
year = {2000},
month = {October},
annote = {This document describes how the simulation of the test cases studied in the integrated testing activity will be performed.},
url = {papers/grinnemo00.ps},
bibdate = {Thursday, October 19, 2000 at 15:10:51 (MEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Montgomery94,
author = {D. Montgomery and G. Runger},
title = {Applied Statistics and Probability for Engineers},
journal = {John Wiley},
year = {1995},
annote = {Introductory textbook for a first course in applied statistics and probability.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, October 24, 2000 at 15:02:26 (MEST)}
}
@article{Ludwig00_diss,
author = {Reiner Ludwig},
title = {Eliminating Inefficient Cross-Layer Interactions in Wireless Networking},
journal = {Ph.D. Thesis, Aachen University of Technology, Germany},
year = {2000},
month = {April},
annote = {The thesis analyses the interaction between TCP and GSM RLP. The result is that inefficient interactions are rare. A flow-adaptive link layer is proposed. This link layer adapts the degree of reliablility provided depending on the type of application traffic. The thesis also describes TCP-Eifel. TCP-Eifel is more efficient than standard TCP because of a more accurate RTT estimate.},
url = {papers/diss_ludwig.pdf.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thursday, November 02, 2000 at 12:30:21 (MET)}
}
@article{DeSimone95,
author = {Antonio De Simone and Sanjiv Nanda},
title = {Wireless data: systems, standards, service},
journal = {Wireless Networks },
year = {1995},
volume = {1},
number = {3},
pages = {241-253},
month = {March},
url = {papers/p241-de_simone.pdf.gz},
bibdate = {Thursday, November 02, 2000 at 13:55:08 (MET)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Chuah97,
author = {Mooi Choo Chuah and Bharat Doshi and Subra Dravida and Richard Ejzak and Sanjiv Nanda},
title = {Link layer retransmission schemes for circuit-mode data over the {CDMA} physical channel},
journal = {Mobile Networks and Applications},
year = {1997},
volume = {2},
number = {2},
pages = {195 - 211},
month = {October},
url = {papers/p195-chuah.pdf.gz},
bibdate = {Thu Nov 2 14:21:27 CET 2000},
submitter = {Annika Wennstr\"{o}m}
}
@article{Salkintzis99,
author = {Apostolis K. Salkintzis},
title = {A Survey of Mobile Data Networks},
journal = {IEEE Communication Surveys},
year = {1999},
volume = {2},
number = {3},
url = {papers/Salkintzis.pdf.gz},
bibdate = {Thu Nov 2 14:21:27 CET 2000},
submitter = {Annika Wennstr\"{o}m}
}
@article{Sarikaya00,
author = {Behcet Sarikaya},
title = {Packet Mode in Wireless Networks: Overview of Transition to Third Generation},
journal = {IEEE Communications Magazine},
year = {2000},
month = {September},
url = {papers/sarikaya.pdf.gz},
bibdate = {Thu Nov 2 14:21:27 CET 2000},
submitter = {Annika Wennstr\"{o}m}
}
@misc{rfc1144,
author="V. Jacobson",
title={Compressing {TCP/IP} Headers for Low-Speed Serial Links},
series="Request for Comments",
number="1144",
howpublished="RFC 1144 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1990,
month=feb,
url="http://www.ietf.org/rfc/rfc1144.txt",
}
@misc{RFC2393,
author = {A. Shacham and R. Monsour and R. Pereira and M. Thomas},
title = {{RFC 2393: IP} Payload Compression Protocol (IPComp)},
year = {1998},
month = {December},
url = {papers/rfc2393.txt.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Nov 2 14:21:27 CET 2000}
}
@misc{RFC2507,
author = {M. Degermark and B. Nordgren and S. Pink},
title = {{RFC 2507: IP} Header Compression},
year = {1999},
month = {February},
url = {papers/rfc2507.txt.gz},
bibdate = {Thu Nov 2 14:21:27 CET 2000},
submitter = {Annika Wennstr\"{o}m}
}
@misc{RFC2508,
author = {S. Casner and V. Jacobson},
title = {{RFC 2508}: Compressing {IP/UDP/RTP} Headers for Low-Speed Serial Links},
year = {1999},
month = {February},
url = {papers/rfc2508.txt.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Nov 2 14:21:27 CET 2000}
}
@misc{RFC2509,
author = {M. Engan and S. Casner and C. Bormann},
title = {{RFC 2509}: {IP} Header Compression over {PPP}},
year = {1999},
month = {February},
url = {papers/rfc2509.txt.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Nov 2 14:21:27 CET 2000}
}
@misc{rfc2401,
author="S. Kent and R. Atkinson",
title="{Security Architecture for the Internet Protocol}",
series="Request for Comments",
number="2401",
howpublished="RFC 2401 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1998,
month=nov,
url="http://www.ietf.org/rfc/rfc2401.txt",
}
@article{pilcerror00,
author = {S. Dawkins and G. Montenegro and M. Kojo and V. Magret and N. Vaidya},
title = {End-to-end Performance Implications of Links with Errors},
journal = {INTERNET DRAFT},
year = {2000},
month = {September},
url = {http://www.ietf.org/internet-drafts/draft-ietf-pilc-error-05.txt},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Nov 2 16:48:37 CET 2000}
}
@misc{RFC2415,
author = {K. Poduri and K. Nichols and },
title = {{RFC 2415}: Simulation Studies of Increased Initial {TCP} Window Size},
year = {1998},
month = {September},
annote = {Karl-Johan - This RFC presents results from a set of simulations with increased TCP initial window (IW). The main objectives were to explore the conditions under which the larger IW proved beneficial, and to determine the effects the larger IW might have on other traffic flows using an IW of one segment. The results from the simulations indicate that increasing the initial window size to 3 packets (or 4380 bytes) helps to improve perceived performance.},
url = {papers/rfc2415.txt.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Nov 2 16:48:37 CET 2000}
}
@misc{RFC1122,
author="R. Braden",
title={Requirements for {Internet} Hosts - Communication Layers},
series="Request for Comments",
number="1122",
howpublished="RFC 1122 (Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1989,
month=oct,
url="http://www.ietf.org/rfc/rfc1122.txt",
}
@article{Allman99c,
author = {Mark Allman},
title = {{TCP} Byte Counting Refinements},
journal = {ACM Computer Communication Review},
year = {1999},
volume = {3},
number = {29},
month = {July},
url = {papers/bc-ccr.ps.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Nov 2 16:48:37 CET 2000}
}
@article{Semke98,
author = {J. Semke and M. Mathis and J. Mahdavi},
title = {Automatic {TCP} Buffer Tuning},
journal = {SIGCOMM 98},
year = {1998},
url = {papers/automatic_buffer_tuning.ps.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Nov 2 16:48:37 CET 2000}
}
@article{Garcia00c,
author = {J. Garcia and J. Gustafsson},
title = {Data Communication in {GSM} networks},
journal = {Project Report, Karlstad University},
year = {2000},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Nov 2 16:48:37 CET 2000}
}
@article{Kondi99,
author = {L. Kondi},
title = {Low Bit Rate SNR Scalable Video Coding and Transmission},
journal = {Ph.D. Thesis, Department of Electrical and Computer Engineering,Northwestern University},
year = {1999},
month = {December},
annote = {This theses gives a short overview ot the H.263 standard. Two SNR scalable H.263 variants are presented which have lower complexity than the scalability proposed in H.263+. Both are based on sending the DCT cefficients in scans, just as porgressive JPEG. The first variant uses a fixed three-layer decomposition, whereas the second provides a more flexible decomposition and a operational rate-distorsion algorithm to provid rate control. An algortihm for bit allocation between source and channel coding and between scalable layers is also discussed.},
url = {papers/Kondi99low_video-phd.pdf},
submitter = {Johan Garcia},
bibdate = {Wednesday, November 08, 2000 at 09:54:06 (MET)}
}
@article{Brown88,
author = {R. Brown},
title = {Calendar Queues: A Fast O(1) Priority Queue Implementation for the Simulation Event Set Problem},
journal = {Communications of the ACM},
year = {1988},
volume = {31},
number = {10},
pages = {1220-1227},
month = {October},
annote = {At the core of a discrete event simulator, we find a scheduler. There are many ways of implementing this scheduler. ns2 provides three types of schedulers: simple linked-list, heap, calendar queue, and "real-time" queue. This article presents a priority queue implementation for the event set problem (representation of the pending event set in discrete event simulation) which has relatively low overhead, and which on the basis of intuiton and experimental evidence has O(1) average performance.},
url = {papers/brown88.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, November 09, 2000 at 09:10:27 (MET)}
}
@misc{rfc2988,
author="V. Paxson and M. Allman",
title={Computing {TCP}'s Retransmission Timer},
series="Request for Comments",
number="2988",
howpublished="RFC 2988 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2000,
month=nov,
url="http://www.ietf.org/rfc/rfc2988.txt",
}
@article{Fall00,
author = {K. Fall},
title = {ns Notes and Documentation},
journal = {The VINT Project},
year = {2000},
month = {February},
annote = {This is the reference manual for the ns2 simulator.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, November 16, 2000 at 17:45:57 (MET)}
}
@article{Marsan82,
author = {M. Marsan and M. Gerla},
title = {Fairness in Local Computing Networks},
journal = {Proceedings IEEE International Conference on Communication (ICC)},
year = {1982},
month = {June},
annote = {In this paper, a quantitative measure of fairness is proposed.},
bibdate = {Wednesday, November 22, 2000 at 18:12:28 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Garcia00d,
author = {Garcia, Johan and Brunstrom, Anna},
title = {Progressive Parsing Transcoding of {JPEG} Images},
journal = {Proc. 7th Int. Workshop on Mobile Multimedia Communications (MoMuC2000), Tokyo, Japan},
year = {2000},
month = {October},
annote = {This paper describes the progressive parsing method of efficiently recoding progressive JPEG images by truncating them. The method is described and the buffering advantage obtained by this method is highlighted. Also, the factors contributing to the improved recoding quality are discussed. These factors are: optimized {H}uffman tables, quantization table skewing and less requantization noise injection. Three implementation methods are also described: Simple, inter-scan and intra-scan truncation. },
url = {papers/Garcia00d_ProgressiveJPEG_momuc.pdf},
bibdate = {Friday, November 24, 2000 at 09:45:23 (MET)},
submitter = {Johan Garcia}
}
@article{Brunstrom99b,
author = {Anna Brunstrom},
title = {{Analysis and Implementation of a Partially Reliable Transport Protocol for Multimedia Applications: Project Specification, Year 2}},
journal = {Project Specification, Karlstad University},
year = {1999},
month = {Dec}
}
@article{Kulik99,
author = {J. Kulik et al},
title = {Paced TCP for High Delay-Bandwidth Networks},
journal = {Proceedings of IEEE Globecom},
year = {1999},
month = {December},
annote = {In this paper, paced TCP is proposed as a solution to the problem of queueing bottlenecks in wireless, high bandwidth-delay networks.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, November 27, 2000 at 15:05:10 (MET)}
}
@article{Aron98,
author = {M. Aron and P. Druschel},
title = {{TCP}: Improving Startup Dynamics by Adaptive Timers and Congestion Control},
journal = {Technical Report (TR98-318), Dept. of Computer Science, Rice University},
year = {1998},
month = {June},
annote = {This paper proposes a framework for the management of timing in TCP. Based on this framework, an estimator of the slow-start threshold is suggested, and it is shown how "pacing" could be used to mitigate the adverse effects of congestion control and timing during the startup phase.},
url = {papers/Aron98.ps},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, November 27, 2000 at 16:14:10 (MET)}
}
@article{Henderson98,
author = {T. Henderson et al},
title = {On Improving the Fairness of {TCP} Congestion Avoidance},
journal = {Proceedings of IEEE Globecom `98},
year = {1998},
volume = {1},
pages = {539-44},
annote = {In this paper, two changes to TCP's congestion avoidance algorithm are evaluated in an effort to improve TCP's fairness. The first one, the Constant-Rate policy proposed by Sally Floyd, turned out to be effective only when RED was used, and all connections adopt the policy. The second one, the increase-by-K (IBK) policy, seemed to be beneficial for long RTT connections to be slightly more aggressive during the additive increase phase of congestion avoidance.},
url = {papers/Henderson98.ps.gz},
bibdate = {Monday, November 27, 2000 at 17:13:51 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Ibanez98,
author = {J. Ibanez and K. Nichols},
title = {Preliminary Simulation Evaluation of an Assured Service},
journal = {INTERNET DRAFT},
year = {1998},
month = {August},
annote = {This draft presents a simulation analysis of Assured Service (AS), an end-to-end service based on the differentiated service enhancements for IP. The idea behind AS is to give the customer the assurance of a minimum throughput, even during periods of congestion, while allowing hime to consume more bandwidth when network load is low. The simulations suggest that AS cannot provide clearly defined and consistent guarantees, at least when applied to the whole Internet.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, November 30, 2000 at 14:01:23 (MET)}
}
@article{Thompson97,
author = {K. Thompson and G. Miller and R. Wilder},
title = {Wide-Area Internet Traffic Patterns and Characteristics (Extended Version)},
journal = {IEEE Network},
year = {1997},
pages = {10-23},
month = {November},
annote = {This paper reports the results from a study of Internet backbone traffic usage and characteristics performed at MCI.},
bibdate = {Thursday, November 30, 2000 at 15:54:54 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Calvert97,
author = {K. Calvert and M. Doar and A. Nexion and E. Zegura},
title = {Modeling Internet Topology},
journal = {IEEE Communications Magazine},
year = {1997},
month = {June},
annote = {This article discusses how graph-based models can be used to represent the topology of large networks, particularly aspects of locality and hierarchy present in the Internet. Two implementations that generate networks whose topology resembles that of typical internetworks are described.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, November 30, 2000 at 16:31:47 (MET)}
}
@article{Chow99,
author = {H. Chow et al},
title = {A Feedback Control Extension to Differentiated Services},
journal = {INTERNET DRAFT},
year = {1999},
month = {March},
annote = {This draft presents a feedback control extension to differentiated services.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, November 30, 2000 at 16:52:17 (MET)}
}
@article{Floyd91,
author = {S. Floyd and V. Jacobson},
title = {On Traffic Phase Effects in Packet-Switched Gateways},
journal = {Computer Communication Review},
year = {1991},
volume = {21},
number = {2},
pages = {115-156},
month = {April},
annote = {In this paper, the notion of traffic phase in a packet-switched network is defined, and it is demonstrated how the phase differences between competing traffic streams can be the dominant factor in relative throughput. Drop Tail gateways in a TCP/IP network with strongly periodic traffic can result in systematic discrimination against some connections. It is demonstrated in simulations how this discrimination can be eliminated with the addition of appropriate randomization to the queue-management algorithms in the gateways. It is also shown how random early detection (RED) gateways correct the bias against bursty traffic existing in random gateways.},
url = {papers/floyd99.ps.gz},
bibdate = {Friday, December 01, 2000 at 08:49:24 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Mathis97,
author = {M. Mathis and J. Semke and J. Mahdavi},
title = {The Macroscopic Behavior of the {TCP} Congestion Avoidance Algorithm},
journal = {Computer Communications Review},
year = {1997},
volume = {27},
number = {3},
month = {July},
annote = {In this paper, a performance model for the TCP Congestion Avoidance algorithm is analyzed. The model is verified through both simulations and live Internet measurements. All simulations fitted the model when losses were infrequent or isolated. Live Internet tests showed a rough agreement with the model.},
url = {papers/mathis97.ps},
bibdate = {Friday, December 01, 2000 at 09:02:23 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Crovella96,
author = {M. Crovella and A. Bestavros},
title = {Self-Similarity in World Wide Web Traffic: Evidence and Possible Causes},
journal = {Proceedings, ACM Sigmetrics Conference on Measurement and Modeling of Computer Systems},
year = {1996},
month = {May},
annote = {Using as set of traces from actual user executions of NCSA Mosaic, the dependence structure of WWW traffic is examined. Evidence is shown that WWW traffic exhibits behavior that is consistent with self-similarity traffic models. Then, it is shown that the self-similarity in such traffic can be explained based on the underlying distributions of WWW document sizes, the effects of caching and user preference, the effect of user "think time", and the superimposition of many such transfers in a local area network.},
url = {papers/Crov96.pdf},
bibdate = {Monday, December 04, 2000 at 14:12:52 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{McKusick96,
author = {M. McKusick and K. Bostic and M. Karels},
title = {The Design and Implementation of the {4.4BSD} Operating System},
journal = {Addison-Wesley},
year = {1996},
month = {May},
annote = {This book describes the design and implementation of the BSD operating system--previously known as the Berkeley version of UNIX. Today, BSD is found in nearly every variant of UNIX, and is widely used for Internet services and firewalls, timesharing, and multiprocessing systems. },
bibdate = {Thursday, December 07, 2000 at 11:35:16 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Jain91,
author = {R. Jain},
title = {The Art of Computer Systems Performance Analysis},
journal = {John Wiley \& Sons, Inc.},
year = {1991},
annote = {This book presents theory essential to anyone performing performance analysis on computers. It gives an overview of measurement techniques, probability and statistics, experimental design, simulation, and much more.},
bibdate = {Monday, December 11, 2000 at 12:19:34 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Montgomery84,
author = {D. Montgomery},
title = {Design and Analysis of Experiments},
journal = {John Wiley \& Sons, Inc.},
year = {1984},
annote = {Introductory textbook dealing with the statistical design and analysis of experiments.},
bibdate = {Monday, December 18, 2000 at 12:39:56 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Wennstrom00,
author = {A. Wennstr\"{o}m and J. Garcia and J. Gustafsson and A. Brunstrom },
title = {{TCP} and {GSM} link layer interactions: initial measurements and analysis},
journal = {Project Report, Karlstad University},
year = {2000},
bibdate = {Friday, December 22, 2000 at 18:20:37 (MET)},
submitter = {Annika Wennstr\"{o}m}
}
@article{pilcpep00,
author = {J. Border and M. Kojo and J. Griner and G. Montenegro and Z. Shelby},
title = {Performance Enhancing Proxies},
journal = {INTERNET-DRAFT},
year = {2000},
month = {November},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Friday, December 22, 2000 at 18:24:50 (MET)}
}
@article{Sauve80,
author = {J. Sauve and J. Wong and J. Field},
title = {On Fairness in Packet-Switching Networks},
journal = {Proceedings of the 21st IEEE Computer Society International Conference},
year = {1980},
pages = {466-470},
month = {September},
annote = {Definition of fairness metric in terms of variance of network delays.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, December 28, 2000 at 15:49:56 (MET)}
}
@article{Wong82,
author = {J. Wong and S. Lang},
title = {Queueing Network Models of Packet Switching Networks, Part 1: Open Networks},
journal = {Performance Evaluation},
year = {1982},
annote = {Definition of fairness metric in terms of the square of the coefficient of variation of delay.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, December 28, 2000 at 15:54:04 (MET)}
}
@article{Wong82b,
author = {J. Wong and F. Sauve and J. Field},
title = {A Study of Fairness in Packet-Switching Networks},
journal = {IEEE Transaction on Communications},
year = {1982},
volume = {30},
number = {2},
pages = {346-353},
month = {February},
annote = {A definition of a fairness metric which sees fairness as a weighted measure of variance of delays.},
bibdate = {Thursday, December 28, 2000 at 15:58:22 (MET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Bodin00,
author = {U. Bodin et al},
title = {Load-tolerant Differentiation with Active Queue Management},
journal = {Computer Communication Review},
year = {2000},
volume = {30},
number = {3},
pages = {4-16},
month = {July},
annote = {Current work in the IETF aims at providing service differentiation in the Internet. One proposal is to provide loss differentiation by assigning levels of drop precedence to IP packets. In this paper, two active queue mechanisms (AQM), RIO (RED In and Out) and WRED (Weighted RED) are evaluated with respect to "sheltering" and "load-tolerance". A drop precedence level is said to be "sheltered" if traffic loads at higher drop precedence levels do not impede on its traffic. The "load-tolerance" property holds for an AQM mechanisms if it prevents starvation of high drop precedence traffic while preserving the hierarchy among drop precedence levels, i.e. traffic at a low drop precedence level must experience less drop probability than traffic at a higher drop precedence level. It is shown through simulations that both RIO and WRED can be configured to offer sheltering. However, it is done at the risk of starvation, i.e. "load-tolerance" cannot be kept. WRED can offer both "sheltering" and "load-tolerance" but not at the same time. To achieve both "sheltering" and "load-tolerance", modifications to RIO and WRED are proposed: Load-tolerant RIO (ltRIO) and WRED with Thresholds (WRT).},
bibdate = {Friday, January 05, 2001 at 13:26:17 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Ming00,
author = {I.T. Ming-Chit and D. Jinsong and W. Wang},
title = {Improving {TCP} Performance over Asymmetric Networks},
journal = {Computer Communication Review},
year = {2000},
volume = {30},
number = {3},
pages = {45-54},
month = {July},
annote = {Bandwidth asymmetry is quite common among modern network technologies. ADSL, cable TV, and a satellite link with a terrestrial return path are but a few examples of cases where asymmetry occur. The peformance of TCP transfer in the high-bandwidth direction can be severly reduced by the delay of acknowledgement packets in the reverse direction. In this paper, a novel approach to speed up the TCP transfer over an asymmetric network is proposed, ACE (Acknowledgement based on Cwnd Estimation). ACE bases the number of packets per acknowledgement on an receiver-based estimation of the size of the cwnd. Both simulations and experiments show that ACE improves the TCP throughput over asymmetric networks very significantly.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, January 08, 2001 at 09:18:05 (CET)}
}
@article{Estrin00,
author = {D. Estrin et al},
title = {Network Visualization with Nam, the VINT Network Animator},
journal = {IEEE Computer},
year = {2000},
pages = {63-68},
month = {November},
annote = {This article in "IEEE Computer" gives an overview over Nam, a network visualization and animation tool developed in the VINT project.},
bibdate = {Monday, January 15, 2001 at 10:08:18 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Fall99b,
author = {K. Fall},
title = {Network Emulation in the {VINT/NS} Simulator},
journal = {Proceedings of the fourth {IEEE} Symposium on Computers and Communications},
year = {1999},
month = {July},
annote = {This paper describes the ongoing work in the VINT project of implementing emulation capability in the ns simulator. Work in the area of emulation has generally proceeded along two paths: network emulation and environment emulation. Network emulation aims to allow simulated components to communicate with protocol implementations in the real world. Environment emulation extends the simulation model to allow real-world protocol implementations to participate. In ns, the first approach was adopted. Two modes of utilizing emulation may be characterized as opaque mode and protocol mode. In opaque mode, no manipulation of the packet contents are allowed as opposed to the protocol mode. At the time this article was published, the opaque mode was well-supported in ns, while protocol mode was only partially implemented. },
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, January 18, 2001 at 08:55:44 (CET)}
}
@article{Guardini00,
author = {I. Guardini and P. Fasano},
title = {The Role of Internet Technology in Future Mobile Data Systems},
journal = {IEEE Communications},
year = {2000},
volume = {30},
number = {11},
pages = {68-72},
month = {November},
annote = {This article discusses the main evolutionary trends for integrating mobile and IP technology. Three plausible scenarios are pictured. In the first one, extant mechanisms such as HLR/VLR (Home Location Register/Visitor Location Register) handles mobility, and a core IP network is used for intercommunication between radio access networks. The second scenario entails using Mobile IP to handle macromobility, relying on cellular telephony facilities (e.g. GPRS) to manage micromobility between base stations belonging to the same radio access network. Finally, in the last scenario, it is envisioned that each base station functions as an IP network access server equipped with IP mobility features, i.e. Mobile IP is used for macromobility as well as micromobility.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, January 22, 2001 at 08:52:48 (CET)}
}
@article{Moridera00,
author = {A. Moridera and K. Murano and Y. Mochida},
title = {The Network Paradigm of the 21st Century and Its Key Technologies},
journal = {IEEE Communications},
year = {2000},
volume = {30},
number = {11},
pages = {94-98},
month = {November},
annote = {This article proposes a new network paradigm which the call "the single-server-view network". In this paradigm, the network is logically separated in two planes; a data forwarding plane and a service control plane. The data forwarding plane is envisioned to use a very simple data transfer mechanism based on WDM (Wavelength Division Multiplexing). The service control plane consists of an agent layer, a service layer, and a dynamic policy management layer. The main objective of this network paradigm is to let the end users view the network as service delivering mechanism rather than a data transport mechanism.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, January 22, 2001 at 08:54:13 (CET)}
}
@article{Stephane00,
author = {A. Stephane and A. Mihailovic and A. Aghvami},
title = {Mechanisms and Hiearchical Topology for Fast Handover in Wireless Networks},
journal = {IEEE Communications},
year = {2000},
volume = {30},
number = {11},
pages = {112-115},
month = {November},
annote = {In this article, a new mechanism for performing fast handovers in IP-based wireless networks is proposed. The mechanism differentiates between two types of handovers; micromobility occurs between base stations within the same subdomain, while inter-subdomain handovers occur when a mobile terminal enters a new subdomain. The novelty of the mechanism is to retransmit buffered packets during micromobility handover, and to use multicasting to reestablish traffic flow during inter-subdomain movement.},
bibdate = {Monday, January 22, 2001 at 08:55:39 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Wennstrom00a,
author = {Annika Wennstr\"{o}m},
title = {Packetizing a Byte Stream},
journal = {Master's Thesis 2000:03, Karlstad University},
year = {2000},
month = {October},
annote = {One of the research directions discussed in the prestudy report is how to provide a message abstraction on top of PRTP. A message abstraction can be provided by a packetizing layer if the overhead introduced is small. For this thesis, a packetizing layer suitable for PRTP was implemented and evaluated. The problems of packetizing data on top of PRTP are examined and the implementation of the packetizing layer is described. The packetizing layer inserts message boundary markers in the PRTP byte stream. In order to guarantee data transparency, byte stuffing is performed. Data loss is detected and in case of loss the packetizing method resynchronizes with incoming data. The experimental results presented in the thesis show that in an environment with nonnegligible delays and limited need for byte stuffing, the overhead introduced by the packetizing layer is small. The conclusion of the thesis is that a message abstraction on top of PRTP can be provided by a packetizing layer.},
submitter = {Annika Wennstrom},
bibdate = {Tue Jan 23 12:12:38 CET 2001 }
}
@article{Brunstrom00a,
author = {Katarina Asplund and Anna Brunstrom and Johan Garcia and Karl-Johan Grinnemo},
title = {Analysis and Implementation of a Partially Reliable Transport Protocol for Multimedia Applications: Project Report},
journal = {Project Report, Karlstad University},
year = {2000},
month = {Dec}
}
@article{Asplund00b,
author = {Katarina Asplund},
title = {The Design and Implementation of {PRTP}},
journal = {Master's Thesis, Karlstad University},
year = {2000},
month = {October},
annote = {This thesis focuses on the design and implementation of PRTP in the Linux kernel. Some initial performance results are also presented, which indicate that PRTP can indeed decrease transfer times in networks where packet loss occurs. In addition, this thesis includes a survey of other higher layer protocols designed for multimedia that has been proposed in the literature. },
bibdate = {Monday, January 29, 2001 at 12:01:31 (CET)},
submitter = {Katarina Asplund}
}
@article{Wijesekera99,
author = {D. Wijesekera et al.},
title = {Experimental Evaluation of Loss Perception in Continuous Media},
journal = {ACM Multimedia Journal},
year = {1999},
volume = {7},
pages = {486-499},
month = {July},
annote = {Perception of multimedia quality, specified by QoS metrics, can be used by system designers to optimize customer satisfaction within resource bounds enforced by general-purpose computing platforms. Media losses, rate variations and transient synchronization losses have been suspected to affect human perception of multimedia quality. This paper presents metrics to measure such defects, and results of a series of user experiments that justify such speculations. Results of the study provide bounds on losses, rate variations and transient synchronization losses as a function of user satisfaction, in the form of Likert values. It is shown how these results can be used by algorithm designers of underlying multimedia systems. },
url = {papers/Wijesekera99.pdf},
bibdate = {Friday, February 02, 2001 at 09:17:11 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@misc{RFC2330,
author = {V. Paxson et al.},
title = {{RFC 2330: Framework for IP Performance Metrics}},
year = {1998},
month = {May},
annote = {In this RFC, a framework for defining performance metrics is presented. The fundamental concepts of 'metric' and 'measurement' are elaborated. In the final sections of the document, issues related to defining sound metrics and methodologies are discussed.},
bibdate = {Friday, February 02, 2001 at 11:54:28 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Karlsson95,
author = {G. Karlsson},
title = {Asynchronous Transfer of Video},
journal = {IEEE Communications Magazine},
year = {1995},
volume = {34},
number = {8},
pages = {118-126},
month = {August},
annote = {This paper gives an introduction to the issues involved in asynchronous video transfers. Brief overviews of video coding, rate control, multiplexing, as well as delay, error and loss control are given.},
url = {papers/karlsson95.ps.gz},
bibdate = {Monday, February 05, 2001 at 09:39:11 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Karlsson96,
author = {G. Karlsson},
title = {Quality Requirements for Multimedia Network Services},
journal = {Proceedings of "Radiovetenskap och kommunikation -96"},
year = {1996},
pages = {96-100},
month = {June},
annote = {This paper surveys the quality aspects of network services imposed by transfers of multimedia information.},
url = {papers/karlsson96.ps},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, February 05, 2001 at 09:51:09 (CET)}
}
@article{Steinmetz96,
author = {R. Steinnetz},
title = {Human Perception of Jitter and Media Synchronization},
journal = {IEEE Journal on Selected Areas in Communications},
year = {1996},
volume = {14},
number = {2},
pages = {61-72},
month = {February},
annote = {In this paper, an infrastructure for multimedia applications are presented. Various characteristics of multimedia data and the effect they have on the network.},
url = {papers/steinmetz96.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, February 05, 2001 at 10:01:19 (CET)}
}
@article{Mehaoua99,
author = {A. Mehaoua and R. Boutaba},
title = {The Impacts of Errors and Delays on the Performance of MPEG2 Video Communications},
journal = {Proceedings of the IEEE International Conference On Acoustics, Speech, and Signal Processing},
year = {1999},
volume = {4},
pages = {2195-2198},
month = {March},
annote = {Transmission of MPEG2-encoded video is one of the most demanding applications in terms of network resources and QoS requirement. It needs high bandwidth with stringent transmission delays. It can not tolerate large variations on delays and it requires low error and loss data rates. In this paper, the effects of errors and delays on both MPEG2 video and audio streams are analyzed.},
url = {papers/Mehaoua99.ps},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, February 05, 2001 at 11:33:21 (CET)}
}
@article{Karlsson96b,
author = {G. Karlsson},
title = {On the Characteristics of Variable Bit-Rate Video},
journal = {Proceedings of the Thirteenth Nordic Teletraffic Seminar},
year = {1996},
pages = {150-161},
month = {August},
annote = {The purpose of this paper is to review the characteristics that may be expected from variable bit-rate video. This is done in the context of ATM traffic classes.},
url = {papers/karlsson96b.ps},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, February 05, 2001 at 12:12:10 (CET)}
}
@article{Dempsey93,
author = {B. Dempsey and J. Liebeherr and A. Weaver},
title = {A New Error Control Scheme for Packetized Voice over High-Speed Local Area Networks},
journal = {Proceedings of the 18th IEEE Local Computer Networks Conference},
year = {1993},
pages = {99-100},
month = {September},
annote = {In this paper, a new error control mechanism for packet voice, referred to as Slack ARQ (S-ARQ). S-ARQ is based on extending the control time for the first packet in each talkspurt to allow for timely retransmission of lost packets.},
url = {papers/dempsey93.ps},
bibdate = {Tuesday, February 06, 2001 at 17:31:02 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Dempsey93b,
author = {B. Dempsey and J. Liebeherr and A. Weaver},
title = {A Delay-Sensitive Error Control Scheme for Continuous Media Communications},
journal = {Second IEEE Workshop on the Architecture and Implementation of High Performance Communication Subsystems (HPCS '93)},
year = {1993},
month = {September},
annote = {This paper presents a novel retransmission scheme, Slack ARQ, which targets retransmission of delay sensitive data, e.g. voice or video.},
url = {papers/dempsey93b.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, February 06, 2001 at 17:55:25 (CET)}
}
@article{Dempsey92,
author = {B. Dempsey and W. Strayer and A. Weaver},
title = {Adaptive Error Control for Multimedia Data Transfers},
journal = {International Workshop on Advanced Communications and Applications for High Speed Networks},
year = {1992},
pages = {279-289},
month = {March},
annote = {In this paper, a new retransmission scheme, PECC (Partially Error-Controlled Connection), is presented. PECC allows the application to parameterize the error-control policy in order to achieve responsiveness to the application-specific trade-off between packet loss on one hand and timeliness of packet delivery on the other.},
url = {papers/dempsey92.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, February 06, 2001 at 18:27:25 (CET)}
}
@article{Dempsey94c,
author = {B. Dempsey and M. Lucas and A. Weaver},
title = {An Empirical Study of Packet Voice Distribution over a Campus-Wide Network},
journal = {19th IEEE Local Computer Networks Conference},
year = {1994},
month = {October},
annote = {This paper provides an empirical investigation of the feasibility of transmitting real-time packet streams across a large extant campus network, the University of Virginia campus network. The network capability for supporting interactive packet voice, an important and demanding application domain, is evaluated.},
url = {papers/dempsey94.ps.gz},
bibdate = {Wednesday, February 07, 2001 at 08:05:43 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Diaz94b,
author = {M. Diaz et al.},
title = {POC: a New Concept for High Speed and Multimedia Service and Protocols},
journal = {Annales des télécoms},
year = {1994},
volume = {49},
number = {5-6},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 07, 2001 at 12:18:20 (CET)}
}
@article{Amer94b,
author = {P. Amer et al.},
title = {Partial-Order Transport Service for Multimedia and Protocols},
journal = {IEEE/ACM Transactions on Networking},
year = {1994},
volume = {2},
number = {5},
pages = {440-456},
month = {October},
bibdate = {Wednesday, February 07, 2001 at 12:25:30 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Lee00,
author = {K. Lee et al.},
title = {An Integrated Source Coding and Congestion Control Framework for Video Streaming in the Internet},
journal = {Procedings of the IEEE INFOCOM 2000},
year = {2000},
annote = {In this paper a framework for video transmission over the Internet that features the coordinated operation of an application-layer video source coding algorithm and a transport-layer rate control mechanism is presented. The proposed video coding scheme operates on a progressively encoded video stream and provides graceful resilience to network packet drops. The robustness is enabled through a generalized Multiple Description (MD) coding strategy, architected as an adaptive array of packet-erasure correction codes. The video coding algorithm is matched to an efficient and reactive rate control mechanism that minimizes the fluctuation of rate and uses the profile of past losses to adjust the rate in a TCP-friendly manner.},
url = {papers/lee00.ps},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 07, 2001 at 14:13:10 (CET)}
}
@article{Girod99,
author = {B. Girod et al.},
title = {Packet Loss Resilient Internet Video Streaming},
journal = {Proceedings of SPIE Visual Communications and Image Processing '99},
year = {1999},
pages = {833-844},
month = {January},
annote = {This paper describes a transmission scheme for Internet video streaming that provides an acceptable video quality over a wide range of connection qualities. The proposed system consists of a scalable video coder which uses a fully standard compatible H.263 coder in its base layer. The scalable video coder is combined with unequal error protection using Reed-Solomon codes applied across packets. We present and verify a two-state Markov model for packet losses over Internet connections. The relation between packet loss and picture quality at the decoder for an unequally protected layered video stream is derived. Experimental results show that, with our approach, the picture quality of a streamed video degrades gracefully as the packet loss probability of an Internet connection increases.},
url = {papers/gerod00.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 07, 2001 at 14:20:00 (CET)}
}
@article{Gong92b,
author = {F. Gong},
title = {A Transport Solution for Pipelined Network Computing},
journal = {Ph.D. Thesis, Washington University at St. Louis},
year = {1992},
month = {December},
annote = {Fengmin Gong's PH.D. thesis. Describes, among other things, the SSTP (Segment Streaming Transport Protocol) transport protocol.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 07, 2001 at 14:54:18 (CET)}
}
@article{Hess98,
author = {C. Hess},
title = {Media Streaming Protocol: An Adaptive Protocol for the Delivery of Audio and Video over the Internet},
journal = {Master's Thesis, Graduate College of Computer Science at the University of Illinois at Urbana-Champaign},
year = {1998},
annote = {This Master's thesis introduces a new protocol, MSP (Media Streaming Protocol), specifically targeting streaming video and audio over the Internet. MSP is a predecessor to VDP (Video Datagram Protocol) and consequently works on top of extant Internet transport protocols.},
url = {papers/hess98.pdf.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 07, 2001 at 16:15:56 (CET)}
}
@misc{RFC2884,
author = {J. Hadi Salim and U. Ahmed},
title = {{RFC 2884:} Performance Evaluation of Explicit Congestion Notification (ECN) in IP Networks},
year = {2000},
month = {July},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 07, 2001 at 20:11:13 (CET)}
}
@article{Davidsson00,
author = {J. Davidson and J. Peters},
title = {Voice over {IP} Fundamentals},
journal = {Cisco Press},
year = {2000},
month = {March},
annote = {This book gives a good, succinct coverage of the techniques behind VoIP. A basic introduction to telecommunications is included.},
bibdate = {Thursday, February 08, 2001 at 15:56:17 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Skoglund98,
author = {M. Skoglund and T. Ottosson},
title = {Soft multiuser decoding for vector quantization over a {CDMA} channel},
journal = {IEEE Transactions on Communications},
year = {1998},
volume = {46},
number = {3},
month = mar
}
@misc{Atmforum96,
title = {ATM Traffic Management Specification Version 4.0},
author = {{ATM Forum}},
month = {April},
year = 1996
}
@article{Gurewitz00,
author = {Omer Gurewitz and Moshe Sidi and Israel Cidon},
title = {The Ballot Theorem Strikes Again:Packet Loss Process Distribution},
journal = {IEEE Transactions on Information Theory},
year = {2000},
volume = {46},
number = {7},
pages = {2588-2595},
month = {November},
annote = {Provides a method for calculating multiple loss probabilities at bounded queues assuming Poisson arrival.},
url = {papers/Gurewitz00PacketLossDistrib.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, February 13, 2001 at 11:28:07 (CET)}
}
@article{JPEG2000fcd,
author = {Martin Boliek (ed.)},
title = {{JPEG} 2000 {Final Committee Draft}},
journal = {{ISO/IEC FCD15444-1}},
year = {2000},
month = {March},
annote = {A draft version of the JPEG2000 standard.},
url = {papers/fcd15444-1.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, February 13, 2001 at 12:29:10 (CET)}
}
@article{Atzori99,
author = {Luigi Atzori and Francesco De Natale},
title = {A Novel Method for the Recovery of Lost Visual Data using Correlated Edges Information},
journal = {Packet Video 99, New York},
year = {1999},
month = {April},
annote = {Uses edge detection detection to interpolate a "sketch" of the data in alost block. This sketch is used to reconstruct lost blocks with good quality although significant processing is required.},
url = {http://www.research.att.com/~mrc/pv99/CONTENTS/PAPERS/ATZORI/PV99.HTM},
submitter = {Johan Garcia},
bibdate = {Tuesday, February 13, 2001 at 15:17:36 (CET)}
}
@misc{rfc2460,
author="S. Deering and R. Hinden",
title={{Internet Protocol}, Version 6 {(IPv6)} Specification},
series="Request for Comments",
number="2460",
howpublished="RFC 2460 (Draft Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1998,
month=dec,
url="http://www.ietf.org/rfc/rfc2460.txt",
}
@article{Sternad01,
author = {M. Sternad and A. Svensson and A. Ahlen and T. Ottosson},
title = {{PCC} Wireless {IP} - Optimizing Throughput and {QoS} over Fading Channels},
journal = {Proceedings Nordic Radios Symposium (NRS01)},
year = {2001},
month = {April},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Tuesday, February 13, 2001 at 15:51:48 (CET)}
}
@article{Floyd98,
author = {Rick Floyd and Barron Housel and Carl Tait},
title = {Mobile Web Access using eNetwork Web Express},
journal = {IEEE Personal Communications},
year = {1998},
volume = {5},
number = {5},
pages = {47-52},
month = {October},
bibdate = {Tuesday, February 13, 2001 at 16:49:50 (CET)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Byers98,
author = {John W. Byers and Michael Luby and Michael Mitzenmacher and Ashutosh Rege },
title = {A digital fountain approach to reliable distribution of bulk data },
journal = {ACM SIGCOMM '98, Vancouver, Canada},
year = {1998},
pages = {56-67},
month = {September},
annote = {Provides a solution to reliable multicast using tornado FEC codes and a TCP friendly layered multicast protocol. Nicely written paper with a good explanation on tornado codes and their superiority over Reed-Solomon codes. },
url = {papers/Byers98_DigitalFountain.pdf},
bibdate = {Thursday, February 15, 2001 at 19:02:15 (CET)},
submitter = {Johan Garcia}
}
@article{Luby97,
author = {Michael G. Luby and Michael Mitzenmacher and M. Amin Shokrollahi and Daniel A. Spielman and Volker Stemann},
title = {Practical loss-resilient codes },
journal = {Proceedings of the 29'th ACM symposium on Theory of computing, El Paso, USA},
year = {1997},
pages = {150-159},
month = {May},
annote = {Provides the mathematical background for constructing tornado FEC code graphs as exemplified in Byers98. Tornado codes is a class of codes based on irregular, multilevel, sparse graphs, giving much better decoding time than Reed Solomon at the cost of a very slightly decrease in decoding efficiency.},
url = {papers/Luby97_TornadoMath.pdf},
submitter = {Johan Garcia},
bibdate = {Thursday, February 15, 2001 at 19:09:09 (CET)}
}
@article{Chuo00,
author = {Phil A. Chou and Alexander E. Mohr and Sanjeev Mehrotra and Albert Wang},
title = {FEC and Pseudo-ARQ for Receiver-driven Layered Multicast of Audio and Video},
journal = {Data Compression Conference (DCC 2000)},
year = {2000},
month = {March},
annote = {Theis paper describes the use of Mohrs FEC distribution algorithm to the layers of an multilayer encoding multicasted to many receivers. Separate multicast groups exist both for controlling the amount of source encoded data received as well as the amount of error correction. Each receiver subscribes to the multicast groups corresponding to its specific bandwidth and loss characteristics. Additionally, multicast groups containing delayed transmission of previously sent data can be joined by a receiver nedding a "retransmission", providing a "Pseudo-ARQ mechanism" that does not inflict any work or scalability problems at the server end. },
url = {papers/Chou00FECMultilayermulticast.ps.gz},
submitter = {Johan Garcia},
bibdate = {Friday, February 16, 2001 at 09:18:44 (CET)}
}
@article{Rizzo97a,
author = {Luigi Rizzo},
title = {Effective erasure codes for reliable computer communication protocols},
journal = {ACM Computer Communication Review},
year = {1997},
month = {April},
annote = {Describes the mathematics behind Reed-Solomon based FEC and provides a discussion of the performance and applications of the implementation of such a coder based on Vandermonde matrices. The code for this coder is available from Rizzo's homepage. },
url = {papers/Rizzo97FEC.ps},
bibdate = {Friday, February 16, 2001 at 11:00:22 (CET)},
submitter = {Johan Garcia}
}
@article{Rizzo97b,
author = {Luigi Rizzo},
title = {Dummynet: {A} simple approach to the evaluation of network protocols},
journal = {ACM Computer Communication Review},
year = {1997},
volume = {27},
number = {1},
pages = {31-41},
month = jan,
annote = {Describes the initial version of the Dummynet network emulator. The current version differs mainly since it uses the ipfw framework and has more configuration options. },
url = {papers/Rizzo97dummynet.ps.gz},
bibdate = {Friday, February 16, 2001 at 11:30:46 (CET)},
submitter = {Johan Garcia}
}
@article{Rizzo97c,
author = {Luigi Rizzo},
title = {An Embedded Network Simulator to Support Network Protocols' Development},
journal = {Proceedings of Tools '97, St.Malo, France},
year = {1997},
month = {June},
annote = {Much the same as the other paper describing Dummynet, but gives an example of the sysctl interface for setting up Dummynet.},
url = {papers/Rizzo97cDummynet.ps.gz},
submitter = {Johan Garcia},
bibdate = {Friday, February 16, 2001 at 11:47:41 (CET)}
}
@article{Ioannis99,
author = {Charitakis Ioannis and Papadakis Charalampos},
title = {Improving TCP Behavior},
journal = {M.Sc Thesis, KTH, Stockholm, Sweden},
year = {1999},
month = {August},
annote = {Provides an overview of TCP and suggests two sender side improvements to TCP. One suggested improvment is to send "new" packets after having sent the packet whose loss triggered the fast retransmit, instead of sending the following Go-back-N style. This approach is found to have some performace benefits, but can be sensitive to dupACK losses. The other idea is to use rate control to negate the grouping effect arising from an the arrival of a group of acks. When the sender paces the packets to be sent over the next RTT uniformly along its duration results in Jains fairness index beeing better, lower buffer usage, better goodput, lower losses, lower average RTT. These results were obtained borh for drop-tail and RED routers for the simple topologies used. When competing with non-rate controlled connections a rate-controlled connection seemed to suffer somewhat. ns2 was used for the experiments. Fredrik Orava was one advisor.},
url = {papers/Ioannis99_ImprovingTCPbehavior_MSthes.pdf },
bibdate = {Friday, February 16, 2001 at 12:42:11 (CET)},
submitter = {Johan Garcia}
}
@article{In99,
author = {J. In and S. Shirani and F. Kossentini},
title = {On RD optimized progressive image coding using JPEG},
journal = {IEEE Transactions on Image Processing},
year = {1999},
volume = {8},
number = {11},
pages = {1630-1638},
month = {November},
annote = {Describes an algorithm for selecting an optimal order of spectral selection and successive approximation steps for composing a standarp progressive image with a specified bitrate.},
url = {papers/In99RDoptimizedProgJPEG.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, February 16, 2001 at 14:02:49 (CET)}
}
@article{Merhav98,
author = {Merhav, Neri },
title = {Embedding Companders in {JPEG} Compression },
journal = {Technical report HPL-98-141, Hewlett-Packard},
year = {1998},
month = {September},
annote = {Provices formulas for obtaining a non-uniform quantizer that is provides better rate distorsion-performance than the uniform quantizer specified by the standard. Using non-uniform quanitizers (jmf u-law) is not part of the JPEG standard but can provide 10-25 % better compressions performance. The increased compression performance is however largest at low compresion levels and decreases as the image becomes more compressed. Compression below 1bpp for grayscale does not seem to benefit from non-uniform quantization. },
url = {papers/Mehrav98JPEGCompanding.ps.gz},
submitter = {Johan Garcia},
bibdate = {Monday, February 19, 2001 at 10:13:32 (CET)}
}
@article{Guo97,
author = {X. Guo and C. Pattinson},
title = {Quality of Service Requirements for Multimedia Communications},
journal = {Proceedings of the Time and the Web Symposium},
year = {1997},
month = {June},
annote = {This paper discusses the particular characteristics of multimedia traffic which make service provisioning so difficult. Current proposals to develop a relationsship between user perceptions of 'quality' and the factors which must be addressed by the service provider to deliver an acceptable level of service are examined. In addition, the authors discuss modelling techiques to develop a more complete picture of the QoS requirements for multimedia.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, February 20, 2001 at 09:19:35 (CET)}
}
@article{Deshpande99,
author = {Nachiket Deshpande},
title = {{TCP} Extensions for Wireless Networks},
journal = {Student Project Report, cis788-99, Ohio State University},
year = {1999},
annote = {Discussion of proposed schemas to improve performance of TCP in wireless networks, including satellite networks. Covers I-TCP, snoop, M-TCP, delayed dupack, fast retransmit, Mobile TCP, Multiple Acks, discriminating congestion losses from wireless losses and distinguishing losses by making two connections.},
url = {papers/deshpande_tcp_wireless.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Wednesday, February 21, 2001 at 13:17:42 (CET)}
}
@misc{Duchamp92,
title = {Measured performance of a wireless {LAN}},
author = {D. Duchamp and N. Reynolds},
year = {1992},
annote = {Measures performance over NCR WaveLAN 2Mbit in a 56 meter corridor. Characterizes errors as "capture without error", "capture with error" and "fail to capture". Concludes that errors tend to be non consecutive biterrors, certain bit error lengths are strongly perferred at all distances and sizes and that the number of errors per bit (error density) is roughly constant over all packet sizes and distances.},
url = {papers/wlan-measurement.ps},
submitter = {Stefan Alfredsson},
text = {D. Duchamp and N. F. Reynolds. Measured performance of a wireless LAN. In Proceedings of the 17th Conference on Local Computer Networks, pages 494--499. IEEE, Sep 1992.}
}
@article{Parsa99b,
author = {C. Parsa and J. Garcia-Luna-Aceves},
title = {Improving {TCP} performance over wireless networks at the link layer},
journal = {ACM Mobile Networks and Applications Journal, vol 5, issue 1.},
year = {2000},
annote = {Presents the transport unaware link improvement protocol, TULIP, which improves the performance of TCP ove lossy wireless links. Tailored for half-duplex radio links. Requires no base-station and keeps no TCP state. TULIP uses the generous timeouts in the transport protocol to perform link-level retransmissions. A simulation comparing TULIP to the Snoop protocol has been conducted, where the performace of TULIP provided improved throughput over the Snoop protocol. },
url = {papers/tulip.ps},
submitter = {Stefan Alfredsson},
text = {C.Parsa and J.J. Garcia-Luna-Aceves. Improving TCP performance over wireless networks at the link layer. ACM Mobile Networks and Applications Journal, vol 5, issue 1. Available at http://www.cse.ucsc.edu/research/ccrg.}
}
@article{Marasli96b,
author = {R. Marasli and P. Amer and P. Conrad},
title = {Optimizing Partially Ordered Transport Services for Multimedia Applications},
journal = {IEEE INFOCOM 96, San Fransisco},
year = {1996},
pages = {621-629},
bibdate = {Thursday, February 22, 2001 at 16:28:42 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Chan97,
author = {A Chan and Danny H.K. Tsang and S. Gupta, },
title = {Impacts of Handoff on TCP Performance in Mobile Wireless Computing},
journal = {IEEE Intl. Conf. on PersonalWireless Communications (ICPWC'97)},
year = {1997},
month = {December},
annote = {Discusses the causes of degraded TCP performance in connection with (long duration) wireless handoffs. Proposes two solutions: PROBE that make TCP handoff aware but requires bothe network, sender and receiver modification. BUFFER+FREEZE that hides the service outage from the sending TCP by letting a basestation modify the ACKs so that they show zero receiver window, in effect freezing the sender.},
url = {papers/Chan97TCPHandoffPerformance.ps},
bibdate = {Thursday, February 22, 2001 at 17:53:31 (CET)},
submitter = {Johan Garcia}
}
@article{Chan97a,
author = {Aldar Chan and D.H.K Tsang and S. Gupta},
title = {TCP over Wireless Links},
journal = {Proc VTC'97, Phoenix, USA},
year = {1997},
month = {May},
annote = {Provides a mathematical analysis of a simplified TCPs behaviour in a wireless setting as well as simulations of two TCP alterations, NACK and LHACK (Last hop ack). Some simulations seem to use a raw wireless channel without any link layer, which would be uncommon in practice. The NACK and LHACK schemes requires modification at both sender and receiver. NACK is a TCP option, essentialy SACK without congestion avoidance behaviour at the sender.},
url = {papers/Chan97aTCPoverWireless.ps.gz},
bibdate = {Thursday, February 22, 2001 at 18:52:21 (CET)},
submitter = {Johan Garcia}
}
@article{Brian93,
author = {B. Smith},
title = {Implementation Techniques for Continuous Media Systems and Applications},
journal = {Ph.D. Thesis, University of California, Berkeley},
year = {1993},
submitter = {Karl-Johan Grinnemo},
bibdate = {Friday, February 23, 2001 at 08:35:12 (CET)}
}
@inproceedings{Grinnemo01,
author = {K-J Grinnemo and A. Brunstrom},
title = {Evaluation of the {QoS} offered by {PRTP-ECN} - A {TCP}-Compliant Partially Reliable Protocol},
booktitle = {Proceedings IWQoS 2001, Karlsruhe, Germany},
year = {2001},
pages = {217-231},
month = {June},
url = {papers/grinnemo01.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, February 26, 2001 at 13:29:50 (CET)}
}
@misc{tcptrace,
author = {Shawn Ostermann},
title = {Tcptrace},
note = {\url{http://www.tcptrace.org/} Visited 2011-09-09},
submitter = {Annika Wennstrom; updated by Stefan Alfredsson 2011},
bibdate = {Tue Feb 27 13:13:17 CET 2001}
}
@misc{ethereal,
author = {Ethereal},
title = {A network protocol analyzer},
misc = {\url{http://www.ethereal.com/} Visited 2011-09-09},
bibdate = {Tue Feb 27 13:13:17 CET 2001},
submitter = {Annika Wennstrom}
}
@misc{wireshark,
author = {Wireshark},
title = {A network protocol analyzer},
misc = {\url{http://www.wireshark.org/} Visited 2011-09-09},
submitter = {Stefan Alfredsson}
}
@article{Shirani00,
author = {Shirani, S and Kossentini, F. and Ward, R. },
title = {Reconstruction of baseline {JPEG} coded images in error proneenvironments },
journal = {IEEE Transactions on Image Processing},
year = {2000},
volume = {9},
number = {7},
pages = {1292 - 1299},
month = {July},
annote = {The paper proposes a method for removing erroneous offsets from diff-encoded DC coefficients subject to losse. Also presents a reconstruction method based on linear combination of corresponding pixels from neighbouring blocks. (A refinement of Hemamis scheme). Quite good results presented, but also quite computationally complex. The advantage DC offset-removal part is not so useful since decoder resynchronization still has to be obtained, so standard JPEG images are not helped. Author Abstract: A two-stage method for the reconstruction of missing data in the transmission of baseline JPEG coded images in error prone environments is proposed. In the first stage, we estimate the values of the missing DC coefficients. As effects of errors in estimating the missing DC values will appear as a number of stripes across the image, a technique for removing such stripes is also developed. In the second stage, the data of missing blocks is reconstructed by exploiting the correlation between adjacent blocks. Simulation results intricate that our reconstruction method performs very well. The two key contributions of our method are that it does not assume nondifferential encoding of the DC coefficients, and that it performs well in the reconstruction of diagonal edges. },
url = {papers/Shirani00JPEGReconstruction.pdf },
submitter = {Johan Garcia},
bibdate = {Wednesday, February 28, 2001 at 12:34:05 (CET)}
}
@article{Ericsson98,
author = {Ericsson, Sweden},
title = {WCDMA Error Patterns at 64 kb/s},
journal = {Proc. ITU-T Study Group 16, Multimedia Terminals and Systems Experts Group, Doc. Q11-F-05 , Cannes, France},
year = {1998},
month = {June},
annote = {The archive contains error patterns with target BER of 10-3 and 10-4 for three different speeds (3,40,120 km/h) and a note including average burst length among other info.},
url = {papers/Ericsson98WCDMA_Error_Patterns.zip},
submitter = {Johan Garcia},
bibdate = {Wednesday, February 28, 2001 at 14:17:24 (CET)}
}
@article{Nokia99,
author = {Nokia, Finland},
title = {WCDMA Error Patterns },
journal = {Proc. ITU-T Study Group 16, Multimedia Terminals and Systems Experts Group, Doc. Q11-I-11, Monterey, CA, USA},
year = {1999},
month = {February},
annote = {Provides a set of simulated Wideband CDMA error patterns for error resilience simulations. The channel models used in generating the error patterns are based on the latest air interface specifications by ETSI. The major change since summer 1998 has been to replace convolutional codes by turbo codes at bit-rates >= 32 kbps. The reason behind that decision is the fact that turbo codes make the errors more bursty than convolutional codes do, and thus in the same bit error rate, typically better video quality can be achieved. The final parameters for turbo codes are still open. In addition to the bit rates who did not have any error patterns available, i.e., 128 and 384 kbps, the set covers also other important bit rates in third generation networks; 32 and 64 kbps, thus providing up-to-date patterns for 64 kbps, and new type of patterns for 32 kbps. Available from http://standards.pictel.com/ftp/lbc-site/LBCmobile/error_pattern/},
url = {papers/Nokia99WCDMA_turbo_error_patterns.zip },
submitter = {Johan Garcia},
bibdate = {Wednesday, February 28, 2001 at 14:30:43 (CET)}
}
@article{Ismaeil00,
author = {Ismaeil, I. and Shirani, S. and Kossentini, F. and Ward, R},
title = {An efficient, similarity-based error concealment method for block-based coded images},
journal = {Proc. Intnl. Conf. on Image Processing (ICIP00)},
year = {2000},
volume = {3},
pages = {388 - 391},
month = {September},
annote = {Suggest using a layer of 2-3 pixels around the edges of a lost block to search for the best matching area in an image and reproducing this area in the lost blocks place. Discusses a fast search algorithm based on initial downsampling and diamondshaped searching. Seems to be interesting but not much detail provided.},
url = {papers/Ismaeil00EffErrConcealBlockBased.pdf },
bibdate = {Wednesday, February 28, 2001 at 15:48:30 (CET)},
submitter = {Johan Garcia}
}
@article{Roberts01,
author = {J. Roberts},
title = {Traffic Theory and the Internet},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {1},
pages = {94-99},
month = {January},
annote = {In this paper, the author demonstrates the usefulness of applying traffic theory for dimensioning a network. IP traffic resulting from the activity of a large population of users can be represented for performance prediction purposes as a stationary stochastic process. This process can be modeled most conveniently at flow level distinguishing the two main categories of elastic and streaming traffic. An interesting observation the author make is that service differentiation is advantegous only in those cases we have classes of traffic with qualitatively different QoS requirements. It is much less obvious that it is useful in those cases the classes represent the same type of traffic and only differs in the degree of quality, e.g. lower packet loss or higher throughput.},
bibdate = {Thursday, March 01, 2001 at 14:07:03 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Shirani99,
author = {Shirani, S. and Kossentini, F. and Ward, R.},
title = {Error concealment methods, a comparative study },
journal = {Proc. 1999 IEEE Canadian Conference on Electrical and Computer Engineering},
year = {1999},
volume = {2},
pages = {835 - 840},
month = {May},
annote = {This paper discusses various methods for image and video error concealment and classifies them based on the assumptions made. The methods are classified into classes: smoothness, Statistical correlaiton, Edge continuity, fractal behavior and particle behavior (i.e. smoothness). The complexity of these methods are also discussed in general terms. The paper states that DCT coefficients in a lost block is likely to be close in value to the corresponding DCT coefficient in a spatially adjacent block, which is debatable. },
url = {papers/Shirani99ErrConcealCompariStud.pdf },
bibdate = {Thursday, March 01, 2001 at 15:15:32 (CET)},
submitter = {Johan Garcia}
}
@article{Shirani98,
author = {Shirani, S. and Kossentini, F. and Ward, R. },
title = {Packet loss concealment in baseline JPEG coded images },
journal = {IEEE Symposium on Advances in Digital Filtering and Signal Processing},
year = {1998},
pages = {16-19},
month = {June},
annote = {Essentially a shorter version of the Shirani00 paper, the same idea to remove stipes due to DC-coefficient diff encoding offsets after errors coupled with a linear combination reconstruction improved to better handle diagonal edges. States as a key contribution the possiblity of using diff-encoded DC coefficients, but decoder resynchronization must still be achieved which is not covered.},
url = {papers/Shirani98PacketLossConcealJPEG.pdf},
submitter = {Johan Garcia},
bibdate = {Thursday, March 01, 2001 at 15:31:03 (CET)}
}
@article{Cote99,
author = {Cote, G. and Erol, B. and Gallant, M. and Kossentini, F. },
title = {H.263+: video coding at low bit rates},
journal = {IEEE Transactions on Circuits and Systems for Video Technology},
year = {1998},
volume = {8},
number = {7},
pages = {849 - 866 },
month = {November},
annote = {Abstract: We discuss the ITU-T H.263+ (or H.263 Version 2) low-bit-rate video coding standard. We first describe, briefly, the H.263 standard including its optional modes. We then address the 12 new negotiable modes of H.263+. Next, we present experimental results for these modes, based on our public-domain implementation (http:ilspmg.ece.ubc.ca) (JG:no longer avilable) . Tradeoffs among compression performance, complexity, and memory requirements for the H.263+ optional modes are discussed. Finally, results for mode combinations are presented. },
url = {papers/Cote98h263plusoverview.pdf},
bibdate = {Thursday, March 01, 2001 at 15:39:40 (CET)},
submitter = {Johan Garcia}
}
@article{Hamann99,
author = {T. Hamann and J. Walrand},
title = {A New Fair Window for {ECN} Capable {TCP (New-ECN)}},
journal = {Tech. Rep. Memorandum No. UCB/ERL M99/35, Electronics Research Laboratory, College of Engineering, UC Berkeley},
year = {1999},
month = {June},
annote = {In this paper, a new congestion control algorithm which addresses the bias against connections with long round-trip times in TCP is presented. The idea is to prevent a fast connection from opening its congestion window too quickly and to enable a slow connection to open its window more aggressively. The algorithm is is called "New-ECN TCP-Reno Algorithm", and as implied by the name, it involves ECN. During slow-start, the New-ECN algorithm works in the same way as TCP-Reno, but during congestion avoidance, the rate at which the congestion window increases depends on the round-trip time. A longer round-trip time leads to a larger increment than a smaller one. The New-ECN algorithm also reacts differently to congestion during congestion avoidance than TCP-Reno. It distinguish between ECN-Echo packets, duplicate ACKs, and retransmit timeouts. If congestion is signalled by duplicate ACKs or timeout, New-ECN works the same way as TCP-Reno. However, congestion signalled by an ECN-Echo packet do not lead to the congestion window being halved. Instead, the reduction is far less, e.g. 10%. With a variety of simulation scenarios, they demonstrate that the New-ECN algorithm achieves a fair sharing of bandwidth. },
url = {papers/hamann99.ps.gz},
bibdate = {Monday, March 05, 2001 at 09:56:32 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Allman97,
author = {M. Allman and C. Hayes and H. Kruse and S. Ostermann},
title = {{TCP} Performance over Satellite Links},
journal = {Proceedings of the 5th International Conference on Telecommunication Systems},
year = {1997},
month = {March},
annote = {The experiments presented in this paper indicate that a version of TCP using slow start, congestion avoidance, fast retransmit, fast recovery, and the window scale option can allow an application that uses TCP to achieve high throughput over satellite links. When network congestion and higher bit error rates are present, the addition of selective acknowledgements (SACK) can also improve performance in some cases. Two TCP mechanisms appear to have a limiting effect on TCP's satellite performance: slow start and congestion avoidance.Slow start over satellite links takes approximately six seconds to reach maximum throughput. When lost segments trigger avoidance, the resulting throughput decrease can continue as long as several minutes.},
url = {papers/allman97.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, March 05, 2001 at 10:17:14 (CET)}
}
@article{Gurtov00,
author = {A. Gurtov},
title = {TCP Performance in the Presence of Congestion and Corruption Losses},
journal = {Master's Thesis, University of Helsinki, Department of Computer Science, Helsinki},
year = {2000},
month = {December},
annote = {This master's thesis is written within the IIP-Mobile project. The author has performed an experimental evaluation of TCP in an emulated wireless environment. Experiments with TCP connections with different values of the initial window, receiver window, with or without SACK and New Reno have been done. One result was that TCP with SACK performed significantly better than other modifications under all conditions. Linux TCP and the Seawind network emulator was used for the experiments. The thesis also describes properties of a wireless link, problems with TCP performance and suggested enhancements, in addition to the experiments and their results. A section that describes discovered problems in Linux TCP is included as an appendix in the thesis },
submitter = {Katarina Asplund},
bibdate = {Thursday, March 22, 2001 at 16:01:10 (CET)}
}
@article{Padhye01,
author = {J. Padhye and S. Floyd},
title = {Identifying the TCP Behavior of Web Servers },
journal = {ICSI Technical Report 01-002},
year = {2001},
month = {February},
annote = {The author has developed a tool called TCP Behavior Identification Tool (TBIT) to characterize the TCP behavior of a remote web server. The paper presents results about the TCP behaviors of major web servers, obtained using this tool. Behaviors that were tested was size of initial window, version of congestion control algorithm, window reduction after a packet loss, usage of SACK, usage of ECN, and time wait duration.},
submitter = {Katarina Asplund},
bibdate = {Thursday, March 22, 2001 at 16:39:12 (CET)}
}
@article{Floyd01,
author = {S. Floyd},
title = {A Report on Some Recent Developments in TCP Congestion Control },
journal = {IEEE Communications Magazine},
year = {2000},
volume=39,
pages={84--90},
annote = {This paper discusses several changes to TCP's congestion control, either proposed or in progress. These changes include the limited Transmit mechanism to avoid unnecessary Retransmit Timeouts, and D-SACK-based mechanisms to identify and reverse unnecessary congestion control responses to reordered or delayed packets. The paper also discusses corruption notification messages from the link layer to the transport layer ECN.},
bibdate = {Thursday, March 22, 2001 at 16:51:26 (CET)},
submitter = {Katarina Asplund}
}
@article{Schulzrinne01,
author = {H. Schulzrinne and J. Rosenberg},
title = {The Session Initiation Protocol: Internet-Centric Signaling},
journal = {IEEE Communications Magazine},
year = {2000},
volume = {38},
number = {10},
pages = {134-141},
month = {October},
annote = {The paper gives an overview of the SIP protocol where the main protocol features are summarized. A range of extensions currently being discussed within the IETF are also discussed.},
submitter = {Katarina Asplund},
bibdate = {Thursday, March 22, 2001 at 17:00:22 (CET)}
}
@article{Grinnemo01a,
author = {K-J Grinnemo},
title = {Simulation Evaluation of {PRTP-ECN} - Simulation Plan, Version 1},
journal = {Internal},
year = {2001},
month = {January},
annote = {Simulation plan for the simulation evaluation of PRTP-ECN documented in the paper presented at IWQoS 2001 (sim1-3-1).},
url = {papers/grinnemo01a.ps.gz},
bibdate = {Wednesday, April 04, 2001 at 10:04:26 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Montgomery00,
author = {D. Montgomery},
title = {Design and Analysis of Experiments},
journal = {John Wiley \& Sons, Inc.},
year = {2000},
month = {June},
annote = {Textbook in statistical design of experiments.},
bibdate = {Thursday, April 05, 2001 at 16:47:21 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Wright99,
author = {G. Wright and W. Stevens},
title = {{TCP/IP} Illustrated, Volume 2: The Implementation},
journal = {Addison-Wesley},
year = {1999},
month = {December},
annote = {This book contains a thorough explanation of how the 4.4BSD-Lite release of TCP is implemented.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, April 09, 2001 at 15:25:17 (CEST)}
}
@article{Wu00,
author = {D. Wu and Y. Hou and Y. Zhang},
title = {Transporting Real-Time Video over the Internet: Challenges and Approaches},
journal = {Proceedings of the IEEE},
year = {2000},
volume = {88},
number = {12},
pages = {1853-1875},
month = {December},
annote = {Delivering real-time video over the Internet is an important component of many Internet multimedia applications. However, the current Internet does not offer any QoS guarantees to video transmission over the Internet. In this paper, the authors present a holistic approach to QoS over the Internet. Their framework comprises two components: congestion control and error control. For the design of each of these two components, a survey and classification of contemporary research is performed.},
bibdate = {Tuesday, April 17, 2001 at 08:03:47 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Digital01,
author = {Digital Fountain},
title = {A Quantum Leap in Content Delivery: Digital Fountain's Meta-Content Technology},
journal = {White Paper published by Digital Fountain},
year = {2001},
annote = {This paper introduces Digital Fountain's technology for solving the reliability and scalability problems associated with distribution of files and streaming media over the Internet. The Fountain Server architecture consists of two main functional components: a meta-content engine, and a replication system. The key to the architecture's scalability and efficiency is the concept of meta-content: For each piece of content, the meta-content engine generates a stream of meta-content, which consists of mathematical metaphors that describe the original content. Meta-content has the following fundamental properties: - they are independently generated, i.e. there are no interrelationship between the meta-content produced for two consecutive packets. - an exact copy of the original content can be generated from any portion of meta-content of sufficient length, i.e. it is not which packets that is received that matters, but how many. The underlying coding technology behind the meta-content is the, so called, Luby-transform. Conceptually, the Luby-transform generates meta-content as follows: the content is first logically partitioned into a number of equal length pieces. The partition is performed using combinatorial mathematics. The meta-content pieces could be viewed as random linear equations over the pieces of the original content. Once a client receives enough of these equations, it can solve them to reconstruct the pieces. The heart of the Luby-transform is the structure of the meta-content equations making them linerarly independent. Multicast clients are able to directly receive the meta-content packets generated by the meta-content engine. However, meta-content sent to unicast clients cannot be delivered directly. Instead, unicast clients subscribes to the replication system which uses unicast UDP for packet distribution. Digital Fountain has also designed a new congestion control protocol. Their protocol is called Fair Layered Increase Decrease (FLID). FLID was designed with the following goals: - Scalable. The sever should not se any difference when the number of clients changes. - Receiver-driven. - Fair to contending flows. FLID is a receiver-driven, end-to-end congestion control protocol that does not require any changes to existing routers. In FLID, the server sends a separate stream of meta-content packets generated for a piece of content to several multicast groups called "layers". A client independently adjusts its reception rate in response to network conditions by joining and leaving the layers. In periods of no congestion, a client increases its reception rate by joining new layers. The client decreases its reception rate in response to congestion by leaving layers. },
url = {papers/Digital01.pdf.gz},
bibdate = {Thursday, April 19, 2001 at 10:00:53 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Chandra01,
author = {P. Chandra et al.},
title = {Darwin: Customizable Resource Management for Value-Added Network Services},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {1},
pages = {22-35.},
month = {January},
annote = {In this article, a resource management system called Darwin is presented. Darwin differs from most other resource management systems in that it takes an integrated approach to resource management. By packaging storage/computation services together with communication services, Darwin is able to provide sophisticated services like intelligent caching, video/audio transcoding and mixing, virtual private networking, virtual reality games, data mining etc. The Darwing system consists of four interrelated resource management mechanisms: resource brokers called Xena, runtime resource management using so called delegates, H-FSC (Hierarchical Fair Service Curve) schedulers, and a resource allocation protocol called Beagle. The key property of these four mechanisms is that they can be customized according to service-specific needs. A proof-of-concept prototype of Darwin has been implemented. Although rudimentary, a series of experiments with the Darwin prototype clearly shows the advantages of taking this integrated approach to resource management.},
url = {papers/chandra01.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, April 26, 2001 at 07:58:35 (CEST)}
}
@article{Wu01,
author = {D. Wu and Y. Hou and Y. Zhang},
title = {Scalable Video Coding and Transport over Broadband Wireless Networks},
journal = {Proceedings of the IEEE},
year = {2001},
volume = {89},
number = {1},
month = {January},
annote = {In recent years, we have witnessed a rapid growth of research and development to provide mobile users with video communication through wireless media. In this paper, the authors examine the challenges in QoS provisioning for wireless video transport. To address the challenges, three techniques have been studied individually: scalable video coding, network-aware adaptation of end systems, and QoS support from the network. The main contribution from this paper is to unify the three techniques simultaneously, and present and present an adaptive framework which specifically addresses scalable video transport over wireless networks.},
bibdate = {Wednesday, May 02, 2001 at 09:30:00 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Wu01a,
author = {X. Wu and S. Cheng and Z. Xiong},
title = {On Packetization of Embedded Multimedia Bitstreams},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {132--140},
month = {March},
annote = {In this paper, algorithms are proposed that minimizes packetization inefficiencies due to bitstream alignment. Packetization schemes are studied against both high and low packet-drop rates. The key result of the paper i a general dynamic programming paradigm to design globally optimal packetization schemes for minimum distortion. Since the globally optimal packetization schemes sometimes are too computationally intensive, some suboptimal schemes are also presented.},
bibdate = {Wednesday, March 13, 2002 at 08:42:05 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{McDonald91,
author = {C. McDonald},
title = {A Network Specification Language and Execution Environment for Undergraduate Teaching},
journal = {Proceedings of the ACM Computer Science Education, Technical Symposium '91, San Antonio, Texas},
year = {1991},
pages = {25-34},
month = {March},
annote = {This paper describes the design and implementation of a network simulation environment, cnet, targeting undergraduate education. It concludes with a discussion of experiences gained using the csnet simulator in an undergraduate course at the University of Western Australia. Even though, the csnet simulator is indended for students, it has been used by researchers too. For example, a researcher, Aric Stewart, has conducted extensive studies on distance vector routing using cnet.},
url = {papers/mcdonald91.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, May 03, 2001 at 09:03:36 (CEST)}
}
@article{Dixit01,
author = {S. Dixit and Y. Ye},
title = {Streamlining the Internet-Fiber Connection},
journal = {IEEE Spectrum},
year = {2001},
pages = {52-57},
month = {April},
annote = {In this article, the authors contemplate on how TCP/IP will be able to accommodate the traffic volume in the future. As they see it, the most feasible solution for the future multimedia traffic is wavelength-division multiplexing (WDM) with TCP/IP atop. We have already today WDM-systems able to handle 10 Gb/s, and systems rated at 40 Gb/s are undergoing field testing. In order to support QoS, the authors argue that multiprotocol lambda switching together with multiprotocol label switching (MPLS) should be employed to enable efficient traffic control.},
bibdate = {Monday, May 14, 2001 at 09:11:40 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Bos01,
author = {L. Bos and S. Leroy},
title = {Toward an All-IP-Based UMTS System Architecture},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {1},
pages = {36-45},
month = {January},
annote = {In 1998, European standard bodies like ARIB, T1, TTA, started to standardize the third-generation mobile system in a group called 3GPP (Third Generation Partnership Project). A year later, US, got involved in the standardization by establishing a new forum, 3GPP2. Since mid-1990, two major trends have appeared in 3GPP. The first trend was the shift toward an all-IP UMTS network architecture, and the second trend was the evolution toward an open service architecture. One of the main incentives for introducing an open service architecture was that it in the long run would enable users to port telecommunications services between networks and terminals. In 3GPP, this concept was called the virtual home environment (VHE). There are two possible scenarios for introducing VHE in UMTS. First, it could be done using the existing IN infrastructure (IN = Intellogent Network, i.e. telecommunications services). Second, you could open up the existing infrastructure for third-party companies, and let the IN-services in UMTS reside on their platsforms, i.e. outside the perimeters of the network itself. This article gives a detailed treatment of these two scenarios, and their implications.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, June 28, 2001 at 09:32:57 (CEST)}
}
@article{Becher01,
author = {R. Becher and M. Dillinger and M. Haardt and W. Mohr},
title = {Broad-Band Wireless Access and Future Communication Networks},
journal = {Proceedings of the IEEE},
year = {2001},
pages = {58-75},
month = {January},
annote = {The UMTS Forum expects that mobile multimedia services will create around 60\% of the whole traffic in Europe (in terms of transmitted bits) by 2010. Future communication systems, beyond the third generation, will be characterized by a horizontal communication model between different access technologies such as cellular, cordless, WLANs, short-range connectivity, broadcast systems, and wired systems. These systems will be connected to a common, flexible, and seamless IP core network. The different access systems are organized in a layered structure. Interworking between the different systems are accomplished by horizontal handover within an access system, and by vertical handover between different access systems. The layered structure comprises the following five layers: distribution layer (broadcasting systems, satellite systems etc.), cellular layer (mobile radio systems), hot spot layer (WLAN-technologies), personal network layer (Bluetooth, DECT etc.), wired (fixed) layer (optical fiber, twisted-pair etc.). In this article, technologies pertaining to each of these layers are surveyed.},
bibdate = {Thursday, June 28, 2001 at 10:02:50 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Cao01,
author = {Y. Cao and V. Li},
title = {Scheduling Algorithms in Broad-Band Wireless Networks},
journal = {Proceedings of the IEEE},
year = {2001},
pages = {76-87},
month = {January},
annote = {Scheduling algorithms that support quality of service differentiation and guarantees for wireless data networks are crucial to the development of broad-band wireless networks. The characteristics of wireless communication pose special problems that do not exist in wireline networks. These include: 1) high error rate and bursty errors; 2) location-dependent and time-varying wireless link capacity; 3) scarce bandwidth; 4) use mobility; and 5) power constraint of the mobile hosts. All of the above charactersitics make developing efficient and effective scheduling algorithms for wireless networks very challenging. This paper presents a comprehensive survey and in-depth discussion on packet scheduling algorithms in wireless networks. The focus of the paper is on cell-structured wireless networks, and the aspects studied are transmission link variability, fairness, QoS, data throughput and channel utilization, and power constraint and simplicity.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, June 28, 2001 at 11:36:20 (CEST)}
}
@article{Fricker98,
author = {M. Fricker and A. Eberlein and S. Littlejohns and R. Workman and F. Halsall},
title = {Quality of Service in Multimedia Networks},
journal = {Proceedings of the First GEMISIS 2000 Symposium on Multimedia Networking Technology},
year = {1998},
month = {May},
annote = {This paper investigates three avaiable approaches to provide various degrees of QoS: the approach taken by ATM, reservation-based hard QoS guarantees, and the soft-QoS approach used with differentiated services. The ATM approach can provide the necessary bandwidth and hard QoS guarantees to support a wide variety of multimedia applications. However, its implementation has proven to be very expensive, making it most suited for use as a backbone technology. Similar hard QoS guarantees can be offered by RSVP in IP-networks. Concerns have been expressed though, over the complexity and scalability of RSVP. The differentiated services approach attempts to provide scaleable service discrimination in the Internet through fairly simple extensions to current Internet technology. However, differentiated services only provide soft QoS guarantees which may be unsuitable for some real-time applications.},
url = {papers/fricker98.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, June 28, 2001 at 12:46:41 (CEST)}
}
@article{McKinley00,
author = {P. McKinley and S. Gaurav},
title = {Experimental Evaluation of Forward Error Correction on Multicast Audio Streams in Wireless LANs},
journal = {Proceedings of the ACM Multimedia 2000 Conference},
year = {2000},
pages = {416-418},
month = {November},
annote = {This paper describes an experimental study of a proxy server to enhance interactive multicast audio streams when transmitted across wireless LANs. The main contribution of the paper is to evaluate the effectiveness of forward error correcting codes on improving the quality of audio channels for collaborating mobile users.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, June 28, 2001 at 13:52:47 (CEST)}
}
@article{Kleijnen01,
author = {J. Kleijnen},
title = {Experimental Designs for Sensitivity Analysis of Simulation Models},
journal = {Tutorial at the Eurosim 2001 Conference},
year = {2001},
month = {June},
annote = {This introductory tutorial gives a survey on the use of statistical designs for sensitivity analysis (what-if) in simulation. The author emphasizes on the need to apply design of experiments on simulations, and use ANOVA-techniques when analyzing results from simulations. Three regression models are introduced: 1) the main-effects model (a first-order polynomial); 2) the main-effects model biased by interactions (two-factor interactions); 3) the quadratic-effects model (a second-order polynomial).},
bibdate = {Thursday, June 28, 2001 at 14:22:43 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Ewerlid01,
author = {A. Ewerlid},
title = {Reliable Communication over Wireless Links},
journal = {Proceedings of Nordic Radio Symposium 2001},
year = {2001},
month = {April},
annote = {Communication over wireless links is often characterized by sporadic high bit error rates, and intermittent connectivity due to handoffs. TCP performance in such networks suffers from significant throughput degradation and very high interactive delays. In this paper, schemes to improve the performance of TCP in such networks are discussed, and a novel solution is proposed: the double-split connection. The double-split connection is a variant of the split connection. The first split should be at the base gateway. At this gateway, the TCP packets should be converted to some special wireless protocol. After having been trasferred over the wireless link, the wireless protocol packets are converted back to TCP at the second split-point, which resides in the wireless device. The major advantages with this scheme are: 1) delays that occur due to fadings over the wireless link can be hidden from sender; 2) explicitly feed back information of the channel conditions to the gateways at the wireless link edges; 3) not dependent on future changes to the TCP protocol.},
url = {papers/ewerlid01.pdf.gz},
bibdate = {Monday, July 02, 2001 at 08:51:27 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Liu98,
author = {C. Liu},
title = {Multimedia Over IP: RSVP, RTP, RTCP, RTSP},
journal = {Technical Report, Ohio State University},
year = {1998},
month = {January},
annote = {RSVP, RTP, RTCP, and RTSP are the foundation of real-time services in the Internet. This paper is a detailed survey of these four related protocols. I found this paper very useful; Especially the compilation of references at the end of the paper.},
url = {http://www.cis.ohio-state.edu/~cliu/ipmultimedia/},
bibdate = {Tuesday, July 03, 2001 at 08:21:04 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Steinmetz95,
author = {R. Steinmetz and K. Nahrstedt},
title = {Multimedia: Computing Communications and Applications},
journal = {Prentice Hall},
year = {1995},
annote = {This book aims to achieve a complete and balanced view on the multimedia field covering three main domains: devices, systems and applications.},
bibdate = {Wednesday, July 04, 2001 at 08:22:06 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Mukherjee00,
author = {B. Mukherjee and T. Brecht},
title = {Time-lined {TCP} for the {TCP}-friendly Delivery of Streaming Media},
booktitle = {Proceedings of the International Conference on Network Protocols (ICNP)},
address={Osaka Japan},
year = {2000},
pages = {165-176},
month = nov,
annote = {This paper introduces the Time-lined TCP (TLTCP) transport protocol: a TCP-friendly transport protocol for delivery of streaming media. TLTCP retains most of TCP's congestion control, however introduces deadlines. Each section of data is associated with a deadline by which it should be sent. The sender maintains a linked-list, called time-line list, that stores the deadlines for the time-lined data. The sender performs data sends as a normal TCP sender would until the expiry of the lifetime timer which indicates that the deadline for the current section of data has expired. It then selects the next section of data to be sent from the time-line list and sets the lifetime timer to the deadline for this section. All data up to the lowest sequence number of the new section of data is discarded. Extensive simulations using ns show that TLTCP under a wide range of network conditions share bandwidth equally with competing TCP flows and performs time-lined data delivery.},
url = {papers/mukherjee00.pdf.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, July 05, 2001 at 09:09:22 (CEST)},
journal = {International Conference on Network Protocols (ICNP), Osaka Japan}
}
@misc{CAIDA99,
title = {Computer Association for Internet Data Analysis ({CAIDA}) - Traffic Workload Overview (June 1999)},
key = {CAIDA99},
journal = {http://www.caida.org/outreach/resources/learn/trafficworkload/tcpudp.xml},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thu Jul 5 10:20:09 MDT 2001}
}
@article{Byers01,
author = {J. Byers and M. Luby and M. Mitzenmacher},
title = {Fine-Grained Layered Multicast},
journal = {Proceedings of INFOCOM '01},
year = {2001},
month = {April},
annote = {Traditional approaches to receiver-driven layered multicast have advocated the benefits of cumulative layering, which can enable coarse-grained congestion control that complies with TCP-friendliness equations over large time scales. In this paper, the authors propose a new family of non-cumulative layered congestion control schemes that base their congestion control upon Fibonacci-based number series. They prove, mathematically, that this family of congestion schemes offer a fine-grained congestion control. They support their findings with simulations using ns-2.},
url = {papers/byers01.pdf.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, July 09, 2001 at 09:15:54 (CEST)}
}
@article{Mena00,
author = {A. Mena and J. Heidemann},
title = {An Empirical Study of {Real Audio} traffic},
journal = {Proceedings of the IEEE Infocom '00},
year = {2000},
pages = {101-110},
month = {March},
bibdate = {Tuesday, July 10, 2001 at 11:38:18 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{realnetworks,
author = {{Real Networks}},
title = "{Real Player}",
journal = {http://www.real.com/},
submitter = {Karl-Johan Grinnemo}
}
@misc{mediaplayer,
author = {{Microsoft Corporation}},
title = {Microsoft Windows Media Player},
note = {\url{http://www.microsoft.com/windows/mediaplayer} Visited 2011-09-09},
submitter = {Karl-Johan Grinnemo; updated by Stefan Alfredsson}
}
@article{Bansal01a,
author = {D. Bansal},
title = {Congestion Control for Streaming Video and Audio Applications},
journal = {Master's thesis, Massachusetts Institute of Technology (MIT)},
year = {2001},
month = {January},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, July 10, 2001 at 11:50:12 (CEST)}
}
@article{Bansal01b,
author = {D. Bansal and H. Balakrishnan and S. Floyd and S. Shenker},
title = {Dynamic Behavior of Slowly-Responsive Congestion Control Algorithms},
journal = {Proceedings of ACM SIGCOMM '01 (To appear)},
year = {2001},
month = {September},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, July 10, 2001 at 12:05:43 (CEST)}
}
@inproceedings{Piecuch00,
author = {M. Piecuch and K. French and G. Oprica and M. Claypool},
title = {A Selective Retransmission Protocol for Multimedia on the {I}nternet},
booktitle = {Proceedings SPIE Multimedia Systems and Applications},
year = {2000},
pages = {79-90},
month = {November},
annote = {Katarina: This paper proposes the Selective Retransmission Protocol (SRP). SRP is partially reliable protocol which retransmits only a percentage of the data that was lost, providing a compromise between TCP and UDP. The amount that is retransmitted depends on several QoS factors including current loss and latency, round-trip time, network congestion and the desired quality requested by the user. Experiments show that SRP outperformed both TCP and UDP in all network conditions except over a low loss, high latency network, in which UDP did slightly better.},
bibdate = {Tuesday, July 10, 2001 at 17:54:25 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Li97,
author = {X. Li and S. Paul and P. Pancha and M. Ammar},
title = {Layered Video Multicast with Retransmission (LVMR): Evaluation of Error Recovery Schemes},
journal = {Proceedings of the Sixth International Workshop on Network and Operating System Support for Digital Audio and Video},
year = {1997},
month = {May},
bibdate = {Tuesday, July 10, 2001 at 18:03:43 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Floyd00,
author = {S. Floyd and M. Handley and J. Padhye and J. Widmer},
title = {Equation-Based Congestion Control for Unicast Applications},
journal = {Proceedings of SIGCOMM 2000},
year = {2000},
volume = {30},
number = {4},
pages = {43-56},
month = {August},
bibdate = {Wednesday, July 11, 2001 at 10:11:52 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Mukherjee00b,
author = {B. Mukherjee},
title = {Time-Lined TCP: a Transport Protocol for Delivery of Streaming Media over the Internet},
journal = {Master's thesis, University of Waterloo, Ontario, Canada},
year = {2000},
bibdate = {Wednesday, July 11, 2001 at 12:23:21 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Bansal00,
author = {D. Bansal and H. Balakrishnan},
title = {TCP-friendly Congestion Control for Real-time Streaming Applications},
journal = {MIT Technical Report, MIT-LCS-TR-806},
year = {2000},
month = {May},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, July 11, 2001 at 12:28:44 (CEST)}
}
@article{Handley01,
author = {M. Handley and J. Padhye and S. Floyd},
title = {{TCP} Friendly Rate Control {(TFRC)}: Protocol Specification},
journal = {Work in progress (Internet-Draft draft-ietf-tsvwg-tfrc-02.txt)},
year = {2001},
month = {May},
submitter = {Karl-Johan Grinnemo}
}
@article{Ramesh99,
author = {S. Ramesh and I. Rhee},
title = {Issues in Model-Based Flow Control},
journal = {Technical Report TR-99-15, Department of Computer Science, North Carolina State University},
year = {2000},
bibdate = {Thursday, July 12, 2001 at 08:41:24 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Jorguseski01,
author = {L. Jorguseski and J. Farserotu and R. Prasad},
title = {Radio Resource Allocation in Third-Generation Mobile Communication Systems},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {2},
pages = {117-123},
month = {February},
annote = {This paper addresses radio resource allocation in WCDMA and time-division CDMA systems. A set of resource allocations algorithms is proposed that consist of resource estimation, QoS scheduling, and power and rate allocation.},
bibdate = {Monday, July 30, 2001 at 15:43:01 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Dixit01b,
author = {S. Dixit and Y. Guo and Z. Antoniou},
title = {Resource Management and Quality of Service in Third-Generation Wireless Networks},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {2},
pages = {125-133},
month = {February},
annote = {In this article, the various evolution scenarios from 2G to 3G networks are discussed. This is followed by a discussion of the UMTS QoS architecture. The article concludes with a discussion of how QoS is to be achieved in UTRAN and the UMTS core network.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, July 30, 2001 at 15:53:17 (CEST)}
}
@article{Kuo01,
author = {G. Kuo and P. Ko and M. Kuo},
title = {Probabilistic Resource Estimation and Semi-Reservation Scheme for Flow-Oriented Multimedia Wireless Networks},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {2},
pages = {135-141},
month = {February},
annote = {One of the most important and at the same time most complicated issue with 3G wireless networks are QoS guarantees. Two common approaches to resource reservation in wireless networks are: adaptive resource reservation and resource reservation using the shadow cluster concept. However both of these schemes result in low bandwidth utilization. In this article, a new resource estimation scheme is proposed: a semi-reservation scheme. It differs from adaptive resource reservation in that bandwidth is reserved in neighboring cells in proportion to the probability the mobile host will move there. Simulations show that both the connection blocking probability as well as the connection dropping probability decreases with the semi-reservation scheme compared to the adaptive scheme.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, July 31, 2001 at 10:02:55 (CEST)}
}
@article{Chakrabarti01,
author = {S. Chakrabarti and A. Mishra},
title = {QoS Issues in Ad Hoc Wireless Networks},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {2},
pages = {142-148},
month = {February},
annote = {Ad hoc networks consist of mobile nodes interconnected by multihop communication paths. Unlike conventional wireless networks, ad hoc networks have no fixed network infrastructure or administrative support. This article addresses some of the problems concerning QoS one faces with ad hoc networks. It gives a brief introduction to the area of ad hoc networks and points out issues which need to be further studied.},
bibdate = {Tuesday, July 31, 2001 at 10:14:24 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Hu01,
author = {Y. Hu and V. Li},
title = {Satellite-Based Internet: A Tutorial},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {3},
pages = {154-162},
month = {March},
annote = {This article gives a succinct introduction to satellite-based Internet systems. Satellite systems is not a homogeneous group. It comprises everything from geostationary satellites to low earth orbit satellites. The article covers in a very balanced way, all these systems and pertaining problems. One major problem is medium access control. A quite thorough treatment of MAC protocols for satellite links are given. An emphasis is on discussing open issues. An attractive solution for satellite-based Internet architectures are LEO systems with on-board processing and inter-link communication. However, this makes routing a real problem. In the article, the authors first discuss the drawbacks with those routing protocols used in terrestial networks, and then mention some routing protocols that have been specifically designed for satellite communication, e.g. DT-DVTR and VN. The article concludes with a discussion of the performance of TCP over satellite links.},
bibdate = {Tuesday, July 31, 2001 at 13:59:29 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Nguyen01,
author = {T. Nguyen and F. Yegenoglu and A. Sciuto and R. Subbarayan},
title = {Voice over IP Service and Performance in Satellite Networks},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {3},
pages = {164-171},
month = {March},
annote = {For VoIP over satellite links, several issues need to be addressed. These include transmission and QoS issues. The focus of this article is the performance of VoIP over satellite under various link and loading conditions. The result from their tests show that VoIP over satellite links is feasible. Over error-prone links VoIP operation is quite robust, tolerating BERs as high as 10^-5. Hence, loss rates should not be the limiting factor. However, VoIP is inherently not very bandwidth efficient. This due to small payloads and RTP/UDP/IP headers. The bandwidth utilization can be improved by using techniques such as voice compression and silence suppression.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, July 31, 2001 at 14:33:56 (CEST)}
}
@article{Gevros01,
author = {P. Gevros and J. Crowcroft and P. Kirstein and S. Bhatti},
title = {Congestion Control Mechanisms and the Best Effort Service Model},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {3},
pages = {16-25},
month = {May},
annote = {In this article the authors revisit the best effort service model and the problem of congestion. The notion of fairness is given a quite thorough treatment. Concepts like max-min fairness, weighted max-min fairness and proportional fairness are discussed. A dichotomy of current congestion control mechanisms is presented. Two broad classes of congestion control mechanisms are identified: host-based and router-based. These are further divided into open-loop and closed-loop control mechanisms. Other aspects of congestion control discussed are scheduling, queue and buffer management. The article concluded with a discussion of open issues and directions for further research.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, August 01, 2001 at 09:04:38 (CEST)}
}
@article{Widmer01,
author = {J. Widmer and R. Denda and M. Mauve},
title = {A Survey on TCP-Friendly Congestion Control},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {3},
pages = {28-37},
month = {May},
annote = {To a large extent, the stability and performance of the Internet depend on flows being responsive to congestion. The Internet community fears that the current revolution of multicast and real-time audio/video streaming applications could lead to congestion collapse and starvation of TCP-friendly flows. In this article, a survey of current approaches to TCP-friendliness is presented. Both unicast and multicast congestion control mechanisms are discussed and evaluated.},
bibdate = {Wednesday, August 01, 2001 at 09:14:59 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Barakat01,
author = {C. Barakat},
title = {TCP/IP Modeling and Validation},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {3},
pages = {38-47},
month = {May},
annote = {In this article, the different issues to be considered when modeling the TCP protocol in a real environment are discussed. The discussion is backed up with actual measurements made over the Internet. In summary, it is shown: - that a linear window increase does not hold for paths where the size of the congestion window is large compared to the bandwidth-delay product; - the simplistic assumption of periodic packet losses could lead to considerable error in some situations. In particular, underestimating the variance of time intervals between packet losses leads to underestimation of the throughput; - that each part of a TCP model needs to be validated separately. Otherwise one runs the risk of errors cancelling out each other. },
url = {papers/barakat01.ps.gz},
bibdate = {Wednesday, August 01, 2001 at 13:42:09 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Athuraliya01,
author = {S. Athuraliya and S. Low and V. Li and Q. Yin},
title = {REM: Active Queue Management},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {3},
pages = {48-53},
month = {May},
annote = {In this article, a new active queue management scheme, Random Exponential Marking (REM), is proposed. The principal idea with REM is to stabilize both the input rate around the link capacity and the queue size around a small target value. It accomplishes this by using a pricing scheme that works analogous with a PI- controller (u(t) = a * (y(t) - b(t)) + b * (y'(t) - b'(t))). A key feature of REM is that it takes into account the congestion over the complete path, not just an individual link, when calculating the marking probability for a specific flow.},
url = {papers/athuraliya01.pdf.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, August 01, 2001 at 15:14:31 (CEST)}
}
@article{Gibbens01,
author = {R. Gibbens and P. Key},
title = {Distributed Control and Resource Marking Using Best-Effort Routers},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {3},
pages = {54-59},
month = {May},
annote = {The problem considered in this article is how to determine an appropriate throughput for competing flows. In the article, the authors sketches on a solution that explores the idea of seeing the choice of throughput as a kind of distributed game. Their solution involves both control theory and theories from economics.},
bibdate = {Wednesday, August 01, 2001 at 15:54:38 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Hurley01,
author = {P. Hurley and J. Boudec and P. Thiran and M. Kara},
title = {ABE: Providing a Low-Delay Service within Best-Effort},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {3},
pages = {60-69},
month = {May},
annote = {In this article an alternative best effort service (ABE) is proposed. The idea with this service is to provide low delay at the expense of maybe less throughput. This makes ABE suitable for interactive adaptive applications. With ABE, every best effort packet is marked as either "green" or "blue". Green packets are guaranteed a low bounded delay in every router. In exchange, green packets are more likely to be dropped during periods of congestion than blue packets. For every packet, the choice of color is made by the application. ABE is realized by the introduction of a new scheduling method: duplicate scheduling with deadlines (DSD). Simulations with ns-2 show that, indeed, blue packet flows have higher throughput while green packet flows have a low bounded delay.},
bibdate = {Thursday, August 02, 2001 at 08:48:00 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Xylomenos01,
author = {G. Xylomenos and G. Polyzos and P. Mahonen and M. Saaranen},
title = {TCP Performance Issues over Wireless Links},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {4},
pages = {52-58},
month = {April},
annote = {Using TCP for wireless communication is coupled with several problems, e.g. high error rates due to atmospheric conditions like multipath fading and interruptions in communication during handoffs. In this article, the authors elaborate on the performance problems that arise when using TCP over wireless links. The characteristics of various wireless systems are discussed and how these characteristics could adversely interact with TCP mechanisms. Explanations of the causes of these problems are given, and various TCP performance enhancements, with focus on the transport and link layer, are presented. The article concludes with a discussion of future directions for wireless systems and what consequences this have on TCP.},
bibdate = {Monday, August 06, 2001 at 09:12:14 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Balakrishnan01,
author = {H. Balakrishnan and V. Padmanabhan},
title = {How Network Assymetry Affects TCP},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {4},
pages = {60-67},
month = {April},
annote = {Several access network technologies exhibit asymmetry in their network characteristics. For example, cable modem networks and ADSL. Network asymmetry can adversely impact the performance of feedback-based transport protocols such as TCP. The reason for this is that even if the network path in the direction of data flow is uncongested, congestion in the other direction can disrupt the feedback flow (ACKs in the case of TCP) and thereby lead to poor performance. In this article, the performance problems caused by network asymmetry in the context of TCP are discussed. Furthermore, solutions to these problems are presented. Experiments conducted by the authors show that these solutions indeed leed to improved TCP performance over asymmetric links.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, August 06, 2001 at 09:14:24 (CEST)}
}
@article{Chase01,
author = {J. Chase and A. Gallatin and K. Yocum},
title = {End System Optimizations for High-Speed TCP},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {4},
pages = {68-74},
month = {April},
annote = {Delivered TCP performance on high-speed networks is often limited by the sending and receiving hosts, rather than by the network resources and the TCP protocol implementation in itself. In this case, systems can achieve higher bandwidth by reducing host overheads. This article surveys some of these optimizations. More specifically it discusses the techniques: extended frames, interrupt coalescing, zero-copy, and checksum offloading. The suitability of these techniques are affirmed by experiments which show that bandwidth improvements of up to 70 percent can be achieved.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, August 06, 2001 at 09:16:48 (CEST)}
}
@article{Hong01,
author = {D. Hong and C. Albuquerque and C. Oliveira and T. Suda},
title = {Evaluating the Impact of Emerging Streaming Media Applications on {TCP/IP} Performance},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {4},
pages = {76-82},
month = {April},
annote = {Emerging streaming media applications in the Internet primarily use UDP and application-level control mechanisms that are not as responsive to congestion as TCP. As a result, streaming media applications can cause two major problems in the Internet: congestion collapse and unfair allocation of bandwidth. A common transmission technique employed in the Internet backbones and in WANs are ATM. In this article, the ABR service class is proposed to mitigate the effects of streaming media applications. Simulations and experiments show that ABR indeed is able to prohibit congestion and prevent unfair bandwidth allocation.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, August 06, 2001 at 09:19:15 (CEST)}
}
@article{Floyd01b,
author = {S. Floyd},
title = {A Report on Recent Developments in TCP Congestion Control},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {4},
pages = {84-90},
month = {April},
annote = {This article discusses several changes to TCP's congestion control, either proposed or in progress. The changes discussed include: limited transmit, D-SACK, and corruption notification. In the last part of the article, changes to the network infrastructure are discussed. More specifically, ECN and the role of active queue management.},
bibdate = {Monday, August 06, 2001 at 09:21:20 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Trimintzios01,
author = {P. Trimintzios et al.},
title = {A Management and Control Architecture for Providing IP Differentiated Services in MPLS-Based Networks},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {5},
pages = {80-88},
month = {May},
annote = {The Traffic Engineering for Quality of Service in the Internet at Large Scale (TEQUILA) project works towards ways of using DiffServ in order to obtain end-to-end QoS guarantees. In this article, the authors, which all are members of the TEQUILA project, propose a model and a framework for supporting DiffServ-based end-to-end QoS in the Internet, assuming underlying MPLS-based explicit routed paths. A template for service-level specifications (SLSs) are proposed followed by a functional architecture for supporting the QoS required by contracted SLSs, while trying to optimize the use of network resources.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, August 06, 2001 at 09:23:59 (CEST)}
}
@article{Park01,
author = {J. Park and J. Baek and J. Hong},
title = {Management of Service Level Agreements for Multimedia Internet Service Using a Utility Model},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {5},
pages = {100-106},
month = {May},
annote = {In this article a high-level conceptual SLA (Service-Level Agreement) management framework for multimedia Internet service using a utility model. The utility model is taken from microeconomics theory. An example is given that shows how the SLA framework could be used in controlling the QoS in a VoIP network.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, August 06, 2001 at 09:26:21 (CEST)}
}
@article{Nilsson01,
author = {O. Nilsson},
title = {Fundamental Limits and Possibilities for Future Telecommunications},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {5},
pages = {164-167},
month = {May},
annote = {Using fundamental physical and information theoretical relations, the author considers theoretical capacity limits and possibilities of fiber optical, cellular radio, and satellite communication systems. The overall conclusion is that it seems physically and in a longer perspective also economically feasible to build telecommunication networks offering subscribers a thousandfold increase in capacity.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, August 06, 2001 at 09:28:11 (CEST)}
}
@article{Grinnemo01b,
author = {K-J Grinnemo and A. Brunstrom},
title = {Enhancing {TCP} for Applications with Soft Real-Time Constraints},
journal = {Proceedings of SPIE (Multimedia Systems and Applications IV)},
year = {2001},
volume = {4518},
month = {August},
annote = {Describes PRTP-ECN as an extension to TCP. Presents the outcome of the theoretical analysis of the packet-loss behavior of PRTP-ECN, and summarizes the result from the simulation experiment described in proceedings of IWQoS 2001.},
url = {papers/grinnemo01b.pdf.gz},
bibdate = {Tuesday, August 28, 2001 at 11:48:39 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Bennani01,
author = {F. Bennani and A. Boutignon and N. Simoni},
title = {IP QoS issues: Efficient cooperation of DiffServ, MPLS and Management},
journal = {Proceedings of SPIE, Quality of Service over Next-Generation Data Networks},
year = {2001},
volume = {4524},
pages = {1-11},
month = {August},
annote = {The purpose of this paper is to study the relevance of an architecture where DiffServ, MPLS and network management would cooperate to offer a QoS-capable IP network. After having evaluated the following alternatives: 1) Over-provisioned IP network; 2) IP over ATM; 3) IntServ/RSVP; 4) DiffServ; 5) MPLS, the authors come to the conclusion that indeed DiffServ and MPLS are the most promising alternatives.},
bibdate = {Monday, September 03, 2001 at 08:26:30 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Noda01,
author = {Y. Noda and T. Sakai and H. Shigeno and Y. Matsushita},
title = {Performance Evaluation of Transport Layer Protocols for Transmitting Real-time Data over Diffserv Networks},
journal = {Proceedings of SPIE, Quality of Service over Next-Generation Data Networks},
year = {2001},
volume = {4524},
pages = {23-32},
month = {August},
annote = {In this paper, transport layer protocols to support soft real-time data transmission over networks using differentiated services are discussed. Assuming that an application transmits streaming video data over a DiffServ network, UDP and TCP have been evaluated. Currently, an increasing number of real-time media applications are implemented using UDP. The reason to this being that the flow and congestion control of TCP may impede timing constraints. However, in the simulations presented in this paper, TCP make use of bandwidth guaranteed by DiffServ and help to provide a more reliable delivery of real-time data than UDP without serious impacts on timing constraints. The authors therefore argue that TCP could actually be more suitable for real-time traffic in DiffServ networks than UDP.},
bibdate = {Monday, September 03, 2001 at 08:28:23 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Vutukury01,
author = {S. Vutukury and J. Garcia-Luna-Aceves},
title = {A Simple MPLS-based Flow Aggregation Scheme for Providing Scalable Quality of Service},
journal = {Proceedings of SPIE, Quality of Service over Next-Generation Data Network},
year = {2001},
volume = {4524},
pages = {91-98},
month = {August},
annote = {Today, there is a great demand to extend the best-effort service of IP networks to provide service classes that support QoS-sensitive real-time multimedia applications. To address this demand, IETF has proposed several architectural frameworks such as IntServ, DiffServ, and Traffic Engineering. A key underlying requirement in all these frameworks is a mechanism to pre-allocate bandwidth so that all packets in a flow follow the same path. In recent years, Multiprotocol Label Switching (MPLS) has received much attention as a way to signal paths in a network. A problem with MPLS is that it doesn't scale if a large number of paths need to be setup in the network. To reduce the label-state size, the label switching paths (LPSs) must be aggregated. However, to date, apart from multipoint-to-point LSP aggregation, there have been no proposals for more sophisticated LSP aggregation. In this paper, a new type of LSP aggregation is proposed, LSMP, (Label-Switched Multipaths). The LSMP aggregation is a generalization of multipoint-to-point LSP aggregation and is based on the rule that if two packets received by a router follow any of the paths in the same set of multiple paths starting from the router to the destination, then they must have the same label.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 03, 2001 at 08:30:07 (CEST)}
}
@article{Claypool01,
author = {M. Claypool and G. Kannan},
title = {Selective Flooding for Improved Quality-of-Service Routing},
journal = {Proceedings of SPIE, Quality of Service over Next-Generation Data Network},
year = {2001},
volume = {4524},
pages = {33-44},
month = {August},
annote = {A key component of QoS is QoS routing. Most proposed techniques to compute QoS routes require dynamic update of link-state information. Given the growing size of networks, it is becoming increasingly difficult to gather up-to-date state information. In this paper, a technique to compute QoS routes in a fast and efficient manner without any need for dynamic updates. The basic idea with their technique called Selective Routing is to evaluate multiple routes from a source to a destination in parallel. Each source on the network is assumed to have a static image of the entire network. This image needs to consist only of the set of nodes and edges which make up the whole topology, and does not require any link-state information. When a QoS request is received from an application, control packets are flooded across all possible routes to the destination as listed in the precomputed routing table. Along each route traversed, at each router, the control packet collects QoS information such as cumulative delay, available bandwidth etc. on the outgoing link. When all control packets arrive at the destination, the best path that satisfies the initial QoS constraints is computed. Selective Flooding was evaluated against source routing, the most common form of QoS routing. The outcome of the evaluation suggest that Selective Flooding outperforms source routing both in terms of call blocking rate, call setup time. However, it requires more storage space.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 03, 2001 at 08:32:22 (CEST)}
}
@article{Mogul01,
author = {J. Mogul and G. Minshall},
title = {Rethinking the TCP Nagle Algorithm},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {1},
pages = {6-20},
month = {January},
annote = {Modern TCP implementations include a mechanism, known as the Nagle algorithm, which prevents the unnecessary transmission of a large number of small packets. This algorithm has proved useful in protecting the Internet against excessive packet loads. However, many applications suffer performance problems as a result of the traditional implementation of the Nagle algorithm and TCP's delayed acknowledgement policy can create an especially servere problem, through a temporary "deadlock". In this paper, the interaction between the Nagle algorithm and the delayed acknowledgement policy is explored in detail. Several solutions are investigated, including some which have not been proposed before. The result of their investigation implies that a combination of the so-called DLDET and Minshall variants might work well in almost every case. However, there are a few hard cases that require more study, especially multithreaded applications using relatively small application buffers.},
bibdate = {Monday, September 10, 2001 at 09:22:28 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Guerin01,
author = {R. Guérin and V. Pla},
title = {Aggregation and Conformance in Differentiated Service Networks: A Case Study},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {1},
pages = {21-32},
month = {January},
annote = {The DiffServ framework relies on a small number of service models, or Per Hop Behaviors (PHBs) that each specifies how a router should treat the corresponding packets. The aggregate model of DiffServ is highly scalable, but it also raises questions in terms of the type of service it can provide. This paper elaborates on the potential penalty imposed by aggregating traffic into a small number of classes with respect to conformance as they exit a DiffServ domain. One contribution of this paper is to quantify the expected importance of reshaping, and when it is not available, identify parameters and alternatives that can be used to mitigate the impact of the non-conformance induced by traffic aggregation. The findings of this paper confirm that reshaping is by far the most efficient way to eliminate egress non-conformance. The amount of reshaping buffers required is typically small, i.e. of the order of a few packets, and when multiple flows are mapped into the same class, the relative amount of buffering required for each stream decreases, i.e. effiency improves.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 10, 2001 at 09:27:54 (CEST)}
}
@article{Hnatyshin01,
author = {V. Hnatyshin and A. Sethi},
title = {Achieving fair and predictable service differentiation through traffic degradation policies},
journal = {Proceedings of SPIE, Quality of Service over Next-Generation Data Networks},
year = {2001},
volume = {4524},
pages = {170-181},
month = {August},
annote = {It is widely acknowledged that DiffServ provides proper service differentiation in well-provisioned networks under normal traffic conditions. However DiffServ may fail to provide proper service differentiation in the presence of extreme network conditions such as congestion or in under-provisioned networks, resulting in unfair service degradation and unpredictable traffic behavior. In this paper, a new approach for service differentiation is proposed based on the observation that during periods of congestion all traffic that passes through a congested node should experience different levels of degradation of their quality of service. In particular, a degradation policy model is proposed. If incoming traffic violates the service level agreement then it is punished according to the value of the so-called non-conforming traffic field of the service level agreement. For example, non-conforming traffic could be dropped, assigned a lower service level etc. The degradation policy model is evaluated using an OPNET simulation model. The simulations show that the degradation policy model indeed works as expected.},
bibdate = {Monday, September 10, 2001 at 09:30:08 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Cano01,
author = {M. Cano and F. Cerdan and J. Garcia-Haro and J. Malgosa-Sanahuja},
title = {Performance Evaluation of Profiler Mechanisms for Internet Assured Service},
journal = {Proceedings of SPIE, Quality of Service over Next-Generation Data Networks},
year = {2001},
volume = {4524},
pages = {182-193},
month = {August},
annote = {This paper presents the result of a comparative simulation study between different types of profilers to realize the Assured Service in DiffServ. The main contribution of this paper is a new profiler mechanism: the Counters Based Algorithm. It is shown that this algorithm achieves very good performance in terms if contracted rates even better than the common Leaky Bucket Algorithm.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 10, 2001 at 09:31:46 (CEST)}
}
@article{Mishra01,
author = {A. Mishra and J. Korah},
title = {QoS of streaming video traffic over UMTS networks},
journal = {Proceedings of SPIE, Quality of Service over Next-Generation Data Networks},
year = {2001},
volume = {4524},
pages = {238-242},
month = {August},
annote = {In this paper, a simulation model for streaming video over an UMTS network is presented. The simulation model consists of a video server generating MPEG-4 video data over an UMTS connection. Only parts of the UMTS core network is modeled. For example, the signaling required for connection establishment and the packet data protocol (PDP) are not modeled. The simulations have not yet generated any results, but I think that the paper is interesting anyhow since it is one of a few papers concerning simulation of an UMTS network.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 10, 2001 at 09:33:47 (CEST)}
}
@article{Manner01,
author = {J. Manner et al.},
title = {Exploitation of wireless link QoS mechanisms in IP QoS architectures},
journal = {Proceedings of SPIE, Quality of Service over Next-Generation Data Networks},
year = {2001},
volume = {4524},
pages = {273-283},
month = {August},
annote = {In this paper, the challenges introduced to IP by wireless networks are discussed. In general, without an appropriate support from the link layer, it is very difficult to provide different QoS levels to IP flows. A generic convergence layer approach for QoS provision between the IP and link layers is presented. The key requirements for such a convergence layer is discussed and how the impact of sudden changes in link-layer QoS levels can be alleviated with the help of appropriate support in the convergence layer.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, September 10, 2001 at 09:35:27 (CEST)}
}
@article{Bouch99,
author = {A. Bouch and M. A. Sasse},
title = {Network Quality of Service: What do Users Need?},
journal = {proc. of the 4th International distributed conference (IDC'99)},
year = {1999},
pages = {78-90},
month = {September},
annote = {Most proposals for QoS networks assume that user's assessments of the QoS they receive mirrors the objective quality delivered at the network level (measurable through characteristics such as packet loss and delay). This paper tries to demonstrate that this assumption may not be correct. The authors report on experiments in which users' QoS requirement for interactive audio were investigated. The results indicate that users' assessments of the value of QoS received is influenced by a number of different factors, where one of the most important was to get a predictable level of QoS. The paper also makes recommendations for the design of a resource allocation system. },
bibdate = {Sunday, September 23, 2001 at 16:00:11 (CEST)},
submitter = {Katarina Asplund}
}
@inproceedings{Bouch00,
author = {A. Bouch and A. Kuchinsky and N. Bhatti},
title = {Quality is in the Eye of the Beholder: Meeting Users' Requirements for Internet Quality of Service},
booktitle = {Proceedings CHI'2000},
year = {2000},
month = {April},
annote = {This paper presents results from a study into how users perceive latency on the World Wide Web. The users performed a task that involved purchasing a home computer system. The Web pages in the task had predetermined delays ranging from 2 to 73 seconds, and the users were asked to rate the latency received for each Web page access. One finding was that the threshold where QoS is judged as "Low" is around 11 seconds (consistent with previous Web usage studies). However, if Web pages was loaded incrementally users are far more tolerant of latency (treshold for low QoS = 56 seconds). A number of contextual factors also seemed to influence users' tolerance of latency, e.g. duration of interaction, the company whose products are advertised, and type of task.},
submitter = {Katarina Asplund},
bibdate = {Sunday, September 23, 2001 at 16:37:52 (CEST)}
}
@article{Bouch00a,
author = {A. Bouch and M. A. Sasse and H. DeMeer},
title = {Of Packets and People: A User-centered Approach to Quality of Service},
journal = {Proceedings IWQoS 2000},
year = {2000},
pages = {189-197},
month = {June},
annote = {Differentiation of QoS classes within the DiffServ framework is based on the definition of various per-hop behaviors. What is currently missing is a technique for specification and mapping of application and user QoS preferences onto service profiles. The paper presents results from a study that investigates the QoS dimensions that are salient to users for six different applications (e-mail, ftp, video, audio, WWW,and audio/video). The results show that users' requirements vary according to the subjective value ascribed by the user and depends on the purpose of the task.},
bibdate = {Sunday, September 23, 2001 at 17:08:13 (CEST)},
submitter = {Katarina Asplund}
}
@article{Wilson00,
author = {G. M. Wilson and M. A. Sasse },
title = {Do Users Always Know What's Good For Them? Utilising Physiological Responses to Assess Media Quality},
journal = {proc. of HCI'2000},
year = {2000},
pages = {327-339},
month = {September},
annote = {This paper reports on a study that investigates physiological responses to two levels of video quality (5 vs. 25 frames per second). Results show that a video quality of 5fps caused responses to indicate stress although only 16% of the users noticed the difference subjectively. Therefore, the authors propose a 3-tier approach to multimedia quality assessment that combines task performance, user satisfaction and user cost (in terms of stress). },
bibdate = {Sunday, September 23, 2001 at 17:25:11 (CEST)},
submitter = {Katarina Asplund}
}
@article{Bhatti98,
author = {S. N. Bhatti and G. Knight},
title = {Notes on a QoS information model for making adaptation decisions},
journal = {proc. of HIPPARCH'98},
year = {1998},
month = {June},
annote = {The paper presents a network model that allows applications to dynamically adapt to the currently available network QoS. Adaptation decisions are made in response to fluctuations in QoS seen by a flow, but are also governed by preferences specified by the user. },
submitter = {Katarina Asplund},
bibdate = {Sunday, September 23, 2001 at 17:54:49 (CEST)}
}
@article{Casilari01,
author = {Casilari, E.; Gonzblez, F.J.; Sandoval, F.},
title = {Modeling of HTTP traffic},
journal = {IEEE Communications Letters},
year = {2001},
volume = {5},
number = {6},
pages = {272-274},
month = {June},
annote = {Argues for the inclusion of several levels of analysis, namely the session, page, connection and packet level when looking at HTTP transactions. Presents a study of tcpdump traces of IE HTTP traffic examining the relation between the page level and the connection level, as manifested in the number of connection neededed per page versus the time between page requests. Three regions were found as could be expected. Does not mention pipelining persistent HTTP. },
url = {papers/Casilari01_HTTPModelling.pdf},
bibdate = {Monday, September 24, 2001 at 08:20:18 (CEST)},
submitter = {Johan Garcia}
}
@article{Akyildiz01,
author = { Akyildiz, I.F.; Morabito, G.; Palazzo, S.},
title = { Research issues for transport protocols in satellite IP networks},
journal = {IEEE Personal Communications},
year = {2001},
volume = {8},
number = {3},
pages = {44-48},
month = {June},
annote = {This article provides a nice overview of the problems facing satellite and discusses the areas Slow start, Link Errors and Bandwidth Assymetry. A number of propasals are briefly discussed with their merits and disadvantages. Also touches the problems of real-time applications.},
url = {papers/Akyildiz01_SatelliteTPproto.pdf},
submitter = {Johan Garcia},
bibdate = {Monday, September 24, 2001 at 08:44:11 (CEST)}
}
@article{Akyildiz01a,
author = {I. F. Akyildiz and G. Morabito and S. Palazzo},
title = {{TCP-Peach}: A New Congestion Control Scheme for Satellite IP Networks},
journal = {{IEEE/ACM} Transactions on Networking},
year = {2001},
volume = {9},
number = {3},
pages = {307--321},
month = {jun},
annote = {In this paper, a new congestion control scheme for networks with long propagation delays and high link error rates, e.g., satellite networks, is presented. The new congestion control scheme, TCP-Peach, comprises two new algorithms - Sudden Start and Rapid Recovery. Both algorithms are based on the use of so called dummy segments to probe the availability of bandwidth. The dummy segments are treated as low priority packets and thus should not have any substantial impact on the actual data traffic. Simulation experiments show that TCP-Peach outperforms other TCP schemes for satellite networks in terms of goodput, and that TCP-Peach connections share bandwidth fairly in between each other. However, when TCP Reno and TCP-Peach both run over a satellite link, the bandwidth is not shared evently between the two. For example, with a packet loss of 1E-4 over a geostationary satellite channel, TCP-Peach obtains 80% of the bandwidth.},
url = {papers/Akyildiz01.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{McCreary00,
author = {S. McCreary and K. Claffy},
title = {Trends in Wide Area {IP} Traffic Patterns - A View from Ames {Internet} Exchange},
journal = {Proceedings of the 13th ITC Specialist Seminar on Internet Traffic Measurement and Modelling, Monterey, CA},
year = {2000},
month = {September},
annote = {Abstract: We report results from a longitudinal analysis of the IP traffic workload seen at a single measurement site inside a major Internet traffic exchange point. Using data collected by the NLANR/MOAT Network Analysis Infrastructure (NAI) project [NAI] and analysis software from CAIDA's CoralReef project [CoralReef], we present trends in application usage seen at the NASA Ames Internet Exchange over 10 months, from May 1999 through March 2000. We show changes in the fraction of traffic from streaming media and online gaming, as wellas an increase in traffic from new applications such as Napster and IPSEC tunneling. We also show that our data does not indicate any overall change in the TCP/UDP traffic ratio at the Ames Internet Exchange during this period, or significant differences from the analyses by MCI Worldcom and CAIDA in 1998. },
bibdate = {Wednesday, September 26, 2001 at 11:07:56 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{WMH01,
author = {{Microsoft Corporation}},
title = {{Windows Media Homepage}},
journal = {http://www.microsoft.com/windows/mediaplayer/default.asp},
year = {2001},
month = {September},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, September 26, 2001 at 11:16:20 (NFT)}
}
@inproceedings{Pang00,
author = {Q. Pang and A. Bigloo and V.C.M. Leung and C. Scholefield},
title = {Performance Evaluation of Retransmission Mechanisms in {GPRS} Networks},
booktitle = {Proceedings of IEEE Wireless Communications and Networking Conference (WCNC'00), Chicago, IL},
year = {2000},
month = {September},
annote = {In GPRS retransmissions can be performed at three levels: TCP, LLC and RLC/MAC. OPNET simulations were used to evaluate combinations of retransmission mechanisms: TCP, TCP+RLC, and TCP+LLC+RLC. As previous studies, this study indicates that HTTP/FTP goodput is improved with RLC/MAC retransmissions. LLC retransmissions do not improve performance further. The performance may even degrade because of inefficient TCP and LLC interaction.},
url = {papers/pang00_GPRS_retrans.pdf.gz},
bibdate = {Monday, October 15, 2001 at 09:59:33 (CEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Padmanabhan01,
author = {Venkata N. Padmanabhan and Lakshminarayanan Subramanian},
title = {An Investigation of Geographic Mapping Techniques for Internet Hosts},
journal = {Proceedings of SIGCOMM'2001},
year = {2001},
pages = {13},
month = {August},
annote = {The paper questions if it is possible to build an IP address to geographic location mapping service for Internet hosts. This would be useful for location-based targeted information on local events, regional weather, targeted advertising or terrestrial rights management [like DVD region coding --SA interpretation]. To facilitate this, three techniques were investigated, collectively called IP2Geo; It consists of GeoTrack which tries to use DNS information for location. GeoPing measures the delay from different sites to the target in combination with reference sites to determine the location. GeoCluster uses BGP prefix information to infer the location of the target host. They note a fundamental difficulty with users using proxies (the proxy location is discovered, not the users location), especially with America Online in which all proxies are clustered at the same site. To verify the results, they used IP adresses and user information collected from HotMail, bCentral and "FooTV" (WebTV?), where the users had voluntarily given their state or zip codes. The results showed that hosts could be located with a median error distribution of between 28km to a few hundred kilometers depending on the type of host (well connected university host, or more heterogenous sets) },
url = {papers/Pabmanabhan01_Geographic_Mapping_of_Internet_Hosts.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Tuesday, October 16, 2001 at 10:29:19 (CEST)}
}
@article{Stuckmann99,
author = {P. Stuckmann and P. Seidenberg},
title = {Quality of Service of Internet Applications over {GPRS}},
journal = {Proceedings of European Wireless (EW'99)},
year = {1999},
month = {October},
annote = {A simulation tool, GPRSim, developed at Aachen University of Technology was used to examine Internet applications over GPRS. QoS from the user's point of view was represented by throughput. Channel utilization was used to measure system performance. The Internet traffic models were based on analytical models for the Internet and ETSI models for UMTS. Mean throughput was measured for three C/I levels (ideal, 16 dB, 9 dB) for 1-10 active stations sharing one PDCH. Throughput degraded severely when more than two stations shared the channel. High throughput gives low channel utilization and low throughput gives high channels utilization. A mean throughput of 700 byte/s was achieved for 10 stations sharing 4 PDCHs (C/I ideal). Throughput was examined also for two service classes. For the privileged class throughput was of course higher. User behavior has an impact on QoS. The number of stations that can be served depends on the time users spend on reading HTTP pages. The longer the read times, the more station can be served. },
url = {papers/Stu_ECWT99.pdf.gz},
bibdate = {Tuesday, October 16, 2001 at 10:44:52 (CEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Stuckmann00,
author = {P. Stuckmann and F. M\"{u}ller},
title = {{GPRS} Radio Network Capacity and Quality of Service using Fixed and On-Demand Channel Allocation Techniques},
journal = {Proceedings of Vehicular Technology Confernece (VTC spring 2000)},
year = {2000},
month = {May},
annote = {Fixed and on-demand channel allocation in GPRS was evaluated using GPRSim, a simulation tool developed at Aachen University of Technology. The same traffic models were used as in [Stuckmann99]. Fixed channel allocation gives higher throughput than on-demand allocation. Channel allocation time depends on read time of HTTP pages. The conclusion from the simulations are that PDCHs are released too early and that the Master PDCH should be a dedicated channel and hence not allocated on-demand.},
url = {papers/Stu_VTC00.pdf.gz},
bibdate = {Tuesday, October 16, 2001 at 11:32:41 (CEST)},
submitter = {Annika Wennstr\"{o}m}
}
@phdthesis{Iren99a,
author = {Sami Iren},
title = {Network-conscious Image Compression},
school = {University of Delaware},
year = {1999},
annote = {This thesis presents the notion of network-conscious image compression. This means that the coding should support Progressive display, use ALF, have relaxed order and reliability requirements and provide graceful degradation.A multiresolution representation is also desireable. In short network-conscious image compression focuses on optimizing overall progressive display performance over communications networks and not only compression performance. The thesis provides backgrounds on compression and the transport layer. The GIF-NC network conscious GIF variant is described and experimental results reported. The experiments aimed to evaluate the difference in progressive display. Bandwidths between 2.4 and 38.4 kbps were used. GIF-NC provided better progressive display but took longer time for the complete image. The test images sent were however better suited for JPEG or wavelet based compression. A smaller section at the end presents a wavelet network conscious compression scheme. },
url = {papers/Iren99_PhDdiss_netimage.ps.gz},
bibdate = {Tuesday, October 16, 2001 at 12:08:56 (NFT)},
submitter = {Johan Garcia}
}
@inproceedings{Stuckmann00a,
author = {P. Stuckmann and F. M\"{u}ller},
title = {{GPRS} Radio Network Capacity Considering Coexisting Circuit Switched Traffic Sources},
booktitle = {Proceedings of European Conference on Wireless Technologies 2000},
address = {Paris, France},
year = {2000},
month = {October},
annote = {Internet applications in co-existence with circuit switched applications (e.g. speech) is investigated using GPRSim (see [Stuckmann99]). Throughput and delay for IP packets were measured. The Internet traffic generated consisted of 1/3 HTTP and 2/3 SMTP. A Poisson process was used to model the arrival of circuit switched traffic/calls. C/I was 12 dB which corresponds to a BER of 13.5\%. The Grade of Service was guaranteed to be at least 1\%. The results indicate that the gain from using fixed allocation of PDCHs is small. More collisions with circuit switched traffic is expected if there are more GPRS traffic. QoS would be increased if TBFs could be redistributed on available channels and PDCHs could be reused.},
url = {papers/Stu_ECWT00.pdf.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Tuesday, October 16, 2001 at 12:07:04 (CEST)}
}
@phdthesis{Bolliger00,
author = {Jürg Bolliger},
title = {A framework for network-aware applications},
school = {ETH Zürich},
year = {2000},
annote = {This theseis provides a framework for adaptation that takes both network and end-system resources characteristics into account. Three aspect of application-specific functionality is modeled: 1 The object types handled by the application, 2 The algorithms applicable to transcoding and 3 the rource models for the costs incurred versus the size reductions for these algorithms. and also discusses several realted points. A number of experiments using JPEG adaptation in a framework is presented, mainly regarding communication-computation tradeoff. Costs and utility of adaptation is also discussed. The second part of the thesis discusses how to find out dynamic bandwidth changes, and how to communicate these to the application. The suggested solution is small changes to the transport level to collect the information and a small widening of the API, which is not discussed to communicate this to the application.},
url = {papers/Bolliger00_Framweork_netw_aware_appPhDthesis.ps.gz},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 16, 2001 at 12:18:56 (NFT)}
}
@article{Mahanti01,
author = {Anirban Mahanti and Derek Eager and Mary Vernon and David Sundaram-Stukel},
title = {Scalable On-Demand Media Streaming with Packet Loss Recovery},
journal = {Proceedings of SIGCOMM'2001},
year = {2001},
pages = {12},
month = {August},
annote = {The paper develops scalable protocols for reliable on-demand delivery of streaming media. Introduces Periodic Broadcast Protocols and Bandwidth Skimming Protocols, and states that they are not suitable for streaming media. Their protocol Reliable Periodic Broadcast (RPB) improves in this area by allowing the client to quickly start listen to a stream when a segment is received (using overlapping segments on different channels). Reliable Bandwidth Skimming (RBS) allows the client to adjust media bitrate to clients available bandwidth.},
url = {papers/Mahanti01_Scalable_OnDemand_Media_Streaming.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Tuesday, October 16, 2001 at 12:55:49 (CEST)}
}
@article{Baboescu01,
author = {Florin Baboescu and George Varghese},
title = {Scaleable Packet Classification},
journal = {Proceedings of SIGCOMM'2001},
year = {2001},
pages = {12},
month = {August},
annote = {Argues that more and more routers will do packet classification in order to provide differentiated output queueing, taking security-related actions, load balancing and doing traffic measurement. This is done by defining a flow based on packet header fields, for example source+dest address/ports. Also mentions that classifiaction rules will grow very large at edge routers, if/when DiffServ is realized. Most practical classification searches through rules sequentially which uses linear time (i.e. not good when rule lists grow large). The paper uses the Lucent bit vector (BV) algorithm as a point of departure since it scales to medium size databases, and can be implemented in both HW and SW. The BV is made more scaleable through aggregation and reordering of matching rules (Aggregate Bit Vector, ABV). It targets the high cost in memory lookups in the bit vector, by assuming the vector is sparse. Experiments were made, showing ABV to be at least an order of magnitude faster that the BV scheme on all tests performed. They believe their algorithm should have sufficient speed for OC-48 links even for large databases using SRAM.},
url = {papers/Baboescu01_Scalable_Packet_Classification.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Tuesday, October 16, 2001 at 12:55:55 (CEST)}
}
@phdthesis{Raman00,
author = {Suchitra Raman},
title = {A framework for Interactive Muticast Data Transport in the Internet},
school = {University of Califonia at Berkeley},
year = {2000},
annote = {This theseis provides several multicast-related contributions such as a model for soft-state transport mathematical model. A scalable naming and announcement protocol aims to allow a receiver to request application-derived data instead of just ranges of transport level sequence numbers. Asymptotic timer analysis based on randomized receiver framework is also multicast related. The thesis also presents ITP, Image Transport Protocol that is a partially reliable protocol for JPEG image transfer built on UDP. },
url = {papers/Raman00_ITP_ochImagetranfPhD.ps.gz},
bibdate = {Tuesday, October 16, 2001 at 12:18:56 (NFT)},
submitter = {Johan Garcia}
}
@article{Cohen01,
author = {Edit Cohen and Heim Kaplan},
title = {Aging Through Cascaded Caches: Performance Issues in the Distribution of Web Content},
journal = {Proceedings of SIGCOMM'2001},
year = {2001},
pages = {13},
month = {August},
annote = {Discusses the interactions between multiple levels of web caching, and defines it as a performance factor. Examines the effect of refreshing non-expired objects and extending the object lifetimes.},
url = {papers/Cohen01_Performance_Issues_in_Cascaded_Caches.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Tuesday, October 16, 2001 at 17:48:22 (CEST)}
}
@article{Rasheed00,
author = {Y. Rasheed and A. Leon-Garcia},
title = {Towards a generic real-time transport and adaptation protocol for IP networks},
journal = {Computer Communications},
year = {2000},
volume = {23},
pages = {1448-1458},
month = {August},
annote = {This paper presents the protocol GRAP (Generic Real-time transport and Adaptation Protocol). GRAP provides real-time applications with a set of flexible and programmable network adaptation capabilities that can be tailored to the specific application requirements. A novel feature of GRAP is that part of the packet format is configurable, using GRAP profiles. For example, delay-sensitive applications may require tight timing recovery functionality with strict delay bounds. This would necessitate use of long time stamps and relatively less Forward Error Correction (FEC) capabilities. On the other hand, loss-sensitive applications may require stronger FEC capabilities at the expense of tolerating relatively higher delay bounds. However, a GRAP profile cannot be changed dynamically, but are fixed for the duration of the connection. In addition, the only available error control scheme is FEC (optionally combined with interleaving).},
bibdate = {Thursday, October 18, 2001 at 11:55:19 (CEST)},
submitter = {Katarina Asplund}
}
@misc{RFC2914,
title = {{RFC 2914}: Congestion Control Principles},
author = {Sally Floyd},
month = september,
year = {2000},
note = {Category: Best Current Practise},
annote = {The goal of this document is to explain the need for congestion control in the Internet, and to discuss what constitutes correct congestion control. One specific goal is to illustrate the dangers of neglecting to apply proper congestion control. A second goal is to discuss the role of the IETF in standardizing new congestion control protocols.},
url = {ftp://ftp.isi.edu/in-notes/rfc2914.txt,},
bibdate = {Sun Oct 21 16:27:50 CEST 2001},
submitter = {Anna Brunstr\"{o}m},
online = {yes}
}
@article{Brakmo95,
author = {Lawrence S. Brakmo and Larry L. Peterson},
title = {{TCP Vegas}: End to End Congestion Avoidance on a Global {Internet}},
journal = {IEEE Journal on Selected Areas in Communications},
year = 1995,
volume = 13,
number = 8,
pages = {1465--1480},
month = oct ,
annote = {Vegas is an implementation of TCP that achieves between 37 and 71% better throughput on the Internet, with one-fifth to one-half the losses, as compared to the implementation of TCP in the Reno distribution of BSD Unix. This paper motivates and describes the three key techniques employed by Vegas, and presents the results of a comprehensive experimental performance study--using both simulations and measurements on the Internet--of the Vegas and Reno implementations of TCP.},
url = {papers/Brakmo96_TCPVegas.ps.gz},
references = 18,
bibdate = {Sun Oct 21 16:27:50 CEST 2001},
submitter = {Anna Brunstr\"{o}m},
keywords = {TCP; Vegas; congestion control}
}
@inproceedings{Hengartner00,
author = {Hengartner, Urs and Bolliger, Juerg and Gross, Thomas},
title = {{TCP} Vegas Revisited},
booktitle = infocom,
year = 2000 ,
address = {Tel Aviv, Israel},
month = mar,
annote = {The innovative techniques of TCP Vegas have been the subject of much debate in recent years. Several studies have reported that TCP Vegas provides better performance than TCP Reno. However, the question which of the new techniques are responsible for the impressive performance gains remains unanswered so far. This paper presents a detailed performance evaluation of TCP Vegas. By decomposing TCP Vegas into the various novel mechanisms proposed and assessing the effect of each of these mechanisms on performance, we show that the reported performance gains are achieved primarily by TCP Vegas's new techniques for slow-start and congestion recovery. TCP Vegas's innovative congestion avoidance mechanism is shown to have only a minor influence on throughput. Furthermore, we find that the congestion avoidance mechanism exhibits fairness problems even if all competing connections operate with the same round trip time.},
url = {papers/hengartner00_TCP_Vegas_Revisited.ps.gz},
keywords = {Communication protocols implementation and software; Protocol design, analysis and performance; Congestion, admission and flow control},
days = {28--30},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Sun Oct 21 16:27:50 CEST 2001}
}
@article{Blumenthal01,
author = {M. Blumenthal and D. Clark},
title = {Rethinking the design of the Internet: The end to end arguments vs. the brave new world},
journal = {ACM Transactions on Internet Technology},
year = {2001},
volume = {1},
number = {1},
pages = {70-109},
month = {August},
annote = {This paper looks at the Internet and the changing set of requirements for the Internet that are emerging as it becomes more commercial, more oriented towards the consumer, and used for a wider set of purposes. In a sense this paper is a sequel to the seminal paper "End-to-End Arguments in System Design". The question that the authors try to answer is whether or not the end to end arguments in the design of the Internet will survive the range of new requirements emerging. In short, the end to end arguments of Internet could be formulated in the following way: A service in a network can only be completely and correctly implemented with the knowledge and help of the appplication standing at the endpoints of the network. Therefore, providing that service as a feature of the network itself is not possible. As a consequence of the end to end arguments, the Internet has evolved to have certain characteristics. The functions implemented "in" the Internet, by the routers that forward packets, have remained rather simple and general. In recent years, the stakeholders of the Internet: ISPs, governments and others, have begun to questioned the ability of Internet to live up to the new requirements imposed on it. In particular, the stakeholders expresses serious doubts as to Internets ability to: 1) operate in an untrustworthy world. The end to end argument assume that endpoints are in willing cooperation to achieve their goals. Today, there is less and less reason to believe that we can trust other endpoints to behave as desired. Making the network more trustworthy, while the endpoints cannot be trusted, seems to imply more mechanisms in the core of the network. 2) accommodate more demanding applications. The simple service model of the Internet provides for example no guarantee about the throughput a particular application will have at any moment. Today, a new set of applications is emerging, typified by streaming audio and video, that appear to demand a more sophisticated Internet service that can assure each data stream a specified throughput, an assurance which seems to imply deviations from the end to end argument. 3) provide ISP service differentiation. The ISPs seem to view enhanced data transport service as something to be provided within the bounds of the ISPs which surely goes against the ideas behind the end to end argument. The authors of this paper conclude that even though there is a risk that the range of new requirements emerging could have the consequence of compromising the fundamental design principles of the Internet, they argue that the open, general nature of the Internet, which derived from the end to end arguments, is a valuable characteristic that encourages innovation, and therefore should be preserved.},
url = {papers/blumenthal01.pdf.gz},
bibdate = {Monday, October 22, 2001 at 14:15:18 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Vojnovic01,
author = {M. Vojnovic and J. Boudec},
title = {Some Observations on Equation-Based Rate Control},
journal = {Proceedings of the Seventeenth International Teletraffic Congress (ITC-17)},
year = {2001},
month = {September},
annote = {This paper investigates under what circumstances an equation-based rate control mechanism is conservative. A rate control mechanism is conservative if given a target loss-throughput function,f, the throughput x is less than or equal to f. In the internet f() is obtained as a function of the loss-event function and the round-trip time. However, in this paper f() is seen as only depending on the loss-event rate, i.e. the round-trip time is assumed to be constant. The major finding of this paper are the formulation of two conditions that must be satisfied in order for an equation-based rate control mechanism to be conservative. The first condition states that f(p) must be concave with 1/p and the second states that the expected time between loss-events must not increase with increased throughput, x. To verify these conditions, the authors have conducted a series of simulations, whose result is also presented in the paper.},
url = {papers/vojnovic01.pdf.gz},
bibdate = {Monday, October 22, 2001 at 14:22:50 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Alouf01,
author = {S. Alouf and P. Nain and D. Towsley},
title = {Inferring Network Characteristics via Moment-Based Estimators},
journal = {Proceedings of the IEEE INFOCOM 2001 Conference},
year = {2001},
month = {April},
annote = {In this paper, two inference models are developed based on Markov processes. The two models differ in that the first model assumes exponentially distributed packet sizes while the second model prescribes a constant packet size. Five metrics describing the QoS provided to an application are considered: 1) the packet-loss probability; 2) the expected response time; 3) the conditional loss probability, i.e. the probability that two consecutive packet losses occur; 4) the conditional non-loss probability, i.e. the probability that a packet is not dropped given that the previous packet was not dropped; 5) the server occupation, i.e. the probability that a packet arrives at a non-empty queue. Estimators are derived for these QoS metrics and it is shown that based upon these estimators it is possible to calculate estimates for cross traffic (r), i.e. rate of contending flows, router queue size (s), and router servicing time (t). Validation of the estimates r, s, and t are performed through simulations with ns2.},
url = {papers/alouf01.pdf.gz},
bibdate = {Monday, October 22, 2001 at 14:28:43 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Biaz97b,
author = {S. Biaz and N. Vaidya},
title = {Using End-to-End Statistics to Distinguish Congestion and Corruption Losses: A Negative Result},
journal = {Technical Report 97-009, Dept. of Computer Science Texas A\&M University},
year = {1997},
month = {August},
annote = {In a wireless environment the number of packet losses having other causes than congestion can be significant. Since the sending rate of TCP only should be reduced in case of congestion, there are much performance to be gained if we were able to differentiate between packet losses due to congestion and packet losses having other causes, e.g. fading. Several schemes have been proposed to detect congestion by using some statistics on round-trip delays and/or throughput. If these schemes were accurate then one possible algorithm to decide whether or not to reduce the congestion window at times of packet losses would be to see what the congestion detection scheme said just before the packet losses occurred. In this paper, the accuracy of three of the more common congestion detection schemes proposed are evaluated: Jain's criterion, Wang and Crowcroft's Normalized Throughput Gradient (NTG) criterion, and the Vegas criterion. The three criteria were evaluated using FreeBSD. Two parameters were measured for each criteria: 1) Accuracy of Prediction (AP) computed by dividing the number of times the congestion detection scheme said that the congestion window should be decreased just before a loss occurred by the total number of losses; 2) Frequency of Congestion Prediction (FCP) computed by dividing the number of times the congestion detection scheme said that the congestion window should be decreased by the total number of predictions. A congestion detection scheme that is good is supposed to have significantly greater AP than FCP. However, the result of the evaluation was very discouraging. For the three congestion detection schemes evaluated, AP was only marginally better than FCP. Hence, these detection schemes did not perform much better than a random predictor. The authors argue that the reason the three congestion detection schemes did not perform better was mainly due to the fact that a single flow has very little impact on the overall load variations and the variations in round-trip times. Since there is no synchronization between competing flows, congestion losses will occur almost randomly, which invalidates the assumptions underlying the three schemes.},
url = {papers/biaz97b.pdf.gz},
bibdate = {Monday, October 22, 2001 at 14:33:52 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Yano00,
author = {K. Yano and S. McCanne},
title = {A Window-based Congestion Control for Reliable Multicast based on TCP Dynamics},
journal = {Proceedings of ACM Multimedia 2000},
year = {2000},
pages = {249-258},
month = {October},
annote = {In this paper a novel multicast congestion control scheme for reliable bulk data transfer is proposed. The scheme, named Rainbow, is intended to work on top of a specific multicast forwarding service, BCFS (BreadCrumb Forwarding Service). The primary objective with Rainbow was to develop a window-based congestion control scheme for multicast that is scalable, shows high link utilization, and is TCP-friendly. To this end, Rainbow is designed as a receiver-based congestion control scheme. Each receiver, independently of each other, executes a TCP-like congestion control scheme. A transmission is initiated by the receiver sending a transmission request (TRQ) as a BCFS request to the sender. The size of the receiver's congestion window progresses in the same way as TCP's congestion window, i.e. with one packet per ack in slow start and one packet per round-trip in congestion avoidance. However instead of ack's Rainbow uses the so-called breadcrumbs in BCFS. Furthermore, Rainbow uses BCFS labels instead of sequence number in the TRQs. The efficiency of Raindow is demonstrated through a simulation experiment in which Rainbow is evaluated against RLC (Reliable Layered multicast Congestion control). The simulation experiment suggested that Raindow is more scalable than RLC in terms of overhead. Furthermore, the simulation experiment suggested that Rainbow is equally TCP-friendly as TCP and shows the same delay dependency as TCP does. This was not the case for RLC which did not compete fairly with TCP. However, RLC exhibited intra-session fairness, something neither Rainbow nor TCP did. },
url = {papers/yano00.pdf.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, October 22, 2001 at 14:41:05 (CEST)}
}
@article{Alfredsson01,
author = {S. Alfredsson},
title = {{TCP Lite} - {A} Bit Error Transparent Modification of {TCP}},
journal = {Master's Thesis 2001:06, Karlstad University},
year = {2001},
month = {June},
annote = {Abstract: TCP is a reliable transport protocol designed for heterogenous networks. To provide a reliable service over possibly unreliable networks, retransmission of lost or damaged data is performed. These retransmissions incur a delay and increases the total transmission time. However, certain applications can make use of damaged data, while taking advantage of the decreased delay created by fewer retransmissions. Currently there is no way to allow the applications to access this data. This thesis proposes a modification to TCP which would allow applications to decide when damaged data can be accepted and not. The idea has been implemented in the Linux operating system. As errors often occur over wireless links, the implementation has been tested with a number of emulated wireless links. The experiments showed that there are gains to be made by letting errors through.},
url = {papers/alfredsson01_mthesis_tcplite.pdf.gz},
bibdate = {Monday, October 29, 2001 at 18:18:46 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Insolvibile01,
author = {Gianluca Insolvibile},
title = {Linux Socket Filter: Sniffing Bytes over the Network },
journal = {Linux Journal },
year = {2001},
pages = {24-31},
month = {June},
annote = {Describes the packet socket available in Linux, starting from kernel 2.2, and how to sniff packets from the network using it. Gives a nice overview of the mechanisms and also provides detailed code examples on how to use it. One interesting detail given is that you can obtain the Berkeley packet filter code from tcpdump using the -d switch.},
bibdate = {Sunday, November 04, 2001 at 18:07:07 (CET)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Cole01,
author = {R.G. Cole and J.H. Rosenbluth },
title = {Voice over {IP} Performance Monitoring},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {2},
pages = {9-24},
month = {April},
annote = {The paper describes a method for monitoring VoIP applications based on ITU-T's E-Model. The E-Model is an analytic model of voice quality. The R factor (measure of voice quality) ranges between 0-100 and is given by: 100 - signal_to_noise impairements of SCN path - mouth_to_ear delay of impairment - equipment impairment (loss etc.) + expectation factor. In the paper the R factor is reduced to depend on measures available at the transport level (delay, network packet loss and loss in the de-jitter buffer) for some codecs. The idea is interesting, but many many assumptions and simplifications must be made before arriving at the final method. A monitoring tool that uses the method has been implemented.},
bibdate = {Sunday, November 04, 2001 at 18:40:37 (CET)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Krasic01,
author = {Charles Krasic and Kang Li and Jonathan Walpole},
title = {The Case for Streaming Multimedia with {TCP}},
journal = {International Workshop on Interactive Distributed Multimedia Systems (IDMS)},
address = {Lancaster, UK},
year = {2001},
month = sep,
annote = {The paper revisit and challenge the dogma that TCP is an undesirable choice for streaming multimedia, video in particular. It is argued that providing smooth video quality requires relatively large receiver buffers. The buffers can then also accomodate TCP retransmissions and the variability in rate that results from congestion control. Employing a new transport protocol is hard and the gains for streaming media may be marginal.},
url = {Krasic_streaming_TCP.pdf},
bibdate = {Monday, November 05, 2001 at 08:51:09 (CET)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Ballintijn00,
author = {Gerco Ballintijn and Maarten van Steen and Andrew S. Tanenbaum.},
title = {Characterizing Internet Performance to Support Wide-area Application Development. },
journal = {Operating Systems Review},
year = {2000},
volume = {34},
number = {4},
pages = {41-47},
month = {October},
annote = {An informal study of the relation between wide-area latency, the number of routers, and geographical distance between Internet sites is presented. The study was done using ping and traceroute measurements between 19 sites distributed across the globe. Contrary to the authors expectation the study indicate that there is almost no correlation between distance, latency, and number of routers in the current Internet. Only educational sites were used in the study and the mesurements were taken only on one occation for each site. },
url = {Ballintijn_Internet_perf.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Monday, November 05, 2001 at 09:00:17 (CET)}
}
@article{Dutta01,
author = {A. Dutta-Roy},
title = {An Overview of Cable Modem Technology and Market Perspectives},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {6},
month = {June},
annote = {Describes the benefits of using CATV as broadband carrier; no need to tie up the phone line, instantly online, 95 percent of american households are passed by cable. The CATV architecture is built of a 'head-end' that receives transmissions via satellite. In a hybrid fiber coax net, the signal is sent to "fiber nodes" via fiber, and converted from optical to electric signals. Each fiber node can support 500-2000 households. The signal is enhanced in the coax net via one-way amplifiers. The asymmetry of upstream/downstream data on the internet is discussed, affecting the design of the two channels accordingly. One band of 6Mhz is allocated for upstream traffic, which can support 400 users. This is because not all of the registered subscribers access the net all the time. Describes the problem with one-way amplifiers, the upstream signal must be demultiplexed and remultiplexed at the amps. Discusses the DOCSIS standard for data over cable service, and that the upcoming standard will support QoS and better security. Problems with voice data packets are discussed. Assuming speech at 64kb/s, sampled every 5ms, sent on a 2Mbit/s link. If 1500 ethernet frames are sent at the same time, one frame will take 6ms to transmit, meaning the user would have to choose to speak or surf, but not both. Solved with "fragment and reassembly" in the cable modem. Further the problem of transmission with repeaters is discussed; The sender can only detect collisions within its segment. Sovled by a polling / TDMA hybrid, controlled by the head-end.},
bibdate = {Monday, November 05, 2001 at 09:11:36 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Donaldson01,
author = {G. Donaldson and D. Jones},
title = {Cable Television Broadband Network Architectures},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {6},
month = {June},
annote = {The cable networks have gone from being purely coax to hybrid fiber-coax, with the trend of more and more "fiber-deep" networks. Discusses Power Domain Node, PDN, which solves a problem with power overprovisioning -- instead of setting a fixed number of users in a node, they are dependent on what one, two, ..., powersupplies can handle. To get increase the fiber depth, "mini fiber nodes" are used on the last coax mile, which eliminates the need for all coax amplifiers. Instead of having 500 users in one node, this is narrowed to 50 users, improving bandwidth and reliability.},
bibdate = {Monday, November 05, 2001 at 09:12:46 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Becchetti01,
author = {L. Becchetti, F. D. Priscoli, T. Inzerilli, P. M\"{a}h\"{o}nen, L. Munoz},
title = {Enhancing IP Service Provision over Heterogenous Wireless Networks: A Path toward 4G},
journal = {IEEE Communications Magazine},
year = {2001},
volume = {39},
number = {8},
month = {August},
annote = {This article notes the increasing demand for wireless communication, and presents speculations that future 4G networks will have several heterogenous network technologies. 4G is believed to be an "all-IP" architecture. To ensure QoS they propose a wireless application layer, which is a generic framework combining local error and traffic control. The WAL is situated between IP and the link layer, with modules adapted for the relevant access technology. The article reiterates the problem of differentiate congestion losses from losses based on bit errors. The snoop protocol is a good candidate to cope with this. A figure with bit error distributions is included for a 802.11b network. For example, when using 11Mbit/s, 30 percent of the packets never exceed 10b/packet. The average error burst length associated with that error is approx 1.5 bits. Reed-Solomon codes can correct more than 25 percent of the erroneous packets.},
bibdate = {Monday, November 05, 2001 at 09:14:43 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Krasic01a,
author = {Charles Krasic and Kang Li and Jonathan Walpole},
title = {The Case for Streaming Multimedia with {TCP}},
journal = {OGI (Oregon Graduate Institute) CSETechnical Report, CSE-01-003},
year = {2001},
month = {March},
annote = {The paper revisit and challenge the dogma that TCP is an undesirable choice for streaming multimedia, video in particular. It is argued that providing smooth video quality requires relatively large receiver buffers. The buffers can then also accomodate TCP retransmissions and the variability in rate that results from congestion control. Employing a new transport protocol is hard and the gains for streaming media may be marginal. A QoS adaptive video system that was originally designed for a network with simple QoS support, priority-labeled packets is presented. It is based on a priority drop and can support multiple adaptation strategies (temporal and spatial). A system for streaming the QoS adaptive video over TCP is then described.},
url = {Krasic_streaming_TCP_techreport.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Monday, November 05, 2001 at 09:14:52 (CET)}
}
@article{Horn01,
author = {G. Horn and P. Knudsgaard and S. Lassen and M. Luby and J. Rasmussen},
title = {A Scalable and Reliable Paradigm for Media on Demand},
journal = {IEEE Computer},
year = {2001},
volume = {34},
number = {9},
pages = {40-45},
month = {September},
annote = {In this article, Gavin B. Horn and his colleagues address the problems faced when buildning a media-on-demand (MoD) system for live streaming. They present a survey over existing MoD strategies including user-centered MoD schemes and data-centered MoD schemes. Especially data-centered schemes are discussed. In particular, the principles for the data-centered MoD schemes pyramid broadcasting and harmonic broadcasting are presented. However, the emphasis of the article is on the MoD scheme designed by the authors. Their scheme is based on so-called Luby transform codes, a FEC coding scheme developed by Michael Luby, a chief technology officer and founder of Digital Fountain. Luby transform codes differ from other commonly used FEC codes, e.g. Reed Solomon codes, in that it only requires a constant number of XOR operations per coded symbol and are able to code a limited number of packets into an essentially infinite stream of unique packets. The principal idea with Luby codes are that an Luby encoder generates interchangeable packets, which means that the client can receive any packets in any order. Only the number of packets received determines when the client can reconstruct the original segment. In their MoD scheme, the media server partitions a media file into segments of varying sizes and then transmits each segment on a separate multicast group. The client starts by joining one or more multicast groups and collects the Luby-encoded packets. The client only starts playing back the media file once it has received enough packets to decode the entire first segment. Thus, all clients with approximately the same packet loss will experience the same start-up latency, independent of when they start.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, November 12, 2001 at 08:52:34 (CET)}
}
@article{Elsen01,
author = {I. Elsen and F. Hartung and M. Kampmann and L. Peters},
title = {Streaming Technology in 3G Mobile Communication Systems},
journal = {IEEE Computer},
year = {2001},
volume = {34},
number = {9},
pages = {46-52},
month = {September},
annote = {By offering data-transmission rates up to 384 Kbps for wide-area coverage and 2 Mbps for local-area coverage, 3G systems will be able to provide high-quality streamed Internet content to the rapid growing mobile market. The authors of this article and their colleagues at Ericsson Eurolab have built a platform for multimedia communication in 3G systems that follows the 3GPP mobile streaming standard. To demonstrate the capabilities of their mobile multimedia platform in a network application, they have developed demonstrators for GPRS and simulated UMTS networks. The GPRS demonstrator was an entertainment application that allows mobile users to view movies on demand. Although bandwidth was limited to 14.4 Kbps, video and audio quality were both good. To demonstrate the platform in UMTS, they employed a cinema ticketing application. This application enables mobile users to preview and by tickets for films. Also in this scenario, the audio and video quality were found acceptable.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, November 12, 2001 at 08:55:47 (CET)}
}
@article{Fahmi01,
author = {H. Fahmi and M. Latif and S. Sedigh-Ali and A. Ghafoor and P. Liu and L. Hsu},
title = {Proxy Servers for Scalable Interactive Video Support},
journal = {IEEE Computer},
year = {2001},
volume = {34},
number = {9},
pages = {54-59},
month = {September},
annote = {Deploying proxy servers as part of a network infrastructure helps to meet the increasing demand for access to multimedia documents. Using multiple proxy servers for distributed caching achieves scalability and load balancing. In this article, a novel approach to caching of streaming media contents is proposed: hotspot caching. A hotspot is in this context an object in a video clip that indicates embedded hyperlinks to related text and video/audio information. The main incentive with all types of multimedia proxy servers is to let the proxy servers take care of the majority of small requests from interactive applications and thereby enabling for the video/audio servers to devote more connections to continuous playouts. However, in order to accomplish this, most caching schemes require huge buffers. It is in this respect hotspot caching excels. Caching only hotspot segments significantly reduces the buffer space requirements. To demonstrate the feasibility of hotspot caching, a Web-based streaming multimedia application, HyperStreaming document player, was built. In an experiment, hotspot caching was compared with video caching in terms of hit rate. The result of this experiment clearly showed that hotspot caching decreases the average response time due to improved hit rates. However, it also showed that hotspot caching can increase server load because caching smaller portions of a video file increases the number of lengthy playouts. Small requests are deferred to proxies, but servers manage long playouts. In video caching (complete caching), proxies also serve long playouts, but this requires significantly more buffer space than hotspot caching.},
bibdate = {Monday, November 12, 2001 at 08:59:14 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Eagle01,
author = {Eagle Consulting Services},
title = {Streaming Video on the Internet},
journal = {White Paper},
year = {2001},
annote = {This white paper considers how streaming video works, technical requirements, video compression standards. However, the greatest contribution of this paper is a survey over the most prevalently used video streaming technologies including RealSystem G2, Windows Media Player, QuickTime, Xing StreamWorks, VDOLive, NET TOOB Stream, and VivoActive. I think it is important as a researcher to always have some knowledge of what is happening in the industry, and I think this paper give some insights into which approaches are taken by the leading players in the Streaming Media business. For myself, I recommend you to read this paper.},
bibdate = {Monday, November 19, 2001 at 13:51:27 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Mascolo99,
author = {S. Mascolo},
title = {Congestion Control in high-speed communication networks using the Smith principle},
journal = {Automatica},
year = {1999},
volume = {35},
pages = {1921-1935},
annote = {The major contribution of this paper, as I see it, is that it takes a control theoretic approach to congestion control in ATM and TCP/IP networks. Classical control theory provides an established set of tools to design congestion control algorithms whose performance can be predicted analytically, rather than relying on simulations and small-scale experiments. The important advantage of mathematical analysis is that it allows us to derive the properties of control laws in a general setting, whereas simulations and experiments only cover particular cases. To begin with, Mascolo, develops an analytical model for a general store-and-forward packet switching network. Worth noting is that his network model assumes per-flow buffering, which is done to uncouple congestion and fairness issues. In a traditional FIFO-queueing network model it is very difficult to enforce congestion control while at the same time upholding max-min fairness. There are, however, drawbacks with per-flow queueing, for example it impedes scalability, therefore active research is devoted to find single-queue solutions which approximate per-flow buffering. Based on a simple transfer function for the bottleneck buffer, Mascolo develops a Smith regulator which controls the bottleneck queue level. The reason for employing a Smith regulator is that this type of control scheme enables full compensation for the feedback delay, i.e. the control system becomes equivalent to a system with the delay moved out of the control loop. In the following, it is shown that the developed Smith regulator i stable and guarantees full link utilization. Futhermore, the Smith regulator is evaluated against ERICA, an explicit rate congestion control algorithm developed by Jain et al. ERICA requires that switches compute an explicit rate. Thus, the complexity of maintaining per-flow buffering, as is the case for the Smith regulator, are replaced by the problem of computing the explicit rate. A particularly hard task is the measurement of the available bandwidth which is mostly bursty due to interaction with VBR traffic. In the last part of the paper, Mascolo shows that TCP/IP indeed already employs a Smith regulator for congestion control. This result has serveral ramifications. For example, it gives an important validation of the Smith regulator developed in this paper since the congestion control mechanism of TCP is the only congestion control algorithm really tested in an actual large-scale setting.},
submitter = {Karl-Johan Grinnemo},
url = {papers/Mascolo99_TCP_Smith.pdf},
bibdate = {Monday, November 19, 2001 at 13:55:38 (CET)}
}
@article{Charny95,
author = {A. Charny and D. Clark and R. Jain},
title = {Congestion Control With Explicit Rate Indication},
journal = {In Proceedings of IEEE International Conference on Communications},
year = {1995},
pages = {1954-1963},
annote = {As the number of so-called "long fat pipes" increases, window-based congestion control schemes face significant challenges. Large window sizes and long feedback delays enable large packet bursts to enter the network leading to servere congestion. This dilemma has led to many researchers arguing that a rate-based approach would be a better alternative. In this paper, an asynchronous distributed algorithm for optimal rate calculation is proposed. This algorithm is shown to have several attractive properties. In particular, it is shown that the algorithm quickly converges to the optimal sending rates. Furthermore, it is found to be well-behaved in transience. The cornerstone of the proposed algorithm is that each switch/router calculates its available per-flow capacity which it signals to the source of the flow. The signaling is either done through explicit control packets or through specific fields in the packet headers. Each source maintains an estimate of its optimal sending rate which initially could tentatively be set to the desired sending rate of the application. In the control packets or the header of the data packets, there are two fields. The first field is only one bit long and is called the "u-bit". The second field is several bits long and is used by the switches/routers to communicate the next rate estimate to the source. This field is called the "stamped rate" field. When the source sends a packet, it puts its current rate estimate in the "stamped rate" field and clears the "u-bit". When the packet arrives at a switch, the switch compares the stamped rate in the packet with its pre-calculated per-flow rate ("advertised rate"). If the "stamped rate" is higher than or equal to the "advertised rate", the rate in the "stamped rate" field is reduced to the "advertised rate" and the "u-bit" is set. If the "stamped rate" is less than the "advertised rate", the switch/router does not change the packet control fields. When the packet reaches the destination, the "stamped rate" field contains the bottleneck bandwidth. The destination returns the "stamped rate" field to the source which adjusts its sending rate to the returned "stamped rate". The proposed algorithm is shown to give each competing flow over a bottleneck link its max-min fair share. Furthermore, it is proved that the algorithm converges after three roundtrips.},
bibdate = {Monday, November 19, 2001 at 13:58:41 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Veres01,
author = {A. Veres and M. Boda},
title = {The Chaotic Nature of {TCP} Congestion Control},
journal = {In Proceedings of IEEE INFOCOM'00},
year = {2000},
month = {March},
annote = {Until recently it was believed that the TCP congestion control always behaved in a nice periodic and predictable manner. For example, this was one of the assumptions for the model of the TCP congestion control developed by Padhye et al. In this paper, it is shown that for some parameter settings the TCP congestion control exhibits a deterministic chaotic behavior. While always exhibiting a periodic behavior, the length of the period of some parameter of the TCP congestion control scheme (e.g. the size of the congestion window) can become extremely large. In particular, if the evolution of an arbitrary parameter in the TCP congestion control is depicted as a trajectory, this trajectory in some cases has a fractal dimension. Furthermore, the trajectory converges to an attractor which in those cases the trajectory has a fractal dimension is a so-called strange attractor. It is also shown in this paper that in certain cases the TCP congestion control is very sensitive to the the initial conditions which means that very small perturbations in the system may lead to large differences. In an experiment, the authors measured the mean size of the Lyapunov exponent to 1.11, which means that after a perturbation the difference between the two systems increase at an average rate of 3.03 units/sec. Finally, it is demonstrated that the TCP congestion control creates self-similar traffic with Hurst parameters showing both short-range and long-range dependence depending on the values of the congestion control parameters. More importantly, it is shown that the self-similarity of the traffic generated by TCP does not depend on the particular applications on top of TCP, which is a common belief, but instead is an effect of the TCP congestion control mechanism itself. The deterministic chaotic behavior exhibited by the TCP congestion control generates self-similar traffic.},
bibdate = {Monday, November 19, 2001 at 14:02:13 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Raisinghani00,
author = {V. Raisinghani and A. Patil and S. Iyer},
title = {Mild Aggression: A new approach for improving TCP Performance in Asymmetric Networks},
journal = {In Proceedings of Asian International Mobile Computing Conference},
year = {2000},
month = {November},
annote = {In an asymmetric network there is a large difference in bandwidth between forward and reverse channels. This asymmetry leads to large variations in ACK arrival rate, something which impairs the performance of TCP. In particular, it results in slower initial window growth and burstiness on the forward channel. In this paper, a modification of the TCP congestion control mechanism is proposed which strives to improve the initial sluggish growth of the congestion window in an asymmetric environment. This is done by introducing two so-called mild aggression factors. The key idea behind these aggression factors is that they shall compensate for the asymmetry in the forward and reverse link delay. The first aggression factor reduces the estimated round-trip time, while the second aggression factor makes the increment of the congestion window larger during slow start than 1 packet per round-trip. Simulations comparing TCP with mild aggression with Asymmetric TCP, a previous proposal to handle asymmetry in networks, show that the performance of TCP in terms of throughput can be significantly improved by chosing mild aggression instead of Asymmetric TCP.},
bibdate = {Monday, November 19, 2001 at 14:05:07 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Braden00,
author = {R. Braden and D. Clark and J. Wroclawski},
title = {Developing a Next-Generation Internet Architecture},
journal = {Internal Whitepaper, MIT Lab for Computer Science},
year = {2000},
month = {July},
annote = {The design of today's Internet technology was guided by an Internet architecture that was developed in the 1970s. At present, we are vitnessing that current reality with new and changing requirements are eating away at the viability of the original Internet architecture. Much of the coherence and consistency of the original architecture is being lost in a patchwork of technical ad hoc solutions and overlays, each intended to solve a particular few requirements without any considerations on its consequences on the overall Internet framework. The authors of this paper, all of which are well renowned researchers in the Internet research community, express a deep concern of the need to re-evaluate the current Internet architecture. In so doing, they emphasize the need to rewrite the list of fundamental requirements underlying the current Internet architecture. They suggest that such a list should include aspects on internetworking, robustness, heterogeneity, management, cost effectiveness, ease of attachment, and accountability. There are also som crucial points that they think will permeate a new Internet architecture. First, the Internet of today is not the same as it was in the 1970s. Today, the number of stake-holders are magnitudes larger and much more diversified. There are probably no singe list of requirements that should satify all stake-holders. Therefore, a new Internet architecture must deal with a multi-ordered requirements set; with many requirements taking on different importance at different times, and in different regions of the network. The authors are actually proposing to introduce ``meta-requirements'' that assure interoperability between different parts of the Internet. Second, they believe that a future architecture must take into account the needs and concerns of commercial providers if it is to be accepted and thus to be able to influence the overall direction of the design of the Internet. Third, the authors points to the need of a future architecture to focus on those issues not taken into account by the current architecture, e.g. mobility, policy-driven auto-configuration, highly time-variable resources, allocation of capacity. The authors conclude this paper with some guidelines on how a project responsible for designing a new Internet architecture should be organized and work. There are some parts of this discussion which might be a little bit surprising. For example, the believe that in the inception of the project it should only be 6-10 persons involved. Not until a basic architecture has been design should the project be enlarged to encompass a larger group of people. Furthermore, the do not see it as a good idea that the project is under the auspices of IAB since a new architecture probably needs something new, and IAB are today to a large extent populated by network practitioners who more care about short-term solutions than taking a long-term responsibility for the coherence and consistency of Internet.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, November 19, 2001 at 14:08:36 (CET)}
}
@article{Liu97,
author = {Hang Liu and Hairuo Ma and Magda El Zarki and Sanjay Gupta},
title = {Error control schemes for networks: an overview },
journal = {Mobile Networks and Applications},
year = {1997},
volume = {2},
number = {2},
pages = {167-182},
month = {October},
annote = {This well-written paper provides a nice overview of the techniques used to combat errors in wireless links. Discusses FEC with subgroups Block coding (BCH & RS) and convolutional coding. Discusses code shortening and puncturing to adapt block and convolutional codes respectively. Contains a section on code selection and also mentions interleaving. The discussion of ARQ is rather short and does not mention sliding windows but state that GBN takes the best of Stop&Wait and Selective Repeat which is debatable. After the discussion on Hybrid ARQ practical examples of error control are given with examples from GSM, IS-95 and Wireless ATM.},
url = {papers/Liu97ErrorControlOverview.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, November 20, 2001 at 08:28:58 (CET)}
}
@article{Vickers99,
author = {Brett J. Vickers and B.R. Badrinath},
title = {A Generalizable Service Architecture for Mobile Networks },
journal = {The Sixth IEEE International Workshop on Mobile Multimedia Communications (MOMUC'99), San Diego, USA},
year = {1999},
pages = {221-225},
annote = {This paper introduces describes an architecture based on Lottery Based Multiple Access (LBMA) to distribute transmission slots to hosts in a wireless networks. A number of lotteries are performed by the base station, grouped by colors (ie service classes). The article also mentions a MAC protocol for lottery ticket requests based on slotted ALOHA and a hierarchical provisioning model. The paper is discussion only, do not contain any experimental results, and lacks details.},
url = {papers/Vickers99MobileNetwLottery.pdf},
bibdate = {Tuesday, November 20, 2001 at 08:34:29 (CET)},
submitter = {Johan Garcia}
}
@article{Oliphant00,
author = {Malcolm W. Oliphant},
title = {Radio interfaces make the difference in 3G cellular systems},
journal = {IEEE Spectrum},
year = {2000},
volume = {37},
number = {9},
pages = {53-58},
month = {October},
annote = {A nice and short paper providing an overview of the radio interface evolution and also mentions a number of relevant radio-level concepts in an easilyunderstood manner. Presents the major structural differences between 2, 2.5 & 3G. Concludes with the observation that it finding the services that is the problem, not the technology.},
url = {papers/Oliphant00RaIf3G.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, November 20, 2001 at 08:38:36 (CET)}
}
@article{Staehle01,
author = {Dirk Staehle and Kenji Leibnitz and Konstantin Tsipotis},
title = {QoS of Internet Access with GPRS},
journal = {The Fourth ACM International Workshop on Modeling, Analysis and Simulation of Wireless and Mobile Systems (MSWiM 2001)},
year = {2001},
annote = {Presents a simulation study of GPRS capacity versus wireline modem access as the session arrival rate varies. Uses an elaborate source model and also models RLC. Losses are said to simulated by assigning more RLC datablocks to one IP packet by adding 7,5% to the number of blocks, no details of this reasoning is provided. CS2 is ussed and the network is provisioned for 1% voice blocking probability and uses on-demand PDCHs only. No mention of RTTs or buffering is done,a dn GGSN and SGSN is not simulated. The paper is well written though. The TCP throughput is ca 30kbps and lowers as the session arrival rate increase.},
url = {papers/Staehle01GPRS_QoS.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, November 20, 2001 at 08:53:39 (CET)}
}
@inproceedings{Raman00a,
author = {Suchitra Raman and Hari Balakrishnan and Murari Srinivasan},
title = {{ITP}: {A}n Image Transport Protocol for the {I}nternet},
booktitle = {Proceedings Intl. Conference on Network Protocols},
year = {2000},
month = {November},
annote = {},
url = {papers/Raman00_ITP_.ps.gz },
submitter = {Johan Garcia},
bibdate = {Friday, November 30, 2001 at 22:51:35 (CET)}
}
@article{Liu00,
author = {Chunlei Liu and Raj Jain},
title = {Improving explicit congestion notification with the mark-front strategy},
journal = {Computer Networks},
year = {2000},
volume = {35},
pages = {185-201},
month = {February},
annote = {Current ECN implementations mark the packet from the tail of the queue. In this paper, the authors propose the mark-front strategy to send an even faster congestion signal. Compared with the mark-tail policy, the mark-front strategy has three advantages. First, it reduces the buffer size requirement at the routers. Second, it provides more up-to-date congestion information to help the source adjust its window in time to avoid packet losses and link idling and thus improves link efficiency. Third, it improves fairness among old and new users and helps alleviate TCP's discrimination against connections with large round-trip time. With a simplified model, the buffer size requirement for both mark-front and mark-tail strategies are analyzed. Link efficiency, fairness and more complicated scenarios are tested with simulations. The results show that the mark-front strategy achieves better performance than the current mark-tail policy. The mark-front strategy is also applied to the RED algorithm. Simulations show that the mark-front strategy used with RED has similar advantages over mark-tail.},
url = {Liu-ECN-markfront.pdf},
bibdate = {Monday, December 03, 2001 at 08:40:08 (CET)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Florissi01,
author = {P. G. S. Florissi and Y. Yemini and D. Florissi},
title = {QoSockets: a new extension to the sockets API for end-to-end application QoS management},
journal = {Computer Networks},
year = {2001},
volume = {35},
number = {1},
pages = {57-76},
month = {January},
annote = {This paper presents the QoSockets APIs that promote code portability and reusability by sheltering heterogeneity in the QoS functions offered by several transport protocols. According to the authors the main advantages of such an approach are the following: (1) A single API that is independent of transport layer specifics. The same application can use services from several transport protocols without any modification. (2) The runtime offers a single QoS negotiation mechanism, which automatically bridges gaps among different transport protocol providers. (3) Upgrades to support new protocols or new OSs can be accomplished by extending the runtime with the interface to the new architecture components. (4) The QoSockets runtime can automatically select the most appropriate transport given QoS requirements. (5) The runtime can automatically monitor the QoS delivered with low overhead. The collected data may be accessed by other local applications as well as external SNMP managers. QoSockets also includes an architecture for QoS management using QoS MIBs. The most interesting part is to see how they specified the QoS. Also contains some interesting references.},
url = {papers/Florissi-QoSocket.pdf},
bibdate = {Monday, December 03, 2001 at 08:55:46 (CET)},
submitter = {Anna Brunstr\"{o}m}
}
@article{Rendon01,
author = {J. Rend\'{o}n and F. Casadevall and J. L. Faner},
title = {Wireline {TCP} Options Behaviour in the {GPRS} Network},
journal = {Proceedings of the IST Mobile Communications Summit},
address = {Barcelona, Spain},
year = {2001},
month = sep,
annote = {The TCP SACK and Timestamps options performance in a busy GPRS network is evaluated through simulations. Due to the highly variable delays of the busy GPRS network analyzed, the SACK option is beneficial, but the Timestamps option degrades the performance. The description of the simulation model and results is fairly brief. The simulation model is described as follows. A FTP server, with the corresponding TCP and IP layers, models the Fixed Host. TCP-Reno version is assumed. The Internet cloud is modeled by means of the loss packet probability and a delay. The delay is statistically characterized as a gaussian random variable. The GGSN is represented with a router, whereas the SGSN is modeled as a fixed delay that represents the node process delay. The GGSN-SGSN and SGSN-BSS links are also modeled with fixed delays, which take into account the limited link capacities, i.e. 2Mbps and 64 Kbps respectively. The Mobile Host is modeled with a TCP/IP host like the FH. The radio link is statistically modeled by means of the packet loss probability as well as by the LLC frame delay histogram. A GPRS radio link with 4 PDCHs, Coding Scheme CS4, a C/I relation of 24 dB and 15 Mobile Stations is assumed. Three types of bursty traffic have been considered in the GPRS radio link, namely: e-mail, WWW and FTP. },
url = {papers/Rendon-SACK-TS.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Monday, December 03, 2001 at 09:13:56 (CET)}
}
@article{Arbor01,
author = {Craig Labowitz and Abha Ahuja and Michael Bailey},
title = {Shining Light on Dark Address Space},
journal = {Technical Report, Arbor Networks (http://research.arbornetworks.com)},
year = {2001},
pages = {9},
month = {November},
annote = {Conventional wisdom holds that the Internet topology represents a complete graph, or spanning tree of inter-connected networks. The authors have collected BGP data under three years, and have come to the conclusison that this is not the case; More than five percent of Internet routed prefixes constitue "dark address space", which is the range of topology reachable from one provider but unreachable from one or more competitor networks. Partitioned networks include cable modems and US military networks. The primary sources of dark address space include agressive path prefix length and IRR filtering, as well as misconfiguration. They also found that short-lived BGP annoucements were used to send unsolicited email.},
url = {Arbor01_dark_address_space.pdf},
bibdate = {Tuesday, December 04, 2001 at 10:09:46 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Ashmawi01,
author = {W. Ashmawi and R. Guerin and S. Wolf and M. Pinson},
title = {On the Impact of Policing and Rate Guarantees in Diff-Serv Networks: A Video Streaming Application Perspective},
journal = {Proceedings of ACM SIGCOMM'2001},
year = {2001},
pages = {83-95},
month = {August},
annote = {The paper explores how policing actions and rate guarantees provided by the Expedited Forwarding PHB in diffserv affects the perceived quality of a video stream. Experiments are carried out over QBone, a QoS enabled part of Internet2, and over a local network. The general conclusion is that the token rate needs to be larger than the encoding rate of the video to achieve high quality. Increasing the token bucket size can also improve quality, but at the cost of possibly inserting EF bursts into the network.},
url = {Ashmawi01_policing_impact_in_diffserv_a_video_streaming_perspective.pdf},
bibdate = {Tuesday, December 04, 2001 at 16:20:06 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Sasse00,
author = {A. Watson and M. A. Sasse},
title = {The Good, the Bad and the Muffled: the Impact of Different Degradations on Internet Speech},
journal = {ACM Multimedia 2000},
year = {2000},
pages = {269-276},
annote = {Presents an experiment comparing the relative impact of different types of degradation on subjective quality ratings of interactive speech transmitted over packet-switched networks. The results from the experiment confirm that the effects of volume differences, echo and bad microphones are rated worse than the level of packet loss most users are likely to experience on the Internet today. Uses biometrics (heardrate, blood pressure) to determine the stress of test-subjects. The paper also includes a mapping of lay-man terms to the particular degradation type (the other end sounds "muffled", "fuzzy", "metallic", etc), which could be used to better aid end users in solving quality problems.},
submitter = {Stefan Alfredsson},
bibdate = {Tuesday, December 04, 2001 at 16:34:38 (CET)}
}
@article{Adamic01,
author = {L. A. Adamic and B. A. Huberman},
title = {The Web's Hidden Order},
journal = {Communications of the ACM},
year = {2001},
pages = {55-59},
month = {September},
annote = {Web site growth and popularity follow rules that can be explained mathematically and are useful for predicting tehe Web's future behavior. Observes that site pages, visitors, inlinks, outlinks follows a power law distribution. The papers does not present proofs, but shows graphs and makes observations/discussions.},
url = {Adamic01_webs_hidden_order.pdf},
bibdate = {Tuesday, December 04, 2001 at 17:23:05 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Huston00,
author = {G. Huston},
title = {The Future for TCP},
journal = {The Internet Protocol Journal},
year = {2000},
volume = {3},
number = {3},
pages = {2-27},
month = {September},
annote = {Today, the Internet spans a broad base of uses, and ensuring that TCP performs effiently in all contexts is a continuing task. This paper surveys mechanisms proposed to adapt TCP to as diverse contexts as short-duration sessions, satellite communication and mobile wireless communication. In addition, the discussion is extended to the Stream Control Transmission Protocol (SCTP) and the reason TCP is found unsuitable for signaling in public telephone networks. The paper concludes with a contemplation of the evolution of TCP and states the importance of maintaining a coherent control architecture and avoiding radical changes to TCP that may stress the deployed network into congestion collapse.},
url = {papers//Huston00_Future_for_TCP.ps.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, December 19, 2001 at 12:18:59 (CET)}
}
@article{Stallings01,
author = {W. Stallings},
title = {{Mobile IP}},
journal = {The Internet Protocol Journal},
year = {2001},
volume = {4},
number = {2},
pages = {2-14},
month = {June},
annote = {Mobile IP was developed to enable computers to maintain Internet connectivity while moving from one Internet attachment point to another. This paper gives a general overview of the operation Mobile IP. For those looking for a succinct description of Mobile IP that is easy accessible, yet informative, then this paper is just what you are looking for.},
bibdate = {Wednesday, December 19, 2001 at 12:25:18 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@misc{Xie01,
title = {{SCTP} Unreliable Data Mode Extension},
author = {Q. Xie et al.},
month = apr ,
year = 2001,
note = {Work in Progress},
annote = {This Internet Draft describes an extension to the Stream Control Transmission Protocol (SCTP) to provide semi-reliable data transfer. The benefits of this extension includes a unified congestion control for reliable as well as unreliable data traffic.},
submitter = {Karl-Johan Grinnemo},
status = {INTERNET DRAFT}
}
@article{Karn00,
author = {P. Karn},
title = {{IP} Data Services over {CDMA} Digital Cellular},
journal = {Mobile Computing and Communications Review},
year = {2000},
volume = {4},
number = {4},
pages = {30--35},
month = {October},
url = {papers/p30-karn.pdf},
annote = {In this article, the author has compiled the majority of aspects of data over CDMA mobile systems. Examples of aspects considered are resource allocation, multipath fading, the properties of the Radio Link Protocol (RLP), the High Data Rate (HDR) CDMA system and issues concerning TCP/IP over CDMA. CDMA's big advantage over the traditional multiple access methods, frequency division multiple (FDMA) and time division multiple access (TDMA), is that the number of CDMA spreading codes is practically infinite in comparison to the severly limited numbers of frequency channels or time slots typically available in FDMA or TDMA. However, there are drawbacks with CDMA too. For example, since CDMA is not an orthogonal access method, as is FDMA and TDMA, it is not even in theory build a receiver that completely rejects unwanted frequency channels. Since the spreading codes of CDMA are effectively unlimited, CDMA offers dynamic resource allocation. In FDMA and TDMA transmission resources are allocated for an entire call, whether or not a party is actually talking; it is impractical to dynamically allocate a frequency channel or time slot each time a party talks. Like FDMA and TDMA, IS-95 CDMA allocates a connection-oriented traffic channel to each call. But unlike FDMA and TDMA, only the hardware resources are actually dedicated to a given IS-95 call; the air resources are dynamically shared among active calls. IS-95 introduced a "soft handoff" where the old and new cells both transmit at the same signal to a mobile during the handoff period and the mobile continually selects the better of the two as they independently fade. Once the mobile has moved solidly into the area served by the new cell, the signal from the old cell is turned off. Even though IS-95 originally was designed for voice traffic, it is also suitable for data. IS-95 is able to use the same mechanisms as developed for variable rate voice for data. In contrast, data services implemented in FDMA and TDMA must either waste capacity on established connections when a user has nothing to send, or implement a complex channel contention scheme. There are many more advantages with CDMA, too many to be mentioned in this comment. However, one thing worth mentioning about TCP over CDMA is that Karn kills the myth that most errors on IS-95 links are caused by radio noise. The capacity of CDMA system is primarily limited to mutual interference among its users. Unwanted signals appear as noise to a CDMA receiver, and in terrestial CDMA systems this inter-user "noise" is much more significant than the "true" noise often thought of as causing most errors on the wireless link.},
bibdate = {Thursday, January 17, 2002 at 08:45:44 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Castelluccia01,
author = {C. Castelluccia},
title = {Extending Mobile IP with Adaptive Individual Paging: A Performance Analysis},
journal = {Mobile Computing and Communications Review},
volume = {5},
number = {2},
pages = {14--26},
month = {April},
annote = {This paper proposes an extension to Mobile IP with an adaptive paging mechanism to reduce the generated signaling load in the network. In the proposed scheme, each mobile computes their optimal location area according to their mobility and incoming call parameters to reach the minimal signaling traffic. Simulations show that the proposed scheme saves significant bandwidth.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, January 17, 2002 at 08:53:06 (CET)}
}
@article{Venkataram01,
author = {P. Venkataram and R. Rajavelsamy and S. Laximaiah},
title = {A Method of Data Transfer Control during Handoffs in Mobile-IP based Multimedia Networks},
journal = {Mobile Computing and Communications Review},
year = {2001},
volume = {5},
number = {2},
pages = {27--36},
month = {April},
annote = {In this paper a mechanism is proposed that enables continuity of data transfer during handoffs in mobile-IP networks and minimizes the data loss during the period. The mechanism performs efficient buffering at the base station by using so-called toggled buffers and calculates the optimal playout time for multimedia applications. Tests have been performed both in-house as well as in wired and wireless environments. The results of the tests showed a significant improvement in the continuity of data transfer during multiple handoffs while running multimedia applications such as video-on-demand and audio-on-demand.},
bibdate = {Thursday, January 17, 2002 at 09:02:25 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Shannon01,
author = {C. Shannon},
title = {A Mathematical Theory of Communication},
journal = {Mobile Computing and Communications Review},
year = {2001},
volume = {5},
number = {1},
pages = {3--55},
month = {January},
annote = {In February 2001, Claude Shannon, the well-renowned mathematician and founder of the subject Information Theory, died. In honor of this brilliant mathematician MC2R has republished the 1948 paper by Shannon "A Mathematical Theory of Communication". In this paper, Shannon lays the foundation of what later will become the subject Information Theory. It is in this paper, he first presents his model of a communication system comprising an information source, transmitter, noise source, receiver and destination. Based on this model, he defines concepts such as channel capacity, entropy etc. Even though the paper includes some mathematics, it is well worth browsing through. },
bibdate = {Thursday, January 17, 2002 at 09:20:06 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Korhonen01,
author = {J. Korhonen and O. Aalto and A. Gurtov and H. Laamanen},
title = {Measured Performance of {GSM} {HSCSD} and {GPRS}},
booktitle = {Proceedings of the IEEE Conference on Communications},
year = {2001},
month = {June},
annote = {TCP performance was measured for bulk data transfer and request-reply transfer over GSM HSCSD and GPRS. For stationary connections the performance was stable for both HSCSD and GPRS. When the mobile terminal was moving the performance was highly variable. TCP could not adapt to the variable quality of service. This resulted in unnecessary retransmissions. HSCSD performed better than GPRS because the variance was higher for GPRS.},
url = {papers/icc01.pdf.gz },
submitter = {Annika Wennstr\"{o}m},
bibdate = {Monday, January 28, 2002 at 08:44:20 (CET)}
}
@inproceedings{Rendon01a,
author = {J. Rendon and F. Casadevall and L. Garcia and R. Jimenez},
title = {Characterization of the {GPRS} radio interface by means of a statistical model},
booktitle = {Proceedings of the IEEE Vehicular Technology Conference (VTC-Spring)},
address = {Greece},
year = {2001},
month = {May},
annote = {The paper describes the development of a statistical model of the GPRS radio interface. The model is suitable for simulations of FTP connections. The input parameters to the model are: number of PDCHs, C/I, coding scheme, and number of mobile stations. As output the model gives LLC packet error rate and a cumulative distribution function for packet delay. },
bibdate = {Monday, January 28, 2002 at 15:31:27 (CET)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Han99a,
author = {Richard Han},
title = {Factoring a Mobile Client's Effective Processing Speed Into the Image Transcoding Decision},
journal = {Proc. 2nd ACM International Workshop on Wireless Mobile Multimedia (WOWMOM), Seattle, USA},
year = {1999},
pages = {91-98},
annote = {Presents a number of inequalities that can be used for controlling the transcoding decision for both store-and-forward as well as streamed image transcoding. Since some operations such as GIF decoding, color conversion and scaling can create considerable processing delay on slow PDA's it is wise to partition the functionally accordingly.},
url = {papers/Han99_processi_transcode.pdf},
bibdate = {Wednesday, January 30, 2002 at 15:54:14 (CET)},
submitter = {Johan Garcia}
}
@article{Chandra00,
author = {Surendar Chandra and Carla Schlatter Ellis and Amin Vahdat},
title = {Differentiated Multimedia Web Services Using Quality Aware Transcoding},
journal = {Proceedings of INFOCOM 2000},
year = {2000},
month = {March},
annote = {Compares the behaivour of a webserver providing differntiated service to the users. Four different server behaviours are defined to use when bandwidth overload occurs: Traditional, Modbandwidth, Denial and Transcoding. Transcoding has several versions of each image at different compression level and chooses higher compressed images when the server becomes heavily loaded. Experiments show that transcoding provides best user experience to the most users since it degrades gracefully.},
url = {papers/Chandra00_Diff_multimedia_web_transcoding.pdf},
bibdate = {Wednesday, January 30, 2002 at 16:31:54 (CET)},
submitter = {Johan Garcia}
}
@article{Iren00,
author = {Sami Iren and Paul Amer},
title = {SPIHT-NC: Network-Conscious Zerotree Encoding},
journal = {Proc. Data Compression Conference (DCC '2000), Snowbird, USA},
year = {2000},
pages = {313-322},
month = {April},
annote = {Presents the SPIHT-NC coding scheme that adds some information to each ADU to make ADUs that arrive out of order useable. The large dependence on the first packewt that contains the coarse scale coefficients however remain. Experiments presented show that the progressive display perfomance of SPIHT-NC begins to be better than regular SPIHT at losses of 10% for a 9.6kbps link.},
url = {papers/Iren00_SPIHT-NC.ps.gz},
bibdate = {Wednesday, January 30, 2002 at 16:51:51 (CET)},
submitter = {Johan Garcia}
}
@book{Box87,
author = {George E. P. Box and Norman R. Draper},
title = {Empirical model-buiding and response surfaces},
publisher = {John Wiley and Sons, Inc},
year = {1987},
annote = {Statistics book on response surface methodology},
submitter = {Katarina Asplund},
bibdate = {Friday, February 08, 2002 at 14:30:34 (CET)}
}
@article{Huffman52,
author = {D.~A. Huffman},
title = {A method for the construction of minimum redundancy codes},
journal = {Proceedings of the Institute of Electronics and Radio Engineers},
year = {1952},
volume = {40},
pages = {1098--1101},
submitter = {Johan Garcia},
bibdate = {Tuesday, February 12, 2002 at 12:30:34 (CET)},
annotation = {Describes huffman coding}
}
@article{Gurtov02,
author = {A. Gurtov and R. Ludwig},
title = {Evaluating the {Eifel} Algorithm for {TCP} in a {GPRS} network},
journal = {Proceedings of European Wireless 2002, Florence, Italy},
year = {2002},
month = {February},
annote = {Annika: The Eifel algorithm is evaluated with TCP Reno, NewReno, and SACK. Eifel detects and responds to spurious timeout events and spurious fast retransmits. By the use of timestamps an acknowledgment of an original segment can be distinguished from an acknowledgment of a retransmitted segment. Unnecessary performance degradation and unnecessary retransmissions are avoided. In GPRS spurious timeout events may occur due to delay spikes caused by cell reselection. Delays between a few seconds up to a few tens of seconds have been measured in a live network. Ns-2 is used because then it is possible to reproduce sequences of cell reselection and the TCP implementations in Ns-2 are flawless. Performance is measured as download time and goodput. Goodput is defined as the ratio of unique segments to the total number of segments transmitted, eg. a goodput of 0.97 implies that 3\% of the data is retransmitted. For receiver window limited connections, Eifel reduces download time up to 12\%, and goodput is increased up to 20\%. Also for network limited connections Eifel increases goodput by almost 10\%, but download time was increased for Eifel with TCP Reno. Eifel with NewReno or SACK performs better than Eifel with Reno.},
url = {Gurtov_Eifel_in_GPRS.pdf},
submitter = {Anna Brunstr\"{o}m},
bibdate = {Thursday, February 21, 2002 at 16:52:03 (CET)}
}
@article{Gurtov01,
author = {A. Gurtov},
title = {Effect of Delays on {TCP} Performance},
journal = {Proceedings of IFIP Personal Wireless Communications 2001, Lappeenranta, Finland},
year = {2001},
month = {August},
annote = {Annika: Long sudden delays, which occur in wireless WANs, can trigger spurious TCP timeouts. To illustrate that this is in fact a problem, some measurements of handovers in a live GPRS network are included. Handover delays of more than ten seconds were not uncommon. The effect of a long sudden delay on standard TCP is that a spurious timeout occurs. Due to the ACK ambiguity problem all outstanding data is retransmitted (go-back-N) and when the unnecessary retransmitted segments arrive at the receiver DUPACKs are sent in response. The DUPACKs trigger spurious Fast retransmit. The Seawind emulator (Kojo01) was used to examine the TCP implementations of FreeBSD 4.1, Windows 98, Linux 2.2, and Linux 2.4. SACK was disabled in all the TCPs. After a long sudden delay all implementations first used a go-back-N retransmission policy. FreeBSD and Windows 98 performed poorly since a spurious Fast retransmit and additional timeouts occurred. Some implementation faults were also found in FreeBSD and Windows 98. In Linux 2.2 a spurious Fast retransmit was triggered and then data was retransmitted by the NewReno algorithm (RFC2582). Linux 2.4 was even more efficient, since it uses the Careful version of NewReno. The author recommend NewReno (Careful version with disabled Fast retransmit after timeout), D-SACK (RFC2883), and Eifel.},
url = {papers/Gurtov_Effect_Delays_TCP_Performance.pdf},
bibdate = {Thursday, February 21, 2002 at 16:55:02 (CET)},
submitter = {Anna Brunstr\"{o}m}
}
@article{RLC00,
author = {ETSI},
title = {{Digital cellular telecommunications system (Phase 2+); General Packet Radio Service (GPRS); Mobile Station (MS) - Base Station System (BSS) interface; Radio Link Control/ Medium Access Control (RLC/MAC) protocol (GSM 04.60 version 6.9.0 Release 1997)}},
journal = {ETSI},
year = {2000},
url = {Loke},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thursday, February 21, 2002 at 20:58:20 (CET)}
}
@article{LLC00,
author = {ETSI},
title = {{Digital cellular telecommunications system (Phase 2+); General Packet Radio Service (GPRS); Mobile Station - Serving GPRS Support Node (MS-SGSN) Logical Link Control (LLC) layer specification (GSM 04.64 version 6.7.0 Release 1997)}},
journal = {ETSI},
year = {2000},
url = {Loke},
bibdate = {Thursday, February 21, 2002 at 21:00:23 (CET)},
submitter = {Annika Wennstr\"{o}m}
}
@inproceedings{McCanne93,
author = {S. McCanne and V. Jacobson},
title = {The {BSD} packet filter: a new architecture for user-level packet capture},
booktitle = {Proceedings of the Winter 1993 USENIX Conference},
year = {1993},
pages = {259--69},
month = {January},
url = {papers/bpf-usenix93.ps.Z},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thursday, February 21, 2002 at 21:51:32 (CET)}
}
@inproceedings{Wennstrom01,
author = {Annika Wennstrom and Johan Garcia and Anna Brunstrom and Jan H. Gustafsson},
title = {{TCP} and {GSM} Link Layer Interactions: Implications for the Wireless Internet},
booktitle = {Proceedings of {IEEE} Vehicular Technology Conference ({VTC}-01 Spring)},
pages = {2198 - 2202},
volume = {3},
year = {2001},
month = {May},
url = {papers/Paper725.ps},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thursday, February 21, 2002 at 23:36:16 (CET)}
}
@article{Jones01,
author = {Gareth T Joneas and David J Parish and Iain W Philiips},
title = {A Transform domain feature detection and concealment algorithm for errors in DCT encoded images},
journal = {Computers \& Graphics},
year = {2001},
volume = {25},
number = {4},
pages = {671-680},
annote = {Presents a technique for error concealment in the DCT domain based on feature detection. The lost blocks neighbours will have their three lowest freq AC coefficients ratios evaluated in order to find direction and displacements for an edge passing through the lost block. The classifier needed to perform this is not described in detail. If such an edge is found the three AC coefficient values need in the lost block to reconstruct the edge is calculated and inserted in order to conceal the loss as a straight egde through the block. Seems to be an efficient method to reconstruct the AC information to some extent. },
url = {papers/Jones01DCTDomainErrconc.pdf},
bibdate = {Sunday, February 24, 2002 at 15:17:36 (CET)},
submitter = {Johan Garcia}
}
@article{Pickering96,
author = {M.R.Pickering and M.R. Frater and J.F. Arnold and M.W. Grigg},
title = {An error concealment technique in the spatial frequency domain},
journal = {Signal Processing},
year = {1996},
volume = {54},
number = {2},
pages = {185-189},
annote = {Suggest an error concealment that handles spurious bit errors in JPEG coded data. Places the VLC coded data in EREC frames so that each block is resynchronizable. Also doesnt use predictive DC-coefficient coding to stop error propagation. The main idea is to calculate an AC standard deviation (ASD) for each coefficient in the image and use this to detect bit errors that cause correct huffman decoding but leads to very large coefficient values which in turn produces visible artifacts. Reported experiments suggest that 6 ACD is a suitable value to use.Coefficients outside in nulled together with the rest of the coefficients in the block. Was specifically developed for an surveillance application which allowed no extra information to be inserted during error concealment, i.e. interpolation could not be used in either in the pixel or coefficient domain.},
url = {papers/Pickering96SpatialErrConc.pdf},
submitter = {Johan Garcia},
bibdate = {Sunday, February 24, 2002 at 17:35:57 (CET)}
}
@article{Wang01,
author = {Ching-Ynag Wang and Shiuh-Ming Lee and Long-Wen Chang},
title = {Designing {JPEG} quantization tables based on human visual system},
journal = {Signal Processing:Image Communication},
year = {2001},
volume = {16},
pages = {501-506},
annote = {Presents a JPEG quantization table based on Daly's model of the HVS which describes the sensitivity to different spatial frequencies. Using this model a image-independent quantization table can be produced which is shown to provide better quality than the example quantization table in the standard.},
url = {papers/Wang01HVSjpegQuantTbl.pdf },
submitter = {Johan Garcia},
bibdate = {Sunday, February 24, 2002 at 19:33:52 (CET)}
}
@article{Shen98,
author = {Bo Shen and Ishwar K. Sethi},
title = {Block-based manipulations on transform-compressed images and videos},
journal = {Multimedia Systems},
year = {1998},
volume = {6},
number = {2},
pages = {113-124},
annote = {This paper describes compressed domain operations based on the presented inner block transforms (IBT). Subtitling and image rotation are described. Image resampling is also mentioned. },
url = {papers/Shen98DCTDomainManip.pdf},
bibdate = {Sunday, February 24, 2002 at 19:51:04 (CET)},
submitter = {Johan Garcia}
}
@article{Huang00,
author = {Yu-Len Huang and Ruey-Feng Chang},
title = {Error Concelament Using Adaptive Multilayer Perceptrons (MLPs) for Block-Based Image Coding},
journal = {Neural Computing \& Applications},
year = {2000},
volume = {9},
number = {2},
pages = {83-92},
annote = {Describes error concealment using neural networks. Classifies missing blocks to one of five classes depending on the block intensity gradient; smooth, left, right, up or down. Different networks is used for each class. Arbitrary sizes of blocks can be processed and the missing blocka are iteratively provessed one outer pixel frame at the time using the two outside frames as inputs to the network. Preproceesing is used to create frames for blocks with neighbouring blocks lost. },
url = {papers/Huang00ErrConcNeurNetw.pdf},
submitter = {Johan Garcia},
bibdate = {Sunday, February 24, 2002 at 20:09:02 (CET)}
}
@article{Eckhardt98,
author = {David A. Eckhardt and Peter Steenkiste},
title = {Improving Wireless LAN Performance via Adaptive Local Error Control},
journal = {Proceedings of IEEE ICNP '98},
year = {1998},
annote = {The paper argues for the use of local error control over wireless links. An experimental LAN has been built, where three error control mechanisms have been added; local retransmission, adaptive packet shrinking and adaptive error coding. Experiments show that "non-snooping local retransmission can deliver vast improvements in TCP performance over lossy wireless LAN links, allowing TCP to use a high fraction of the available link bandwidth.},
url = {papers/Eckhardt98_adaptive_local_error_control.pdf},
bibdate = {Monday, February 25, 2002 at 09:41:38 (CET)},
submitter = {Stefan Alfredsson}
}
@inproceedings{Balan01,
author = {R. K. Balan and B. P. Lee and K. R. R. Kumar and L. Jacob and W. K. G. Seah and A. L. Ananda},
title = {{TCP HACK: TCP} Header Checksum Option to improve Performance over Lossy Links},
booktitle = {Proceedings of 20th IEEE Conference on Computer Communications (INFOCOM)},
address = {Anchorage, Alaska, USA},
year = {2001},
month = apr,
annote = {TCP HACK adds two options to TCP; the header checksum option and header checksum ACK option. If the options are enabled, an extra checksum of the TCP header is added, allowing the receiver to distinguish between errors in the header and the payload. The sender is informed of a received but broken packet via the checksum ACK option, and can resend the packet without initiating slowstart or other techniques to combat congestion. Both sender and receiver must be aware of the modification.},
url = {papers/Balan01_TCP_HACK.pdf},
bibdate = {Monday, February 25, 2002 at 09:47:05 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Huston01,
author = {Geoff Huston},
title = {TCP in a Wireless World},
journal = {IEEE Internet Computing},
year = {2001},
volume = {5},
number = {2},
annote = {The article presents an overview of the problems with TCP in a wireless environment; TCP was designed for wire-based carriage and makes numerous assumptions that are typical for such environments. In a wireless environment, FEC codes and adaptive coding may result in high latency and high variability, depending on the number of retransmissions performed. Instead, the problem can be adressed at the TCP level, with fast retransmit, fast recovery and SACK. However, they are also probably inadequate to allow TCP to function efficiently over all forms of wireless systems, and the problem with TCP over wireless should be adressed separately. One approach is link level signalling via ICMP "corruption experienced" notifications, but problems arise when TCP header may be unreliable (where should the notification be sent?). Another interesting food for thought is how to manage link outages; If a mobile wireless receiver enters an area with no coverage for a short while, the sender will perform exponential backoff, because of believed congestion. When connectivity is restored, the backoff may cause a long delay before the transmission is resumed. An idea to solve this is for the receiver to keep the last packet sent, and resend this when the link condition is good, thus indicating to the sender that it may restart the transmission earlier. The article is available online at http://www.computer.org/internet/v5n2/w2stan.htm},
bibdate = {Monday, February 25, 2002 at 11:58:32 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Dugad01,
author = {Dugad R and Ahuja N},
title = {A fast scheme for image size change in the compressed domain},
journal = { IEEE Trans. on Circuits and Systems for Video Technology},
year = {2001},
volume = {11},
number = {4},
pages = {461-474},
month = {April},
annote = {Describes fast image up- and down-sampling.},
url = {papers/Dugad01DCTDomImgSize.ps.gz},
submitter = {Johan Garcia},
bibdate = {Monday, February 25, 2002 at 15:51:50 (CET)}
}
@article{Lombardo01,
author = {A. Lombardo and G. Schembra and G. Morabito},
title = {Traffic Specifications for the Transmission of Stored MPEG Video on the Internet},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {5--17},
month = {March},
annote = {Specifying QoS guarantees for realtime transmission of stored video in the Internet is a challenging task for video-on-demand (VoD) systems. In the IETF IntServ working group, two QoS classes have been defined: Guaranteed Service and Controlled-Load Service. For both of these classes it is necessary to provide the traffic sources with the capability of calculating the traffic characteristics to be forwarded to the network, i.e. the TSpec. In this paper, an analytical model is proposed that enables traffic sources to calculate the TSpec parameters for MPEG1 video streams. The video source is modeled as a switched Batch Bernoulli Process (SBBS), thereby taking into account both intra- and inter-GoP correlations. },
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 27, 2002 at 13:26:43 (CET)}
}
@article{Puri01,
author = {R. Puri and K-W Lee and K. Ramchandran and V. Bharghavan},
title = {An Integrated Source Transcoding and Congestion Control Paradigm for Video Streaming in the Internet},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {18 -- 32},
month = {March},
annote = {Today, it is widely agreed that efficient media streaming over the Internet to a large part depends on the interaction between the application and the transport protocol. In this paper, a congestion-aware video streaming system is presented. The two key components of the system are a FEC-based multiple description coding transcoder (MD-FEC transcoder) and a transport protocol with a novel congestion control scheme. The MD-FEC transcoder splits the video stream into a number of resolution layers which are concatenated to equally many descriptions. Each description contains all layers, and the \textit{i}th layer is decodable when i or more descriptions have arrived at the receiver. Something which is attained using a Red-Solomon code. In order to conserve bandwidth, MD-FEC performs the partitioning of the video stream using a nearly-optimal optimization algorithm based on Lagrange multipliers. Contrary to other proposed algorithms, the optimization algorithm used by MD-FEC executes in time O(N). The novel congestion control scheme used by the transport protocol is an augmentation of TCP's Linear Increase Multiplicative Decrease (LIMD). LIMD with History (LIMD/H), which is the name of the congestion control scheme, features a simple loss differentiation mechanism to discern between congestion-induced and non-congestion-induced packet losses. In LIMD/H, the sending rate is adjusted periodically in periods called epochs. At the end of each epoch, the receiver computes the number of received packets during that epoch and sends this information back to the sender in a congestion feedback message. Based on this message, the sender calculates the packet-loss frequency which in turn governs the LIMD/H rate control algorithm. Simulations using ns-2 suggest that the proposed video system is both viable and better than traditional approaches.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 27, 2002 at 13:32:14 (CET)}
}
@article{Furini01,
author = {M. Furini and D. Towsley},
title = {Real-Time Traffic Transmissions over the Internet},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {33--40},
month = {March},
annote = {Providing bandwidth guarantees for multimedia applications in the Internet is not without problems. One of the major problems is that multimedia streams often have variable bandwidth requirements. To solve this problem, the Premium Service class has been proposed for DiffServ. However, Premium Service uses a Bandwidth Allocation Mechanism (BAM) based on the peak sending rate, which means that when Premium Service is used for transmission of a VBR stream (e.g. a video stream), a large amount of bandwidth could be wasted. The first contribution of this paper is the introduction of a new BAM which provides the same QoS as the Premium Service but uses less bandwidth. Briefly, the proposed BAM starts allocating the peak rate, the same as Premium Service, but then it progressively reduces this allocation as the peak rate of the remaining multimedia stream decreases. This is possible since only stored media is considered, i.e. media with known characteristics. Simulations indicate that the bandwidth reductions achieved with the proposed BAM ranges from 5% to 25%. The second contribution is suggestions for frame dropping algorithms to be used with MJPEG and MPEG. Experiments conducted suggest that the frame dropping algorithms suggested for MJPEG result in bandwidth reductions in the neighborhood of 20%. Similar experiments with the algorithms for MPEG gave bandwidth savings in the order of 40%.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 27, 2002 at 13:36:30 (CET)}
}
@article{Radha01,
author = {H. Radha and M. der Shaar and Y. Chen},
title = {The MPEG-4 Fine-Grained Scalable Video Coding Method for Multimedia Streaming over IP},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {53--68},
month = {March},
annote = {Due to the wide variation of available bandwidth, video streaming over the Internet requires scalable coding methods. In this paper, the scalable video-coding framework of the MPEG-4 standard, formally called the Fine-Granular-Scalability (FGS) is described. This framework comprises several so-called tools, three of the important ones being: FGS video coding method, Adaptive Quantization and hybrid temporal-SNR scalability. The focus of this paper is on the design and the implementation of these tools. Furthermore, some preliminary evaluation results are presented that show the viability of the tools.},
bibdate = {Wednesday, February 27, 2002 at 13:39:23 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Cetinkaya01,
author = {C. Cetinkaya and V. Kanodia and W. Knightly},
title = {Scalable Services via Egress Admission Control},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {69--81},
month = {March},
annote = {Allocating resources for multimedia traffic flows with realtime performance requirements is an important challenge for future packet networks. However, in large-scale networks, individually managing each traffic flow has fundamental scalability limitations. In this paper, a new framework for scalable QoS provisioning termed Egress Admission Control (EAC). The key idea with this framework is that reservation is only performed at the egress routers without maintaining per-flow state in the network core. In particular, to establish a new session, a resource reservation message is generated by the user with contains the traffic specification and QoS requirements. For the traffic specification, any deterministic model may be used, e.g. leaky bucket. Contrary to the majority of other proposed admission control frameworks, the traffic specification for a flow in EAC is only used when the flow itself is being established; admission of future sessions are based on measurements of aggregate traffic. The QoS parameters available in EAC are packet-loss probability and delay bound. The resource reservation message is forwarded to the egress router which makes the admission decision. The key point here is that only egress routers processes a reservation request message, all intermediary nodes merely forward the message and neither perform admission control nor store state information. In this way EAC achieves scalability. The technique used in the admission control algorithm for EAC is based on the measurement-based theory of envelopes, which enables the egress router to accurately monitor both arrival and service rates. More specifically, the EAC admission control algorithm bases the admission decision on a measurement-based service envelope. The performance of EAC is evaluated through a series of simulations. Scenarios are considered which exhibit highly variable traffic loads, unknown cross traffic and several packet service disciplines. In conclusion, the simulations indicate that EAC is indeed able to accurately control the network load while still achieving efficient utilization of the network resources.},
bibdate = {Wednesday, February 27, 2002 at 13:42:41 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Su01,
author = {X. Su and B. Wah},
title = {Multidescription Video Streaming with Optimized Reconstruction-Based DCT and Neural-Network Compensations},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {123--131},
month = {March},
annote = {Packet and compression losses are two sources of quality losses when streaming video over unreliable networks like Internet. In this paper, two approaches for concealing such losses are propsed. The first approach concerns a multidescription coding scheme. Multidescription coding divides the video stream into equally important substreams such that the decoding quality is not dependent on which particular substreams are received but on how many substreams are received. In the proposed approach, the different substreams are assigned distinct packets. Packets containing descriptions for one GOB is called an interleaved set and the number of related descriptions from one GOB the interleaving factor. Moreover, each packet may carry more than one description from different GOBs. For example, woth an interleaving factor of two, packet 0 may carry the first description of the first two GOBs, whereas packet 1 may carry the second description. To compensate for quantization errors introduced by MDC, an artifical-neural-network based compensation is proposed. The neural network scheme is trained offline on a training set. Experimental results show that the two approaches complement each other in improving playback quality.},
bibdate = {Wednesday, February 27, 2002 at 13:44:55 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Grinnemo02,
author = {K-J Grinnemo and A. Brunstrom},
title = {A Simulation Based Performance Analysis of a {TCP} Extension for Best-Effort Multimedia Applications},
booktitle = {35th Annual Simulation Symposium ({ANNSS35})},
year = {2002},
address = {San Diego, CA, USA},
month = {apr}
}
@article{Xie01a,
author = {Q. Xie and R. R. Stewart and C. Sharp and I. Rytina},
title = {{SCTP} Unreliable Data Mode Extension},
journal = {Internet draft draft-ietf-tsvwg-usctp-00.txt, Work in progress},
year = {2001},
month = {October},
annote = {Later version than Xie01? Abstract This memo describes an extension to the Stream Control Transmission Protocol (SCTP) [RFC2960] to provide unreliable data transfer services. The benefits of this extension includes unified congestion control over reliable and unreliable data traffics, single association for multi-type content data services, link level fault tolerance for unreliable data applications, unreliable data stream multiplexing, etc. },
url = {papers/draft-ietf-tsvwg-usctp-01.txt},
submitter = {Johan Garcia},
bibdate = {Monday, March 04, 2002 at 17:07:01 (CET)}
}
@article{Svanbro00,
author = {Krister Svanbro},
title = {Lower Layer Guidelines for Robust HC},
journal = {Internet Draft, draft-ietf-rohc-rtp-lower-layer-guidelines-03.txt},
annote = {Describes the lower layer guidelines for robust header compression (ROHC) and the requirements ROHC put on lower layers. The key points are that; lower layers should be able to detect errors in (compressed) headers. It is recommended that invalid packets are passed to the decompressor as it might make use of the packet anyway, but it should then be marked as invalid. ROHC requires atleast that packet length can be inferred from any underlying link layer. It is recommended that demultiplexing of flows onto logical channels is performed. This will reduce the need for context identification in the header compression scheme. To maintain efficient ROHC operation, handover events should not cause significant long events of consecutive packet loss. Unequal error detection/protection schemes may be used to detect errors in the header or protect from errors occuring.},
bibdate = {Monday, March 11, 2002 at 08:47:52 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Jonsson02,
author = {Lars-Erik Jonsson},
title = {Requirements for ROHC IP/TCP Header Compression},
journal = {Internet Draft, draft-ietf-rohc-tcp-requirements-03.txt},
annote = {The draft contains a compilation of requirements to allow TCP header compression. A compression scheme should be transparent to the end points, must not require modification to existing IP, must support IPv4 and IPv6, should compress mobile IP headers (IP tunneling), must be general (handle arbitrary TCP streams), should compress IPSEC sub-headers, support packets with TCP options even if options are not compressed. It should further provide low relative overhead (performance/spectral). Short lived TCP connections should be efficiently compressed (lot of context information is sent at the start of a connection, implying "context-sharing", and "context-reuse"). The compression scheme must be able packet misordering, and efficiently handle moderate misordering (2-3 packets).},
submitter = {Stefan Alfredsson},
bibdate = {Monday, March 11, 2002 at 08:55:21 (CET)}
}
@article{Kohler02,
author = {Kohler, Handley, Floyd, Padhye},
title = {Datagram Control Protocol (DCP)},
journal = {Internet Draft, draft-kohler-dcp-02.ps},
annote = {Specifies the Datagram Control Protocol which implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. Key features: unreliable flow of datagrams with acknowledgements; reliable handshake for connection setup and teardown; reliable negotiation of options; optional mechanism to allow the sender to know with high reliability which packets reached the receiver; Congestion control incorporating ECN and ECN Nonce; and path MTU discovery. DCP is intended for applications that require the flow-based sematics of TCP, but which do not want TCP in-order delivery and reliability mechanisms, or which would like different congestion control dynamics than TCP.},
bibdate = {Monday, March 11, 2002 at 09:48:11 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Askelof01,
author = {Joel Askel\"{o}f, Charilaos Christopoulos, Mathias Larsson Carlander, Fredrik \"{O}ijer},
title = {Wireless image applications and next-generation imaging},
journal = {Ericsson Review},
year = {2001},
number = {2},
annote = {This article briefly discusses third-generation-imaging services, including multimedia messaging services, which is proposed to perhaps be one of the most important services in the future. It also discusses some requirements being put on new imaging applications and formats, and describes the newly finalized ISO/IEC standard JPEG2000.},
url = {2001021_ericsson.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:20:08 (CET)}
}
@article{Deshpande01,
author = {Sachin Deshpande, Wenjun Zeng},
title = {Scalable Streaming of JPEG2000 Images using Hypertext Transfer Protocol},
journal = {MM?OZ, Sept. 30-Oct. 5, 2001, Ottawa, Canada.},
year = {2001},
month = {September},
annote = {This paper describes a scalable architecture for streaming of JPEG2000 images, using HTTP. JPEG2000 support large images and thus downloading the entire image at its full resolution can take a long time depending upon the user's connection speed. By using HTTP for streaming JPEG2000 images, images can be hosted by web servers and streamed to the proposed client helper application. Since only the requested part of the codestream is streamed to the client, users can pick and choose their viewing experience based on their available bandwith and the capability of the display device, thus resulting in a scalable solution. The proposed approach can handle resolution, quality and region of interest scalability features of JPEG2000 images.},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:25:07 (CET)}
}
@article{Rabbani02,
author = {Majid Rabbani, Rajan Joshi},
title = {An overview of the JPEG2000 still image compression standard},
journal = {Signal Processing: Image Communication 17 (2002)},
year = {2002},
url = {papers/science__ob_overview.pdf},
bibdate = {Monday, March 11, 2002 at 17:29:47 (CET)},
submitter = {Hannes Persson}
}
@article{Taubmana02,
author = {David Taubman, Erik Ordentlichb, Marcelo Weinberger, Gadiel Seroussic},
title = {Embedded block coding in JPEG 2000},
journal = {Signal Processing: Image Communication 17 (2002)},
year = {2002},
url = {papers/science__ob1_Embedded.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:31:56 (CET)}
}
@article{Marcellina02,
author = {Michael W. Marcellina, Margaret A. Lepleyb, Ali Bilgina, Thomas J. Flohrc, Troy T. Chinend, James H. Kasnere},
title = {An overview of quantization in JPEG 2000},
journal = {Signal Processing: Image Communication 17 (2002)},
year = {2002},
url = {papers/science__ob2_quantization.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:33:19 (CET)}
}
@article{Zeng02,
author = {Wenjun Zeng, Scott Daly, Shawmin Lei},
title = {An overview of the visual optimization tools in JPEG 2000},
journal = {Signal Processing: Image Communication 17 (2002)},
year = {2002},
url = {papers/science__ob3 visual optimization.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:34:43 (CET)}
}
@article{Askelof02,
author = {Joel Askelof, Mathias Larsson Carlander,Charilaos Christopoulos},
title = {Region of interest coding in JPEG 2000},
journal = {Signal Processing: Image Communication 17 (2002)},
year = {2002},
url = {papers/science__ob4 ROI.pdf},
bibdate = {Monday, March 11, 2002 at 17:35:38 (CET)},
submitter = {Hannes Persson}
}
@article{Santa-Cruz02,
author = {Diego Santa-Cruz, Rapha.el Grosbois, Touradj Ebrahimi},
title = {JPEG 2000 performance evaluation and assessment},
journal = {Signal Processing: Image Communication 17 (2002)},
year = {2002},
url = {papers/science__ob_performance.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:38:01 (CET)}
}
@article{Houchina02,
author = {J. Scott Houchina, David W. Singer},
title = {File format technology in JPEG 2000 enables flexible use of still and motion sequences},
journal = {Signal Processing: Image Communication 17 (2002)},
year = {2002},
url = {papers/science__ob5 File format.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:38:55 (CET)}
}
@article{Hemani94,
author = {Sheila S. Hemani, Robert M. Gray},
title = {Subband Coded Image Reconstruction for Lossy Packet Networks},
journal = {Portions of this work were presented at the Twenty-Eighth Asilomar Conference on Signals, Systems, and Computers, November 1, 1994.},
year = {1994},
month = {November},
url = {papers/ip854.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:40:57 (CET)}
}
@article{Fukuhara01,
author = {Takahiro Fukuhara (Sony Corporation), David Singer (Apple Computer)},
title = {Motion JPEG2000 Final Committee Draft 1.0, ISO/IEC 15444-3 (JPEG2000, part 3)},
journal = {2001},
year = {2001},
annote = {This document specifies the use of the wavelet-based JPEG2000 codec for the coding and display of timed sequences of images. It has been defined by ISO/IEC JTC1 SC29/WG1 as part three of the new JPEG2000 International Standard. In this specification, a file format is defined, and guidelines for the use of the JPEG2000 codec for motion sequences are supplied. The Motion JPEG2000 file format MJ2 is designed to contain one or more motion sequences of JPEG2000 images, with their timing, and also optional audio annotations, all composed into an overall presentation.},
url = {papers/fcd15444-3.doc},
bibdate = {Monday, March 11, 2002 at 17:48:27 (CET)},
submitter = {Hannes Persson}
}
@article{Bajic00,
author = {Ivan V. Baji'c, John W. Woods, Ali Mohammed Chaudry},
title = {Robust Transmission of Packet Video through Dispersive Packetization and Error Concealment},
journal = {Proc. Packet Video Workshop 2000, Sardinia, Italy May 2000},
year = {2000},
annote = {The paper presents a packetization approach for interframe videotransmission, based on space-frequency dispersion. Subband-decomposed video data is packetized so that data appearing in any transmission packet is dispersed in both space and frequency. Errors arising from packet loss are spread across the frame which allows error concealment. },
url = {papers/PVW2000final.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:50:02 (CET)}
}
@article{Arjen01,
author = {Arjen van der Schaff, Koen Langendoen, R. Inald L. Lagendijk},
title = {Design of an Adaptive Interface between Video Compression and Transmission Protocols for Mobile Communications},
journal = {Presented at Packet Video, Korea 2001},
year = {2001},
annote = {The paper discuss the design of an adaptive interface between video compression and transmission protocols to handle QoS fluctuations that are common in mobile communication systems. Qos negotiation is identified as the most promising design alternative to obtain system-wide efficiency. By introducing abstract QoS parameters, mapped to implementation details hidden in each of the subsystems, negotiation can be done between system modules to assure efficient adaptation. The QoS parameters are selected on basis from both the video encoder and protocol modules.},
url = {papers/PV2001submit.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, March 11, 2002 at 17:52:10 (CET)}
}
@article{Li99,
author = {J-R Li and D. Dwyer and V. Bharghavan},
title = {A Transport Protocol for Heterogeneous Packet Flows},
journal = {Proceedings of IEEE INFOCOM '99},
year = {1999},
month = {March},
annote = {Current Internet transport protocols do no support the concept of heterogeneous flows; for example TCP guarantees reliable sequenced delivery of all packets in a flow, while UDP does not guarantee either delivery or sequencing. Thus, applications that have heterogeneous flows are forced to use multiple independent streams and then explicitly synchronize them at the application layer, something that both adds complexity to the application and that is less efficient for adapting to the dynamics of the network. Currently, heterogeneity in packet flows are handled by user-level protocols like RTP. However, while the authors of this paper concur with the general principles behind application-level framing, they also believe that there should be a clean separation of policies and mechanisms. The policies should be set by the application, but the mechanisms, to be effective, should be implemented in the transport protocol rather than in the application. To this end, the authors present in this paper a novel transport protocol called HPF (Heterogeneous Packet Flows) for effectively supporting heterogeneous packet flows in the Internet. The key features of HPF is 1) support for heterogeneous packet flows with different reliability, priority and timing requirements in the same connection. 2) decouples congestion control and reliability mechanisms (cf. TCP in which congestion control and reliability are intertwined) 3) support for application-level framing and extends the socket API with parameters for specifying reliability and timing requirements of each frame 4) support for application-specified priorities as hints for routers to preferentially drop low-priority packets during congestion. HPF is a connection-oriented transport protocol that guarantees sequenced delivery. It comprises three logical sublayers: the application framing sublayer which is responsible for converting back and forth between packets and frames; the windowing, reliability, timing and flow-control sublayer with is responsible for coordinating the window advancement between the sender and the receiver, flow control, reliability, deleting packets which cannot meet their deadline requirements and for providing sequencing, and finally the congestion control sublayer which is responsible for congestion control and for the estimation of the rate and round-trip time parameters for a connection. Some preliminary performance measurements have been conducted. They showed that HPF outperforms TCP. In particular, at 10% packet loss HPF is 3.8 times faster than TCP.},
url = {papers/li99.pdf},
bibdate = {Wednesday, March 13, 2002 at 08:36:16 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Mauve01,
author = {M. Mauve and V. Hilt and C. Kuhmunch and W. Effelsberg},
title = {{RTP/I} - Toward a Common Application Level Protocol for Distributed Interactive Media},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {152--161},
month = {March},
annote = {RTP/I, Real-Time Application Level Protocol for Distributed Interactive Media, has been specifically designed to be a common framework for transportation of control information related to distributed interactive applications. In particular, RTP/I has been designed to confer information about the common state of a distributed application, events, and the so-called environment of the distributed application. The environment of a distributed application comprises the static parts of the common state, e.g. the document discussed in a video conference. RTP/I reuses many aspects of RTP, including the the concept of two distinct protocols for the transportation of the data and the meta information. The protocols used for the transmission of the data is called the RTP/I data protocol while the protocol used for meta information is called the RTP/I control protocol (RTCP/I). These two protocols are carried over distinct transport addresses. The viability of RTP/I have been demonstrated through a 3-D telecooperation application as well as through the development of a number of generic services.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, March 13, 2002 at 08:45:48 (CET)}
}
@article{Servetto01,
author = {S. Servetto and K. Nahrstedt},
title = {Broadcast Quality Video over IP},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {1},
pages = {162--173},
month = {March},
annote = {This paper considers the problem of designing systems for video transmission over high-speed segments of the current Internet. Commonly, this problem is approached in either of two ways. In the first way one tries to adjust the video coder in such a way that it is able to cope with the rate fluctuations occurring at the transport level. The second approach taken is to design a transport protocol specifically targeting the needs of a video application and then use a standard video coder. The authors of this paper argues that neither of these two ways lead to satisfactory solutions. Their main argument is that both these ways adhere to the source/channel separation theorem which typically does not apply to packet networks. Instead, they mean that the transmission of video over the Internet entails a solution comprising a jointly developed video coder and transmission protocol. In this paper, a jointly designed transport protocol, RT-TCP (Real-Time TCP) and error-resilient video coder are proposed. RT-TCP is actually not a completely new transport protocol but a modified version of TCP. The three most salient modifications introduced in RT-TCP are a modification of the retransmission mechanism of TCP in such a way that delay-constraint packets are not retransmitted when their deadline have been reached, a refined acknowledgement strategy in which all packets received are explicitly acknowledged, some extra headers for timestamps and sequence numbers needed for control of real-time traffic. The key property of the proposed error-resilient video coder is that the quality of the decoded video depends only on the number of packets received, not on exactly which packets are received. The most important contribution of this paper according to the authors is the design of a middleware module in between the video coder and the transport protocol which sets the parameters of the video coder based on the channel state. The control logic of the middleware module is based on optimal stochastic control theory using a so-called hidden markov model for modeling the channel.},
bibdate = {Wednesday, March 13, 2002 at 08:49:44 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Gallardo01,
author = {J. Gallardo and D. Makrakis and M. Angulo},
title = {Dynamic Resource Management Considering the Real Behavior of Aggregate Traffic},
journal = {IEEE Transactions on Multimedia},
year = {2001},
volume = {3},
number = {2},
pages = {177--185},
month = {June},
annote = {The theory of effective bandwidth basically states that the probability of buffer overflow in a stationary queueing system decreases exponentially with the buffer size as long as the effective bandwidth of the input traffic stream is less than the server capacity. This paper shows that static allocation of buffer resources using the theory of effective bandwidth is prohibitively inefficient. As an alternative, a dynamic resource management scheme is proposed based on prediction techniques: PALB (Adaptive Leaky Bucket Assisted by Traffic Prediction). Simulations with the OPNET simulator suggest that PALB compared to conventional leaky bucket (CLB) achieved remarkable improvements. In particular the output buffer packet-loss probability was almost 100 times smaller for PALB, the mean delay was reduced by 25% and both the packet-loss rateio and the delay jitter were within the limits acceptable for video and audio traffic.},
bibdate = {Wednesday, March 13, 2002 at 08:52:58 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Squid02,
author = { },
title = {Squid Web Proxy homepage},
journal = {http://www.squid-cache.org/, checked 03/2002},
annote = {Reference to Squid},
bibdate = {Wednesday, March 13, 2002 at 10:01:37 (CET)},
submitter = {Johan Garcia}
}
@article{Ziv77,
author = {Jacob Ziv and Abraham Lempel},
title = {A Universal Algorithm for Sequential Data Compression},
journal = {IEEE Transactions on Information Theory},
year = {1977},
volume = {23},
number = {3},
pages = {337-343},
month = {May},
annote = {Describes the LZ77 string matching compression algorithm.},
bibdate = {Wednesday, March 13, 2002 at 10:22:23 (CET)},
submitter = {Johan Garcia}
}
@article{Witten87,
author = {Ian H. Witten and Radford M. Neal and John G. Cleary},
title = {Arithmetic coding for data compression},
journal = {Communications of the ACM},
year = {1987},
volume = {30},
number = {6},
pages = {520-540},
month = {June},
annote = {Describes the principles of arithmetic coding and provides code for a small C implementation in the paper.},
url = {papers/Witten87ArithmeticCoding.pdf},
bibdate = {Wednesday, March 13, 2002 at 10:33:02 (CET)},
submitter = {Johan Garcia}
}
@article{Welch84,
author = {T. A. Welch},
title = {A Technique for High-Performance Data Compression},
journal = {IEEE Computer},
year = {1984},
volume = {17},
number = {6},
pages = {8-19},
month = {June},
annote = {Describes the LZW improvement of LZ78.},
submitter = {Johan Garcia},
bibdate = {Wednesday, March 13, 2002 at 13:25:44 (CET)}
}
@article{Serrano99,
author = {N. Serrano and D. Schilling and P. Cosman},
title = {Quality evaluation for robust wavelet zerotree image coders},
journal = {Proceedings 2nd Annual UCSD Conference on Wireless Communications},
year = {1999},
pages = {128-134},
month = {March},
annote = {This paper reports on a study using humans to try to recognize image content for wavelet encoded images using two different error control techniques. The techniques are to use higher compression and RCPC FEC or to use error resilient packetization. It was found that the localized distortion produced by packetization is preferrable to the same PSNR amount of distortion for the global distortion resulting from FEC-higher compression.},
url = {papers/Serrano99QualityCompRobWav.ps.gz},
bibdate = {Wednesday, March 13, 2002 at 14:23:15 (CET)},
submitter = {Johan Garcia}
}
@article{Pentikousis02,
author = {K. Pentikousis and H. Badr and B. Kharmah},
title = {On the Performance Gains of TCP with ECN},
journal = {proceedings of ECUMN 2002, Colmar, France},
year = {2002},
month = {April},
annote = {This paper compares the performance of TCP/ECN vs. standard TCP. In contrast to previous work, the paper focuses on situations in which all clients uniformly operate under either drop tail, RED, or ECN. The main results are that ECN leads to far fewer packet drops and to a fairer sharing of the available bandwidth, but that ECN not necessarily leads to improved transfer times, and thus goodput, under uniform conditions.},
submitter = {Katarina Asplund},
bibdate = {Sunday, March 24, 2002 at 15:32:25 (CET)}
}
@article{Pentikousis01,
author = {K. Pentikousis and H. Badr},
title = {TCP with ECN: the Case of Two Simultaneous Downloads},
journal = {proceedings of AIC 2001, Rhodes, Greece},
year = {2001},
month = {July},
annote = {This paper studies the effect of ECN on TCP performance for two simultaneous large (20MB) file transfers. It is shown that the two connections share the bandwidth much fairer when ECN is used, as compared to RED or drop tail. This is only true, however, if the file transfers are sufficiently large and the maximum dropping probability in the RED router is rather high(ca 10%). The authors points out that ECN effectiveness depends on the AQM mechanism implemented at the bottleneck router. },
submitter = {Katarina Asplund},
bibdate = {Sunday, March 24, 2002 at 16:06:37 (CET)}
}
@article{Abdelzaher99,
author = {T. Abdelzaher and N. Bhatti},
title = {Web Server QoS Management by Adaptive Content Delivery},
journal = {proceedings of IWQOS 1999},
year = {1999},
month = {May},
annote = {This paper presents a new approach to web server resource management based on web content adaptation. The idea is to store the web content in multiple copies that differ in quality and size. A client is served from one of these trees depending on the current server load. Clients may also have different priorities, so that lower priority requests are degraded first upon overload.},
bibdate = {Sunday, March 24, 2002 at 16:50:38 (CET)},
submitter = {Katarina Asplund}
}
@article{Abdelzaher99a,
author = {T. Abdelzaher and N. Bhatti},
title = {Web Content Adaptation to Improve Server Overload Behavior},
journal = {International WWW conference, Toronto, Canada},
year = {1999},
month = {December},
annote = {This paper examines different ways of adapting content on the web and evaluates the resulting performance impact drawn from an analysis of representative web sites. Content can be adapted by image degradation, reduction of embedded objects per page, and by reduction of local links. It is shown in the analysis that content adaptation can lead to significant resource savings for a large category of sites. The performance improvement (from the server's perspective) comes mostly from reducing the embedded objects per page. It should be noted that HTTP 1.0 was used for the analysis.},
bibdate = {Sunday, March 24, 2002 at 17:26:14 (CET)},
submitter = {Katarina Asplund}
}
@article{Mullin01,
author = {J. Mullin and L. Smallwood and A.Watson and G. Wilson},
title = {New Techniques for Assessing Audio and Video Quality in Real-Time Interactive Communication},
journal = {IHM-HCI 2001, Lille, France},
year = {2001},
month = {September},
annote = {This tutorial introduces a framework for audio and video quality assessment based on HCI evaluation criteria (task performance, user satisfaction, and user cost), and explains how these criteria should be considered and balanced in any specific evaluation. The tutorial also explains how to put the interaction into the context within which it will be used, i.e. identify any particular characteristics of user, task and physical and/or situational context that may generate particular reuirements or preferences regarding the level and type of audio and video quality in the assessment.},
submitter = {Katarina Asplund},
bibdate = {Sunday, March 24, 2002 at 18:08:34 (CET)}
}
@article{Zhao01,
author = {Y. Zhao and C. Chen},
title = {Coupon TFRC: a mechanism being friendly to both TCP and continuous stream},
journal = {Computer Networks},
year = {2001},
volume = {37},
number = {3-4},
pages = {467-479},
month = {November},
annote = {This paper proposes Coupon TFRC, a mechanism that is friendly to both continuous stream and TCP. With this mechanism, continuous service can trade smaller loss probability with sending rate that the continuous service should own. The idea is to insert a coupon field in the TFRC header, which value is actual bandwidth / probed bandwidth. When the RED router calculates a drop probability for a Coupon TFRC packet, the coupon value is multiplied with the original drop probability to get the actual drop probability for the packet. },
submitter = {Katarina Asplund},
bibdate = {Sunday, March 24, 2002 at 19:17:59 (CET)}
}
@article{Sikdar01,
author = {B. Sikdar and S. kalyanaraman and K. Vastola},
title = {An integrated model for the latency and steady-state throughput of TCP connections},
journal = {Performance evaluation},
year = {2001},
volume = {46},
number = {2-3},
pages = {139-154},
month = {October},
annote = {This paper proposes a model for estimating the transfer times of TCP flows of arbitrary size, taking into account both timeouts and slow start phases that can occur anytime during the transfer. The model is verified using web based measurements of real life TCP connections. The paper also investigates the effect on window limitation and packet size on TCP latency.},
submitter = {Katarina Asplund},
bibdate = {Sunday, March 24, 2002 at 19:40:45 (CET)}
}
@article{Sun02,
author = {Y. Sun and F. Tsou and M. Chang Chen },
title = {Predictive flow control for TCP-friendly end-to-end real-time video on the Internet},
journal = {Computer Communications},
year = {2002},
annote = {This paper proposes a prediction-based flow/congestion control scheme called R^3CP^+ (real-time rate and retransmission control protocol plus) for real-time stored packet video transfer over the best-effort based Internet. The receiver periodically sends flow control packets to the sender to instruct how fast it should send. This sending rate is based on measurements of the packet loss probability and the RTT. Is the requested sending rate less than the desired sending rate, only packets with high significance is sent (e.g. I-frames). To cope with supply shortage during congestion, the receiver makes sure to always have one RTT's worth of packets in its buffer. Simulations show that R^3CP^+ outperforms for example LIMD schemes in all ranges },
submitter = {Katarina Asplund},
bibdate = {Sunday, March 24, 2002 at 20:45:48 (CET)}
}
@article{Conrad01,
author = {P. Conrad and G. Heinz and A. Caro and P. Amer and J. Fiore},
title = {{SCTP} in Battlefield Networks},
journal = {Proceedings MILCOM 2001, Washington, DC},
year = {2001},
month = {October},
annote = {Provides a short overview of SCTP features and compares it with TCP. Focuses on the network-level fault tolerance made possible by the use of multihoming, the DoS resiliency provided by the MAC used in SCTP connection setup and the potential performance increase possible by partial ordering. Provides some results from Conrad's Ph.D. thesis to support the argument for partial ordering. },
url = {papers/Conrad01SCTP_battle.pdf},
bibdate = {Monday, March 25, 2002 at 08:36:55 (CET)},
submitter = {Johan Garcia}
}
@article{Zhao00,
author = {Q. Zhao and P. Cosman and L. Milstein},
title = {Tradeoffs of source coding, channel coding and spreading in {CDMA} systems},
journal = {Proceedings MILCOM 2000, Los Angeles, CA},
year = {2000},
volume = {2},
pages = {846-850},
month = {October},
annote = {Presents work from Zhaos MS thesis about the subject. The overall goal is to examine the tradeoff between different parameters for image communication over spread spectrum channels. Use SPIHT wavelet coder, RCPC and CRC for channel coding and different CDMA spreading. Uses a list based Viterbi algorithm to find possible candidates and accepts or rejects these based on the CRC. In essence the different parameters allow the tradoff between getting a high quality image with a low probability or getting a a lower maximum image quality but a higher probability of achieving actually it. },
url = {papers/Zhao00Tradeoff_img_chan_cdma.pdf},
submitter = {Johan Garcia},
bibdate = {Monday, March 25, 2002 at 08:51:59 (CET)}
}
@article{JBIG93,
author = {ITU-T},
title = {Recommendation T.82 - Coded representation of picture and audio information - Progressive bi-level image compression},
journal = {Geneva, Switzerland},
year = {1993},
month = {March},
submitter = {Johan Garcia},
bibdate = {Tuesday, March 26, 2002 at 12:33:10 (CET)}
}
@article{Zimmerman80,
author = {Zimmerman, H.},
title = {{OSI} reference model - {The ISO} model of architecture for open systems interconnection},
journal = {IEEE Transactions on Communications},
year = {1980},
volume = {28},
number = {4},
pages = {425-432},
month = {April},
annote = {The seminal paper (according to Peterson) on OSI model.},
bibdate = {Thursday, April 04, 2002 at 12:11:25 (CEST)},
submitter = {Johan Garcia}
}
@article{Cerf74,
author = {Vinton Cerf and Robert Kahn},
title = {A Protocol for Packet network interconnection},
journal = {IEEE Transactions on Communications},
year = {1974},
volume = {22},
pages = {637-648},
month = {March},
annote = {Describes the fundametal design of the Internet.},
url = {papers/Cerf74ProtocolInterconnection.pdf},
submitter = {Johan Garcia},
bibdate = {Thursday, April 04, 2002 at 12:34:02 (CEST)}
}
@article{Leiner85,
author = {Leiner, B.M. and Cole, R. and Postel, J. and Mills, D. },
title = {The {DARPA} Internet Protocol Suite},
journal = {IEEE Communications Magazine},
year = {1985},
volume = {23},
pages = {29-34},
month = {March},
submitter = {Johan Garcia},
bibdate = {Thursday, April 04, 2002 at 12:45:11 (CEST)}
}
@article{Stevens98a,
author = {W. Richard Stevens},
title = {{UNIX} Network Programming, Volume 1: Networking {APIs} - Sockets and {XTI}},
journal = {Prentice Hall PTR, Upper Saddle River, NJ},
year = {1998},
submitter = {Johan Garcia},
bibdate = {Thursday, April 04, 2002 at 14:58:31 (CEST)}
}
@article{Bhargava01,
author = {R. Bhargava and A. Goel and A. Meyerson},
title = {Using Approximate Majorization to Characterize Protocol Fairness},
journal = {Proceedings of SIGMETRICS '01},
year = {2001},
month = {June},
annote = {There are broad concensus among researchers that fairness in terms of bandwidth allocation is an essential prerequisite for a transport protocol in order not to threaten the overall performance of the Internet. In this paper, it is demonstrated how approximate majorization (optimization theory) subsumes and generalizes several of the commonly used fairness criteria, e.g. max-min fairness and Jain's fairness index. In particular, a new fairness index is defined, the majorization index, and an algorithm for calculating this index is presented which is able to compute the index in polynomial time. Furthermore, two case studies are described which illustrate the use of the majorization index. In the first case study the bandwidth allocation of TCP is compared with a max-min bandwidth allocation scheme with respect to the majorization index. As expected, the max-min allocation scheme outperforms TCP for small topologies (TCP discriminates connections spanning several links). However, TCP is fairer than the max-min scheme for large topologies. A theoretical analysis reveals that the majorization index for the max-min scheme is approximately (n - 1)/2 where n is the number of nodes. This explains the rather surprising observation that TCP is more fair than a max-min bandwidth allocation scheme for larger topologies. The second case study considers the fairness of different routing schemes with respect to the majorization index. The major finding of this study was that multi-path routing seems to be fairer then single-path routing.},
url = {papers/bhargava01.ps.gz},
bibdate = {Thursday, April 04, 2002 at 17:37:18 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@techreport{Podolsky98,
author = {M. Podolsky and S. McCanne and M. Vetterli},
title = {Soft {ARQ} for Layered Streaming Media},
institution = {Computer Science Division, University of California at Berkeley},
year = {1998},
number = {UCB/CSD-98-1024},
month = {nov},
annote = { Streaming media applications over the Internet commonly use a playback buffer to mitigate fluctuations and losses in the data stream. Depending of the size of the playback buffer, it is possible to use retransmissions for recovery of lossed packets. An error-recovery scheme which enables recovery of some of the lossed packets in a data stream is typically called Soft-ARQ. The reason for calling it Soft-ARQ is that it differ from ordinary ARQ schemes in that it does not dictate retransmission of all lost packets, i.e. it prescribes a ``relaxed'' form of ARQ. While several Soft-ARQ schemes have already been proposed, this paper discusses a technique to find the optimal transmission policy of a Soft-ARQ scheme with respect to packet-loss frequency, packet lifetime and a cost function. Only those Soft-ARQ schemes in which the data stream comprises a hierarchy of layers are considered. Furthermore, the discussion is restricted to only one class of cost functions: rate-distortion functions. To capture the packet-loss effects, a Markov model is used. Analysis of a Soft-ARQ scheme with two layers and with a packet lifetime restricting the number of eligible packets for retransmission to two, shows that for low packet-loss frequencies, the best transmission policy is the one that favors the oldest, still alive, low-priority packet, while for higher packet-loss frequencies the best policy always favors the high-priority packet. An equally important finding is that for packet-loss frequencies where the best policy favors the oldest packet, the policy which favors the highest-priority packet is not only suboptimal, but is also the worst possible policy. The opposite holds for those packet-loss frequencies where the highest-priority packet policy is the optimal one. The theoretical analysis of the Soft-ARQ scheme was complemented with a simulation experiment which showed the importance to dynamically adapt the estimation of the packet-loss frequency and the the packet liftetime. However, it also suggested that dynamically optimize the transmission policy had limited effect on the performance.},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Uhlig01,
author = {S. Uhlig and O. Bonaventure},
title = {Understanding the Long-Term Selft-Similarity of Internet Traffic},
booktitle = {Quality of Future Internet Services},
year = {2001},
address = {Portugal},
month = {sep},
annote = {This paper analyzes the characteristics of Internet traffic by studying a six days trace collected at the border routers of the Belgian research ISP Belnet during December 1999. The network flow collector, NetFlow, from Cisco was used. The trace from NetFlow covers the interdomain traffic received by the ISP on all its access links and comprises 2.1 Tbytes of data. It provides the aggregated information of the transport layer flows by recording the starting time, the ending time and the total volume transferred in bytes for each unidirectional TCP and UDP flow. The time granularity of the trace is one minute, i.e. traffic is sampled over one minute periods. The study shows that the interdomain traffic is self-similar over periods spanning minutes as well as hours. While packet handling may in part be responsible for short-range traffic self-similarity, it is unlikely that it can explain large-scale self-similarity. This paper shows that the self-similarity, irrespective of time scale, can be explained by two factors. First, the traffic volume received from the sources exhibits a heavy-tailed distribution. Second, the number of sources is also self-similar. Furthermore, it is shown that bursty large sources play only a limited role for the generation of self-similar traffic, and that the number of active sources play at least the same role. A consequence of this finding is that it is possible to simulate self-similar traffic using sources generating exponentially distributed traffic (i.e. using a memoryless, short-tail distribution to generate the traffic).},
url = {papers/uhlig01.ps.gz},
submitter = {Karl-Johan Grinnemo}
}
@techreport{Uhlig01b,
author = {S. Uhlig and O. Bonaventure},
title = {The Macroscopic Behavior of {Internet} Traffic},
institution = {Infonet group, University of Namur},
year = {2001},
number = {Infonet-2001-10},
month = {jun},
annote = {This paper is in some respects a sequel to the paper ``Understanding the Long-Term Self-Similarity of Internet Traffic'' by the same authors. It presents a detailed analysis of the characteristics of interdomain traffic by studying two one week long traces of the entire interdomain traffic received by two completely different ISPs. Specifically, the only characteristic common to both ISPs is that they do not offer transit traffic. Besides this, they serve very different customers. Ideally, the traces from the ISPs would cover the same period. However, this is not the case. The first trace comprises the interdomain traffic received by BELNET during six successive days in December 1999. BELNET is the ISP that provides Internet connectivity for research and educational institutions in Belgium. In total, the trace file contains approximately 2 Terabytes of data. The second trace was collected in April 2001 and covers almost 5 successive days of interdomain traffic received by Yucom. Yucom is a commercial ISP that provides Internet access to dialup users through regular modem pools. During the five days of the trace, Yucom received roughly 1 Terabyte of data. The first contribution of the paper is that the Internet topology is becoming more and more flat. This observation is particularly evident when considering the routing tables of the studied ISPs. The first three AS hops of the Yucom ISP provide almost 80% of the reachable IPv4 address space. The corresponding figure for BELNET is 60%. The second major contribution is the observation that traffic in the Internet is not localized. The traffic originating from ASs two or more hops from the BELNET ISP constitutes no less than 89% of the total received traffic. For the Yucom ISP, as much as 94% of the total received traffic originates from ASs two or more hops away. The last major contribution is that despite the large traffic aggregation performed when analyzing the traffic at the two ISPs, the interdomain traffic flows exhibit a very high variability in volume with a difference of an order of magnitude between the average and the maximum rate of interdomain flows. This observation is particularly important when developing traffic engineering mechanisms.},
url = {papers/uhlig01b.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Neureiter00,
author = {G. Neureiter and L. Burness and A. Kassler and P. Khengar and E. Kovacs and D. Mandato and J. Manner and T. Robles and H. Velayos},
title = {The BRAIN Quality of Service Architecture for Adaptable Services with Mobility Support},
address={London, UK},
booktitle={Proceedings of the IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)},
year = {2000},
month = {September},
annote = {Provides an overview of the QoS architecture envisoned for the BRAIN project. It is designed to work well with radio based networks (hiperlan 2 is the focus) and be flexible and extensible.},
url = {papers/Neureiter00BRAIN_QoS.pdf},
bibdate = {Saturday, April 06, 2002 at 23:27:23 (CEST)},
submitter = {Johan Garcia}
}
@misc{RFC2083,
author = {T. Boutell, et. al.},
title = {{RFC 2083}: PNG (Portable Network Graphics) Specification},
journal = { },
year = {1997},
month = {March},
annote = {Describes the PNG image compression standard.},
submitter = {Johan Garcia},
bibdate = {Sunday, April 07, 2002 at 10:51:47 (CEST)}
}
@article{Strutz97,
author = {Strutz, T. and Schwarz, H. and Müller, E.},
title = {Predictive Image Coding With Adaptive Wavelet Transform.},
journal = {Proceedings of SPIE, CA, USA},
year = {1997},
volume = {3164},
pages = {279-290},
month = {July},
annote = {Presents a wavelt encoding that uses the remaining correlation between subband to create an efficient coder that produces a progressive bitstream that can be truncated anywhere.},
url = {papers/Strutz97PredicWavelet.ps.gz},
bibdate = {Monday, April 08, 2002 at 07:53:51 (CEST)},
submitter = {Johan Garcia}
}
@article{Rendon01b,
author = {J. Rend\'{o}n and F. Casadevall and D. Serarols},
title = {Snoop {TCP} Performance over {GPRS}},
journal = {Proceedings of the IEEE Vehicular Technology Conference (VTC-Spring)},
year = {2001},
month = may,
annote = {One of the proposals to increase TCP performance in wireless networks is SNOOP. This paper presents a simulation study of SNOOP in a GPRS network. The SNOOP agent is implemented at the SGSN. In a GPRS network the round trip times are highly variable which degrades TCP performance. Inaccurate round trip time estimates lead to unnecessary TCP retransmission. TCP performance is not enhanced with the use of SNOOP, because the delay between the TCP sender and receiver is almost the same as the delay between the SNOOP agent and the receiver. Timeout events occur at the same time at sender and at the SNOOP agent. },
submitter = {Annika Wennstr\"{o}m},
bibdate = {Monday, April 08, 2002 at 08:52:57 (CEST)}
}
@article{Lindemann01,
author = {Ch. Lindemann and A. Th\"{u}mmler},
title = {Performance Analysis of the General Packet Radio Service},
journal = {Proc. 21st Int. Conference on Distributed Computing Systems (ICDCS)},
year = {2001},
pages = {673-680},
month = {April},
annote = {This paper describes an analytical model of the GPRS radio interface (a continuous Markov chain). The model was used to investigate the number of PDCHs that are required in order to guarantee a QoS for GPRS users. The mobility of GSM and GPRS users is explicitly represented in the model, since arrivals of new GSM and GPRS sessions and handovers from neighboring cells are taken into account. The model was used to study how various parameters (e.g. number of PDCHs, percentage GSM/GPRS users, GSM/GPRS call arrival rate) affect average data traffic, packet loss probability, and throughput per user. },
url = {papers/ICDCS01.ps.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thursday, April 11, 2002 at 15:51:58 (CEST)}
}
@misc{RFC2883,
author = {S. Floyd and J. Mahdavi and M. Mathis and M. Podolsky},
title = {{RFC} 2883: An Extension to the Selective Acknowledgement ({SACK}) Option for {TCP}},
journal = {IETF Network Working Group},
year = {2000},
month = {July},
annote = {This RFC proposes an extension (duplicate SACK or D-SACK) to the SACK option (RFC 2018) for TCP. The first SACK option could be used by a TCP receiver to acknowledge duplicate segments. A TCP sender could use the D-SACK information to infer the order in which segments were received and if retransmissions were unnecessary. D-SACK is compatible with SACK and does not require any negotiation. The RFC describes information that could be inferred by the sender if D-SACK is used, but it does not specify what the sender should do with the D-SACK information. },
url = {papers/RFC2883},
bibdate = {Friday, April 19, 2002 at 16:09:25 (CEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Hsiao01,
author = {Hsiao, P. H. and Kung, H. T. and and Tan, K-S.},
title = {Active Delay Control for TCP},
journal = {Proceedings of IEEE Globecom},
year = {2001},
month = {November},
annote = {Preesents the Idea of using delays at either the receiver or the sender to perform rate control by increasing the percieved RTT by the protocol but not the application. (ie rate=CWND/RTT) This in turn allows to use CWNDs which are sufficiently large to perfrom fast retransmit and avoid timeouts. The avoidance of timeouts is the motivation for this scheme. The mechanism is simulated for the many-flow scenario with differing queue sizes. Interesting idea. },
url = {papers/Hsiao2001Active_delay_TCP.pdf},
bibdate = {Thursday, April 25, 2002 at 08:33:42 (CEST)},
submitter = {Johan Garcia}
}
@article{Debrunner00,
author = {Victor DeBrunner and Linda DeBrunner and Longji Wang and Sridhar Radhakrishnan,},
title = {Error Control and Concealment for Image Transmission},
journal = {IEEE Communications Surveys & Tutorials},
year = {2000},
volume = {3},
number = {1},
annote = {Presents an overview of error control and concealment. Discusses the effects of bit-errors, packet loss, packet delay and packet intrusion (ie insertion of a false packet). Discusses both receiver and sender-based approaches. Also, the paper mentions their own pixel interleaving scheme.},
url = {papers/DeBrunner00ErrContrImg.pdf },
submitter = {Johan Garcia},
bibdate = {Thursday, April 25, 2002 at 08:46:22 (CEST)}
}
@article{Pereira00,
author = {Fernando Pereira},
title = {MPEG-4: Why, what, how and when?},
journal = {Signal Processing: Image Communication 15},
year = {2000},
pages = {271-279},
annote = {A soft introduction to the MPEG-4 standard and its motivations, objectives, and standardization process and workplan. The paper is an introduction to more detailed reading in the Signal Processing series.},
url = {1 science MPEG-4 Why what how and when.pdf},
bibdate = {Thursday, April 25, 2002 at 13:10:21 (CEST)},
submitter = {Hannes Persson}
}
@article{Avaro00,
author = {Olivier Avaro, Alexandros Eleftheriadis, Carsten Herpel, Ganesh Rajan, Liam Ward},
title = {MPEG-4 Systems: Overview},
journal = {Signal Processing: Image Communication 15},
year = {2000},
pages = {281-298},
annote = {The paper gives an overview of Part 1 of ISO/IEC 14496 (MPEG-4 Systems). It first presents the objectives of the MPEG-4 activity. The description of the MPEG-4 specification follows, starting from the general architecture up to the description of the individual MPEG-4 Systems tools. Finally, a conclusion describes the future extensions of the specification, as well as a comparison between the solutions provided by MPEG-4 Systems and some alternative technologies. },
url = {2_science_MPEG-4_Systems_Overview.pdf},
submitter = {Hannes Persson},
bibdate = {Thursday, April 25, 2002 at 13:14:51 (CEST)}
}
@article{Wang98,
author = {Y. Wang and Q. Zhu},
title = {Error control and concealment for video communications: A review},
journal = {Proceedings of IEEE, special issue on Multimedia Signal Processing},
year = {1998},
pages = {974 - 997},
annote = {The paper reviews the techniques that have been developed for error control and concealment in the past 10-15 years. The techniques are described in three categories according to the roles that the encoder and decoder play in the underlying approaches: Forward Error concealment includes methods that add redundancy at the source end to enhance error resilience of the coded bit streams, Error Concealment by Postprocessing refers to operations at the decoder to recover the damaged areas based on characteristics of image and video signals, and last, Interactive Error Concealment covers techniques that are dependent on a dialogue between the source and destination. Both current research activities and practice in international standards are covered. },
url = {ER_WZ98_Error_control_and_concealment_for_video_communications_A_review.pdf},
bibdate = {Thursday, April 25, 2002 at 13:17:39 (CEST)},
submitter = {Hannes Persson}
}
@article{Wang00,
author = {Y. Wang, S. Wenger, J. Wen, and A. G. Katsaggelos},
title = {Error resilient video coding techniques},
journal = {IEEE Signal Processing Magazine, Special issue on Multimedia Communications over Networks},
year = {2000},
volume = {17},
number = {4},
pages = {61-82},
month = {July},
annote = {The paper review error resilience techniques for real-time video transport over unreliable networks. Topics covered include an introduction to today's protocol and network environments and their characteristics, encoder error resilience tools, decoder error concealment techniques, as well as techniques that require cooperation between encoder, decoder and the network. The paper provide a review of general principles of these techniques as well as specific implementations adopted by the H.263 and MPEG-4 video coding standards. The majority of the paper is devoted to the techniques developed for block-based hybrid coders using motion-compensated prediction and transform coding. A separate section covers error resilience techniques for shape coding in MPEG-4.},
url = {papers/Wang00_ErrorResilient.pdf},
bibdate = {Thursday, April 25, 2002 at 13:19:48 (CEST)},
submitter = {Hannes Persson}
}
@inproceedings{Larzon02,
author = {Lars Larzon and Ulf Bodin and Olov Schelén},
title = {Hints and Notifications },
journal = {Proceedings IEEE Wireless and Networking Conference (WCNC) },
address = {Orlando, Florida, USA},
year = {2002},
month = mar,
annote = {Propose the addition of Hints and Notifications as general means to provide cross-layer information flow. Downwards information flow takes place by means of hints, which is realized by IP-options that carries information from the higher layers, and this information can be used by the link layer to influence its behaviour. Notifications are the upwards information flow and are generated at the link layer and sent to sources that have indicated (by a hint oo) that they are capable of using notifications. The paper describes the general idea and has two example scenarios described at a fairly high level.},
url = {papers/Larzon02HintsNotifications.pdf},
bibdate = {Sunday, May 05, 2002 at 12:10:48 (CEST)},
submitter = {Johan Garcia}
}
@article{Heidemann01,
author = {J. Heidemann and K. Mills and S. Kumar},
title = {Expanding Confidence in Networking Simulations},
journal = {IEEE Network},
year = {2001},
volume = {15},
number = {5},
pages = {58--63},
month = {September},
annote = {In May 1999 the National Institute of Standards and Technology (NIST) and the Defense Advanced Research Projects Agency (DARPA) co-sponsored a workshop to discuss approaches to validate network simulations. This article summarizes some of the conclusions of that workshop. As a first step, this article clarifies the difference between verification, validation, and accreditation. It is stated that verification is a process to evaluate how faithfully the implementation of a model matches the developer's intent, as expressed by conceptual descriptions and specifications, provided either in natural language or a formal notation (cf. software functional testing). In contrast to verification, validation relates the model to the real-world phenomenon. More precisely, validation is a process to evaluate how accurately a model reflects the real-world phenomenon that it purports to represent. Accreditation is often used in the absence of technical solutions that can guarantee that a model is free from errors and will provide valid predictions. It denotes a process leading to an official declaration that a given model is fit for its intended use. The successful outcome of most accreditation processes is a written certificate signed by a recognized authority that attests that a prescribed set of processes was correctly applied during the development and testing of a simulation model. As a second step, the article points out some issues to consider when validating network simulations. Very briefly, the article emphasizes the need to clarify what is the baseline; Is it a particular implementation/realization that is the baseline or is the baseline more general? Furthermore, the article points out the importance of accommodating protocol and traffic evolution, as well as the sensitivity of the simulation. As a final step, the article gives some guidelines for successful validation and presents a case study of TCP models. In short, the guidelines states that modelers should compare simulation results with as many alternate representations as possible. This might include laboratory experiments, field tests, as well as analytical models. Furthermore, it is stated that simulation models should be validated in as many contexts as possible and should undergo both statistical and visual validation. It is also stated that simulations must be reproducible and researchers are encouraged to make their simulation models and validation tools publicly available. Finally, in those cases CPU or memory shortage prohibit a full-scale simulation, it is explicitly stated that care must be exercised to avoid introducing artificial boundaries into the model.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, May 06, 2002 at 11:31:51 (CEST)}
}
@article{Feldmann99,
author = {A. Feldmann and A. C. Gilbert and P. Huang and W. Willinger},
title = {Dynamics of {IP} traffic: A study of the role of variability and the impact of control},
journal = {In Proceedings of {ACM SIGCOMM} 1999},
year = {1999},
pages = {301--313},
month = {August},
annote = {This paper presents a systematic investigation into how and why variability and feedback-control contribute to the scaling properties observed in Internet traces. As a by-product, the paper illustrates the usefulness of wavelet-based scaling techniques in identifying and extracting regular patterns in Internet traffic. The results in the paper are based on ns-2 simulations supported by measurements conducted at an ISP. The ISP traces served as benchmarks and were used for validity of the ns-2 simulations. In conclusion the results in the paper suggest that self-similar scaling over large time scales is almost exclusively due to user-related variability and is essentially oblivious to underlying network-specific aspects. On the other hand, multifractal scaling over small time scales cannot be explained by user-related variability but is to a large degree a result of the TCP flow control. Potential practical applications of the results in this paper are for example: gain insight into the performance of a network, identifying non-TCP friendly connections, and serve as input for simulation of more realistic Internet traffic.},
url = {papers/feldmann99.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, May 13, 2002 at 09:34:16 (CEST)}
}
@article{Aurrecoechea97,
author = {C. Aurrecoechea and A. T. Campbell and L. Hauw},
title = {A Survey of {QoS} Architectures},
journal = {Technical Report NY 10027, Columbia University},
year = {1997},
annote = {In this paper, a generic QoS framework is presented. Following this, the paper surveys a number of QoS architectures found in the literature and that have been developed by the telecommunications and computer communications communities. In particular, the following QoS architectures are discussed: the Heidelberg QoS Model, the Extended Integrated Reference Model (XRM), OMEGA, IntServ, the Quality of Service Architecture (QoS-A), the OSI QoS Framework, the Tenet Architecture, the TINA QoS Framework, the MASI End-to-End Model, and the End System QoS Framework. },
url = {papers/aurrecoechea97.pdf},
bibdate = {Monday, May 13, 2002 at 09:39:16 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Larzon99b,
author = {Lars-{\AA}ke Larzon and Mikael Degermark and Stephen Pink},
title = {{UDP Lite} for Real-Time Multimedia Applications},
journal = {Proceedings of the IEEE International Conference of Communications (ICC) - Quality of Service Mini Conference},
year = {1999},
month = {June},
url = {papers/Larzon99b_udp_lite_for_real_time_mm_apps.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Monday, May 13, 2002 at 11:33:57 (CEST)}
}
@article{Lettieri97,
author = {Paul Lettieri, Christina Fragouli, Mani B. Srivastava},
title = {Low Power Error Control for Wireless Links},
journal = {Mobicom 97 Budapest Hungary},
year = {1997},
annote = {The paper studies the amount of battery energy consumed to transmit bits across a wireless link. The energy consumption includes both the physical transmission of useful and redundancy data, as well as the computation of the error control redundancy. In the studie two carriers are evaluated, IP and ATM. The studie show that for IP data transmission SACK is preffered for both good and bad channels, and for ATM data transmission SACK is preffered for good channels and SACK+FEC for bad channels, when considering energy consumed per bit as the key criterion.},
url = {p139-lettieri.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, May 20, 2002 at 08:54:55 (CEST)}
}
@article{Wah01,
author = {Benjamin W. Wah, Xiao Su},
title = {Coding and Transmission of Subband Coded Images in the Internet},
journal = {Proc. SPIE Multispectral Image Processing and Pattern Recognition, Image Compression and Encryption Technologies},
year = {2001},
volume = {4551},
pages = {10},
month = {October},
annote = {Subband-coded images can be transmitted in the Internet using either TCP or the UDP protocol. Delivery by TCP gives superior decoding quality but with long delays on unreliable network, whereas delivery with UDP has negligible delays but with degrading quality when packets are lost. First a reconstruction scheme is proposed to facilitate recovery over UDP. Secondly, studies of the delay-quality trade-offs with just UDP or combined TCP/UDP delivery of images is carried out. Result show that the proposed method is an attractive alternative to pure TCP delivery when delay-quality trade-offs is possible.},
url = {benjamin_w_wah_xiao_su.pdf},
submitter = {Hannes Persson},
bibdate = {Monday, May 20, 2002 at 09:03:11 (CEST)}
}
@article{Lilley00,
author = {Jeremey Lilley and Jason Yang and Hari Balakrishnan and Srinivasan Seshan},
title = {A Unified Header Compression Framework for Low-Bandwidth Links},
journal = {Proc. of the Sixth Annual ACM/IEEE International Conference on Mobile Computing and Networking},
year = {2000},
month = {August},
annote = {Header compression for protocols other than TCP/IP has not kept pace the growth of other protocols. This is due to complexities in implementation which requires detailed knowledge of kernel internals, and a lack of a common way of pursuing the general problem across a variety of end-to-end protocols. This paper presents a unified framework for header compression, including a simple platform-independent description language do describe header properties. A platform-specific code generation tool then generates kernel code from this specification. Provides a nice overview of header compression theory, and discusses the four main attributes of header fields; constant, delta, inferred and random fields. Comparisons have been made between generated compression and VJ compression. The results are fairly even for bulk traffic and compressing ACK's, however VJ is slightly better for interactive traffic due to special handling of echo mode.},
url = {papers/Lilley00_unified_header_compression.pdf},
bibdate = {Tuesday, May 21, 2002 at 08:18:15 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Zhang01tcp,
author = {C. Zhang and V. Tsaoussidis},
title = {{TCP R}eal: Improving Real-time Capabilities of {TCP} over Heterogeneous Networks},
journal = {Proceedings of the The 11th International Workshop on Network and Operating Systems Support for Digital Audio and Video (NOSSDAV)},
address = {Port Jefferson, New York, USA},
month=jun,
year = {2001},
annote = {The paper presents a TCP-compatible and -friendly protocol which abolishes three major shortfalls of TCP for reliable multimedia over heterogenous networks; ineffective bandwidth utilization, unnecessary congestion-oriented responses to wireless link errors and wasteful window adjustments over asymmetric, low bandwidth reverse paths. The technology is based upon the (TCP incompatible) Wave-and-Wait protocol running on top of IP. A wave is a predetermined number of fixed size data segments sent side by side, where the number of segments comprise a wave of a certain "wave level". The less percieved congestion, the higher the wave. The receiver decides about the next wave level according to wave transmission time and the current wave level. TCP-Real is using the wave-based communication pattern, by letting the receiver decide the congestion window for the sender. The wave level is communicated to the sender via ACK piggybacking and the use of TCP options. The protocol has been tested in the x-kernel protocol framework, and found to outperform Reno and Tahoe. However from the graphs it looks like the gain is quite ordinary, and at low packet error rates perform the same or worse than Reno/Tahoe. To simulate error conditions, a two-state markov chain is used with exponential distribution is used.},
url = {papers/Zhang01tcp_TCP_real.pdf},
bibdate = {Tuesday, May 21, 2002 at 08:34:41 (CEST)},
submitter = {Stefan Alfredsson}
}
% NOTE: renamed from Goel00 to Goel98, to have correct year.
% sorry for possible confusion / Stefan
@inproceedings{Goel98,
author = {Samir Goel and Dheeraj Sanghi},
title = {Improving {TCP} Performance Over Wireless Links},
booktitle = {Proceedings of the {IEEE} Region Ten Conference on Global Connectivity in Energy, Computer Communication and Control (TENCON)},
address={New Delhi, India},
year = {1998},
month=dec,
annote = {The paper proposes enhancements to TCP so that it can differentiate between a loss due to congestion in the wired network, and a loss due to noise in the wireless link. Improved performance is gained by reacting differently to these two kind of losses. The idea is that the node connecting the wired and wireless network has information about the wireless transmission success, and should be responsible for generating and transmitting the information. ICMP is used to signal wireless errors, with a new message type, ICMP-DEFER, causing the sender to postpone its RTO expiry. This avoids conflicts between the local retransmissions at the base station and end-to-end retransmissions. If a segments needs retransmission, and have been DEFERed, the cwnd is not reduced at the sender. Simulation experiments have been done, using a two-state markov chain, and contains figures of parameteres used. Improvements ranges from 7 to 24 percent, depending on the advertised window size.},
url = {papers/Goel00_improving_tcp_performance_over_wireless_links.ps},
bibdate = {Tuesday, May 21, 2002 at 08:54:14 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Tsaoussidis02,
author = {Vassilis Tsaoussidis and Ibrahim Matta},
title = {Open Issues on {TCP} for Mobile Computing},
journal = {Journal of Wireless Communications and Mobile Computing},
volume = {2},
number = {1},
pages = {3--20},
year = {2002},
annote = {Presents the design principles of TCP and identifies three shortcomings in TCP behaviour over wireless links; does not distinguish between different types of errors (bit errors vs congestion), error recovery is not responsive to handoffs and fading, and the protcol strategy that does not control the tradeoff between performance measures such as energy consumption. The key problems of TCP that require attention are error detection (tcp probing, wtcp), error recovery (freeze tcp, tcp santa cruz, tcp-real), and the protocol strategy (differentiate between performance measures) . },
url = {papers/Tsaoussidis01_open-issues-tcp-wireless.pdf},
bibdate = {Tuesday, May 21, 2002 at 08:59:49 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Blanton02,
author = {E. Blanton and M. Allman},
title = {On Making TCP More Robust to Packet Reordering},
journal = {ACM Computer Communication Review},
year = {2002},
volume = {32},
number = {1},
month = {January},
annote = {This paper illustrates the impact of reordering on TCP performance and proposes several alternatives to dynamically make the fast retransmission algorithm more tolerant of the reordering observed in the network. Reordering is compensated for by making the number of duplicate ACKs needed before retransmission a variable, dupthresh. The sender uses information provided by DSACK to determine if a spurious retransmission has occurred. If so, dupthresh is adjusted (using one out of four proposed schemes)In the case of a retransmission timeout, dupthresh is rest to three. Simulations show that a TCP that are capable of varying dupthresh performs nearly as well under reordering conditions as a standard TCP without reordering },
bibdate = {Tuesday, May 21, 2002 at 15:11:11 (CEST)},
submitter = {Katarina Asplund}
}
@article{Joo01,
author = {Y. Joo and V. Ribeiro and A. Feldmann and A. C. Gilbert and W. Willinger},
title = {{TCP/IP} Traffic Dynamics and Network Performance: A Lesson in Workload Modeling, Flow Control, and Trace-Driven Simulations},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {2},
pages = {25--37},
month = {April},
annote = {The main objective of this paper is to highlight the extent to which assumptions underlying the nature and characteristics of the network traffic influence the conclusions drawn from an experiment. To that end, two simulation experiments using the ns-2 simulator are discussed. The first simulation experiment concerns two different workload models: the first workload model comprises infinite sources that always have data to send, while the second model is a Web workload model, i.e. a heavy-tailed traffic distribution model. It is shown that the two workload models lead to very different queueing dynamics at one of the routers. While the queueing behavior with the infinite source model shows pronounced periodic fluctuations (often reffered to as synchronization effects), these effects essentially disappear with the Web workload model. The second simulation experiment concerns the impact of the TCP flow control on the traffic variability. In particular, it is illustrated how trace-driven (i.e., open loop) simulations can give rise to misleading or wrong decisions when it comes to dimensioning the size of a router queue.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, May 22, 2002 at 10:23:58 (CEST)}
}
@article{Iannaccone01,
author = {G. Iannaccone and M. May and C. Diot},
title = {Aggregate Traffic Performance with Active Queue Management and Drop from Tail},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {3},
pages = {4--13},
month = {July},
annote = {This paper examines the impact of Active Queue Management (AQM) schemes on the performance of aggregate traffic traversing a congested router. The AQM schemes studied are RED, Gentle RED (GRED), and GRED with instantaneous queue size (GRED-I). Using a testbed which more or less mimics the behavior of an ISP with a set of corporate customers, the performance of the three AQM schemes is compared with the performance of drop-tail (DT) queues in terms of the metrics: TCP goodput, TCP and UDP loss rate, queuing delay, and consecutive loss probability. The main observations of the experiment is that in terms of TCP goodput, there is essentially no difference between the AQM schemes and DT. This result contrasts with previous work which report underutilization of link capacity when DT is used. Furthermore, it is shown that the configuration of the RED parameters can have a large impact on the aggregate performance, and that the set of RED parameters that optimizes the TCP goodput for a given traffic mix not necessarily is the same set of parameters that optimizes RED in terms of other performance metrics, e.g., consecutive losses. One of the few advantages observed with the AQM schemes is that they seem to provide a shorter average queueing delay than DT. However, this come at the cost of a higher loss rate than DT. In conclusion, the authors advocate that DT remains the most appropriate queuing scheme. Maybe, GRED-I would be a viable alternative as it seems to not be as sensitive to traffic characteristics as RED.},
bibdate = {Wednesday, May 22, 2002 at 10:28:20 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Kelly01,
author = {T. Kelly},
title = {An {ECN} Probe-Based Connection Acceptance Control},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {3},
pages = {14--25},
month = {July},
annote = {In this paper, an ECN (Explicit Congestion Notification) probe-based admission control protocol for realtime non-adaptive traffic is presented. Briefly, the admission control protocol works as follows. Suppose a host A wishes to send a realtime stream to a host B with a peak rate, R, and a packet size, s. Host A will initiate the realtime stream if the proportion of probe packets that experienced congestion is less than a threshold level, eps. The probing phase starts with host A sending a packet to host B requesting B to open a so-called probe stream. Throughout the probe phase, host A calculates the current proportion of acknowledged packets that experienced congestion, p_total. When host A updates p_total, if p_total > eps + 1/n then host A stops to send probe packets, and the realtime stream is not admitted. Upon acknowledgement of the first probe packet, the probing takes place in rounds of length approximately one round-trip time. During each round, host A calculates the proportion of packets which experienced congestion, p_last. Initially, the probe packet is sent at a rate of s/RTT. This rate is then increased by 2 - p_last for each consecutive round. When the target rate R has been reached, the final round is entered. The probing terminates when host A has entered the final round and received acknowledgements for at least 1/eps probing packets. The performance of the admission control protocol in terms of delay and loss is simulated using ns-2. Three scenarios are considered: only non-adaptive traffic, multi-hop bottlenecks, and an integrated network, i.e. a network with both non-adaptive and adaptive (TCP) flows. The outcome of the simulation experiment suggests that the admission control protocol works well in the non-adaptive scenario and in the integrated scenario as long as the load is moderate. However, since the TCP flows in the integrated scenario do not use the admission control mechanism, they efficiently block out the non-adaptive flows at high traffic loads. As is to be expected, the blocking probability for multi-hop connections in the multi-hop scenario have higher blocking probability than single-hop connections. However, compared to per-hop measurement-based admission control schemes, the admission control protocol is less discriminate against multi-hop connections, i.e. it has a higher admission probability than the product of the per-hop admission probabilities.},
bibdate = {Wednesday, May 22, 2002 at 10:36:32 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@misc{RFC3155,
author = {S. Dawkins and G. Montenegro and M. Kojo and V. Magret and N. Vaidya},
title = {{RFC} 3155: End-to-end Performance Implications of Links with Errors},
year = {2001},
month = {August},
annote = {To be extended...},
bibdate = {Thursday, May 23, 2002 at 11:20:01 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Changli2001,
author = {Changli Jiao and Loren Schwiebert},
title = {Error Masking Probability of 1's Complement Checksums},
journal = {Proc 10th International Conference on Computer Communications and Networks (ICCCN), Scottsdale, Arizona},
year = {2001},
pages = {505-510},
month = {October},
annote = {(Abstract)In the transport layer of the TCP/IP protocol suite, both TCP and UDP use internet checksum to protect headers and data. Internet checksum uses 1's complement arithmetic to detect errors in the content delivered by the data-link layer. Both research and experience have shown that there are a wide variety of error sources which can not be detected by this lower layer. The error detecting performance of 1's complement checksum determines how many of these errors will be passed to higher layers, including the application layer. The performance analysis will also influence protocol design and improvement, for example, header compression. Unfortunately, previous work on this topic only determined the number of error passing patterns and the probability for 2 and 3 bit errors, and the method used for determining the probability is hard to extend to more bit errors. In this paper, we present a method to generate the formula of error passing probability. When too much calculation is needed to compute an exact result, we achieve a better estimation of the probability, which is around 3 percent of the upper bound achievable with previous techniques when 1's complement checksum is used in TCP/UDP. },
url = {papers/ChangliI2001ChecksumErrMask.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, May 24, 2002 at 03:00:47 (CEST)}
}
@article{Stewart01,
author = {Randall R Stewart AND Qiaobing Xie},
title = {Stream Control Transmission Protocl (SCTP) - A Reference Guide},
journal = {Addison Wesley},
year = {2001},
month = {November},
annote = {A book that capture the essentials of SCTP and also get into all the juicy details. The book also include a USER-SPACE implementation of the SCTP layer. Sadly it don't include a BSD socket API.},
url = {A book in the library},
submitter = {Torbj\"{o}rn Andersson},
bibdate = {Sunday, May 26, 2002 at 15:51:33 (CEST)}
}
@article{Stewart02,
author = {R. R. Stewart AND Q. Xie AND L Yarroll AND J. Wood AND K. Poon AND K. Fujita},
title = {Sockets API Extensions for Stream Control Transmission Protocol (Work in progress)},
journal = {IETF draft},
year = {2002},
month = {May},
annote = {A draft of the BSD socket API that will be used together with SCTP},
url = {http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpsocket-04.txt},
submitter = {Torbj\"{o}rn Andersson},
bibdate = {Monday, May 27, 2002 at 09:27:13 (CEST)}
}
@article{Ebrahimi00,
author = {Touradj Ebrahimi, Caspar Horne},
title = {MPEG-4 natural video conding - An overview},
journal = {Signal Processing: Image Communication 15},
year = {2000},
volume = {15},
pages = {365-385},
annote = {An overview of the MPEG-4 standard, as definded in ISO/IEC 14496-2. Target application areas range from digital television, streaming video, to mobile multimedia and games. The standard consists of a collection of tools to support these areas: tools for shape coding, motion estimation and compensation, texture coding, error resilience and scalability.},
url = {MPEG-4_natural_video_coding-An_overview.pdf},
bibdate = {Tuesday, May 28, 2002 at 15:00:17 (CEST)},
submitter = {Hannes Persson}
}
@book{Taferner02,
author = {M. Taferner and E. Bonek},
title = {Wireless Internet Access over {GSM} and {UMTS}},
year = {2002},
publisher = {Springer-Verlag},
isbn = {3540425519},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Wednesday, May 29, 2002 at 14:15:48 (CEST)}
}
@article{Rendon01c,
author = {J. Rend\'{o}n and F. Casadevall and D. Serarols and J. L. Faner},
title = {{Analysis of Snoop TCP Protocol in GPRS network}},
journal = {IEE Electronics Letters},
year = {2001},
volume = {37},
number = {10},
month = {May},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Wednesday, May 29, 2002 at 14:15:48 (CEST)}
}
@article{GSM0364,
author = {ETSI},
title = {{Digital Cellular Telecommunications System (Phase 2+), General Packet Radio Service, Overall description of the GPRS radio interface (GSM 03.64, v. 7.0.0)}},
journal = {ETSI},
year = {1998},
bibdate = {Wednesday, May 29, 2002 at 14:15:48 (CEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Kalden00,
author = {R. Kalden and I. Meirick and M. Meyer},
title = {{Wireless internet access based on GPRS}},
journal = {IEEE Personal Communications},
year = {2000},
month = {April},
annote = {The first half of the paper gives an overview of the GPRS architecture and protocols. The second half describes a GPRS simulator and simulation results. I recommend the first part of the paper to those who want an introduction to GPRS. The simulator is an event-driven protocol simulator that models web traffic in one cell over one frequency. TCP, IP, SNDCP, LLC, and RLC/MAC have been implemented in detail. System throughput was measured for various loads and channel qualities. The system saturates when the number of users increases. The maximum system throughput is 44kbps to 100kbps depending on the coding scheme. The channel quality simulations indicate that CS-2 is the preferred coding scheme under normal conditions. The packet bit rate is acceptable (6-22kbps) when the number of users is less than 30. As compared to GSM, GPRS can support at least twice as many web user with equal or even higher bit rates.},
url = {papers/Kalden_GPRS.pdf},
bibdate = {Wednesday, May 29, 2002 at 14:15:48 (CEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{nethawk,
author = {NetHawk},
title = {{NetHawk GSM Analyser}},
journal = {http://www.nethawk.fi/},
submitter = {Annika Wennstrom},
bibdate = {Wednesday, May 29, 2002 at 14:15:48 (CEST)}
}
@article{Harris02,
author = {A. Harris and R. Kravets},
title = {The Design of a Transport Protocol for On-Demand Graphical Rendering},
journal = {Proc 12th International Workshop on Network and Operating Systems Support for Digital Audio and Video (NOSSDAV 2002), Miami Beach, FL},
year = {2002},
month = {May},
annote = {This paper describes a partially reliable, partially ordered transport protocol OGP (On-Demand Graphic Rendering Protocol) tailored to one specific application, rendering of 3D models. Reports better throughput than TCP, but details about simulation setup is somewhat lacking.},
url = {papers/Harris02GraphicModPartRel.ps.gz},
bibdate = {Wednesday, May 29, 2002 at 16:58:24 (CEST)},
submitter = {Johan Garcia}
}
@article{Kravets97,
author = {R. Kravets and K. L. Calvert and P. Krishnan and K. Schwan},
title = {Adaptive Variation of Reliability},
journal = {Proc Seventh IFIP Conference on High Performance Networking (HPN'97), White Plains},
year = {1997},
month = {April},
annote = {Describes a window-based partially reliable transport protocol that uses NACKs to request retransmission of lost packet that falls outside the allowed partial reliability. Reliabilty is specified using Max consecutive losses, high loss percentage and losses per window (a la Gong&Parulkar)Several parallell streams with varying relibility can be used and each packet assigned to the stream of matching relibility. Also discuss image transfer and distributed simulation applications using the protocol. How to perform adapatation of the reliability level is also discussed. },
url = {papers/Kravets97AdativeReliability.ps.gz},
bibdate = {Wednesday, May 29, 2002 at 17:26:40 (CEST)},
submitter = {Johan Garcia}
}
@article{Denis00,
author = {Alexandre Denis},
title = {Variable Reliability Protocol:A Protocol with tunable loss tolerance for high performance over a {WAN}},
journal = {Tecnical Report 2000-11, Ecole Normale Superieure de Lyon },
year = {2000},
month = {March},
annote = {Describes the Implementation and some performance experiments for an improved version of Kravets protocol. The experiments show a large (100-800%) increase in throughput, but this seems to be due to unfair congestion behavior.},
url = {papers/Denis00VariableReliability.ps.gz},
bibdate = {Wednesday, May 29, 2002 at 17:43:21 (CEST)},
submitter = {Johan Garcia}
}
@article{Stewart02a,
author = {R. Stewart and M. Ramalho and Q. Xie and M. Tuexen and P. Conrad },
title = {SCTP Partial Reliability Extension},
journal = {Work in progress, draft-ietf-tsvwg-prsctp-00.txt},
year = {2003},
month = {June},
annote = {Describes the new partial relibility extension proposed instead of U-SCTP. The extension allows for the sender to send a Forward TSN chunk that indicates that the receiver should consider the TSNs previous to the Forward TSN as no longer being transmitted or retransmitted. The sender will not retransmit the Previous TSNs. So the retransmission decision is in effect taken on the sender side. The generation of the Forward TSN chunks is governed by a service definition, which creates flexibility. One proposed service definition is timed reliability, which basically enhances the existing lifetime functionality to not retransmit if the lifetime has expired, whereas before a lifetimed packet was always retransmitted if it had been sent in the first place. },
url = {papers/draft-stewart-tsvwg-prsctp-00.txt},
submitter = {Johan Garcia},
bibdate = {Wednesday, May 29, 2002 at 18:22:07 (CEST)}
}
@article{Cen2002,
author = {S. Cen and P.C. Cosman, and G.M. Voelker},
title = {End-to-end differentiation of congestion and wireless losses},
journal = {Proc. Multimedia Computing and Networking (MMCN2002), San Jose, CA},
year = {2002},
pages = {1-15},
month = {January},
annote = {This paper explores loss differentiation algorithms, i.e how to be able to differentiate between losss that occur due to losses on a wireless link as opposed to losses that occur because congestion in the network. They aim to examine LDA in conjunction with a video transport protocol they are working on, which basically is UDP augmented with TFRC congestion control. No retransmissions are performed, but LDA allows TFRC to not do congestion avoidance for wireless losses. The paper examines three schemes, Biaz, Spike and ZigZag. They evaluate these schemes in three different network environments, wireless last hop, wireless backbone and wireless LAN. For the wireless last hop topology, the wired common link was the bottleneck, not the wireless path. The scehemes were evaluated by ns simuylations and some hybrid schemes with heuristsics for which scheme to use in different cases. },
url = {papers/Cen2002EndtoEndLossDiff.ps.gz},
submitter = {Johan Garcia},
bibdate = {Friday, May 31, 2002 at 22:09:29 (CEST)}
}
@article{Krishnan02,
author = {Rajesh Krishnan and Mark Allman and Craig Partridge and James P.G. Sterbenz},
title = {Explicit Transport Error Notification for Error-Prone Wireless and Satellite Networks},
journal = {BBN Technical Report No. 8333, BBN Technologies},
year = {2002},
month = mar,
annote = {Discusses the issues around ETEN and describes the type oracle ETEN(ideal), backward ETEN, forward ETEN and Receiver ETEN. All but receiver-based ETEN, which was no studied, require infrastrucure changes (routers). ETEN mechanisms were simulated and compared to Reno and Westwood TCP. ETEN provide considerable increase in goodput versus RENO especially for ungongested networks, but only slightely better perfomance than Westwood. },
url = {papers/Krishnan02ETEN-revised.ps.gz},
submitter = {Johan Garcia},
bibdate = {Friday, May 31, 2002 at 23:08:14 (CEST)}
}
@article{Parsa00,
author = {C. Parsa and J.J. Garcia-Luna-Aceves},
title = {Differentiating Congestion vs. Random Loss: A Method for Improving {TCP} Performance over Wireless Links},
journal = {Proc. IEEE 2nd IEEE Wireless Communications and Networking Conference (WCNC), Chicago, IL},
year = {200},
month = {September},
annote = {Describes a variation of the authors TCP Santa Cruz scheme where wireless losses are classified differently depending on whether they were precedeed by a buildup in bottleneck queue length (congestion) or not (wireless loss). },
url = {papers/Parsa00DifferentiateLossesWirelss.ps.gz},
submitter = {Johan Garcia},
bibdate = {Friday, May 31, 2002 at 23:24:43 (CEST)}
}
@article{Chen02,
author = {Wei-Peng Chen and Yung-Ching Hsiao and Jennifer C. Hou and Ye Ge and Michael P. Fitz},
title = {Syndrome: a light-weight approach to improving {TCP} performance in mobile wireless networks},
journal = {The Journal of Wireless Communications and Mobile Computing},
year = {2002},
pages = {37-57},
annote = {The paper proposes a light-weight approach to distinguish between losses originating from congestion and bit errors. It is called syndrome, and the technique is for the base station to count the number of packets it relays, and include this counter as a TCP option. The receiver can then use the sequence number and the packet count to determine if a packet was lost on the wireless link (gap in count), or was dropped earlier before the base station received the packet (gap in sequence, count as expected). explicit loss notification is then used to inform the sender about the loss.},
url = {papers/Chen02_syndrome.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Monday, June 03, 2002 at 08:14:45 (CEST)}
}
@inproceedings{Goff00,
author = {T. Goff and J. Moronski and D. Phatak and V. Vipul Gupta},
title = {{Freeze-TCP}: A true end-to-end Enhancement Mechanism for Mobile Environments},
booktitle={Proceedings of the IEEE International Conference on Computer Communications (INFOCOM)},
year = {2000},
month = {March},
annote = {Freeze-TCP is a mechanism to improve the performance of TCP in wireless environments where handoffs are common. By exploiting the properties advertised window, a TCP connection can be "frozen", i.e. the sender keeps it congestion window, by advertising a window of size zero. This is explited when the mobile unit is about to do a handoff, to prevent packets from getting lost and regarded as congestion indication. When the handoff is complete, the receiver sends a triple acknowledgement and opens up the window. The transmission then continues as before. Experiments have been performed by using a modified version of Linux 2.1.101, and show quite promising results. [ed. note: however, I think one must take into account the number of handoffs that are probable in a real environment, in the simulation they have 100 disconnection events during a ~20 sec transmission.]. Key advantages are that it is a true end-to-end modification, and very rarely perform worse than baseline TCP.},
url = {papers/Goff00_freeze_tcp.pdf},
bibdate = {Monday, June 03, 2002 at 08:26:20 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Margaritidis01,
author = {Margaritis Margaritidis and George C. Polyzos},
title = {Adaptation techniques for ubiquitous {I}nternet multimedia},
journal = {Wireless Communications and Mobile Computing},
year = {2001},
number = {1},
annote = {Discusses the importance of adaption for the ubiquitos access to Internet multimedia content. Identified factors that influence the design and implementation are 1) location of adaptation on the end-to-end path (in source, receiver or in a proxy) 2) adaptiation policy (ie. video stream better off with long term stability, while FTP is better with the currently max throughput) 3) awareness of perceptual quality (know about different representations of a stream and the perceptual quality it provides to the user).},
url = {papers/Margaritidis02_adaptation_tech_for_ubiquitous_internet_multimedia.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Monday, June 03, 2002 at 08:38:27 (CEST)}
}
@article{Kneer00,
author = {H. Kneer and U. Zurfluh and G. Dermler and B. Stiller},
title = {A Business Model for Charging and Accounting of Internet Services},
journal = {International Conference on Electronic Commerce and Web Technologies},
year = {2000},
annote = {Introduces a business model that defines and characterizes the business entities and describes their roles and functions within and e-commerce scenario. The model has four phases; 1) Contracting phase (establish business connections, set up contracts and agreements) 2) Reservation phase (end-customer reserves resources with her access ISP, defining what QoS characteristics are needed) 4) Service phase (actual service is performed) 4) Clearing phase (contains the charging/billing and payment). The paper also discusses the foundations of IntServ and RSVP, and DiffServ. Seems to be in favour of RSVP. Finally, service license agreements are discussed.},
url = {papers/Kneer00_businessmodel.pdf},
bibdate = {Monday, June 03, 2002 at 08:51:50 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Magoni01,
author = {D. Magoni and J. J. Pansiot},
title = {Analysis of the Autonomous System Network Topology},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {3},
pages = {26--37},
month = {July},
annote = {This paper provides a detailed analysis of the interdomain topology, i.e., the AS network topology, of the Internet. The source of the analysis is a BGP router that collects routes from its 23 peers from November 1997 to May 2000. The major contribution of this paper is a rich set of observations of the topology of the current AS network, as well as four new power laws concerning the evolution of the AS network. In particular, it is observed that in the current AS network, the average distance between any two nodes is between three and four hops; the majority of ASs form a mesh, while only a small minority form a tree, i.e., the ASs become more and more interconnected. Furthermore, the analysis suggests that the AS trees left become more and more flat. 90\% of the trees is simply composed of leaves directly connected to their root. Less than 10\% of the trees has depth two, and only a few trees have depth three, which was the maximum tree depth observed. Before this paper, Faloutsos reported three power laws in a paper presented at SIGCOMM '99. In this paper, these power laws are complemented with four new power laws, namely: the pair-rank law which states that the number of distinct shortest paths between a pair of ASs is proportional to the rank of the pair to the power of a constant; the number-of-shortest-paths power law which states that the frequency of a number of distinct shortest paths between a pair of ASs is proportional to the number of shortest paths to the power of a constant; the tree-rank law which states that the size of an AS tree is proportional to the rank of the tree to the power of a constant; finally, the last power law, the tree size power law, which states that the frequency of a tree size is proportional to the tree size to the power of a constant.},
url = {papers/magoni01.ps.gz},
bibdate = {Wednesday, June 05, 2002 at 13:20:56 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Fredj01,
author = {S. B. Fredj and T. Bonald and A. Proutiere and G. Regnie and J. W. Roberts},
title = {Statistical Bandwidth Sharing: A Study of Congestion at Flow Level},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {4},
pages = {111--122},
month = {October},
annote = {This paper demonstrates that the conclusions that are drawn from simple fluid flow models are in most cases also applicable to real-world examples. The objective of this paper is to form the basis for deriving provisioning rules and traffic control algorithms. Using results from the theory of stachastic networks, it is shown that first order performance metrics like mean throughput are insensitive to flow size and flow arrival distribution as long as sessions are Poisson distributed. Furthermore, it is shown that stochastic networks are unstable when the demand exceeds the link capacity, however become stable when user impatience and reattempts are accounted for.},
url = {papers/fredj01.pdf},
bibdate = {Wednesday, June 05, 2002 at 13:33:17 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Kunniyur01,
author = {S. Kunniyur and R. Srikant},
title = {Analysis and Design of an Adaptive Virtual Queue (AVQ) Algorithm for Active Queue Management},
journal = {Computer Communication Review},
year = {2001},
volume = {31},
number = {4},
pages = {123--134},
month = {October},
annote = {With the advent of distributed multimedia applications, there has been a strong demand for service differentiation in the Internet. Something which calls for active queue management techniques. This paper presents an easy-to-implement active queue management scheme, Adaptive Virtual Queue (AVQ), which in terms of packet loss, link utilization, and responsiveness seems to outperform the most common current adaptive queue management schemes. The AVQ scheme maintains a virtual queue whose capacity is less than the actual capacity of the link. When a packet arrives at the real queue, the virtual queue is updated as well to reflect a new arrival. Packets in the real queue are marked or dropped when the virtual queue overflows. Contrary to previously proposed schemes, AVQ employs an adaptive virtual queue. The performance of AVQ compared to the well-known schemes: RED (Random Early Discard), REM (Random Early Marking), the PI controller proposed by Hollot et al., and Gibbens KVQ (GKVQ) scheme, has been studied through simulations in ns-2. The AVQ scheme has fewer packet losses than any other scheme except the GKVQ scheme. However, the link utilization of the GKVQ scheme is as low as 75%. Furthermore, the average queue length of AVQ is smaller than for any of the other schemes. },
url = {papers/kunniyur01.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, June 05, 2002 at 13:40:09 (CEST)}
}
@article{Alam01,
author = {M. Alam and R. Prasad and J. R. Farserotu},
title = {Quality of Service among IP-Based Heterogeneous Networks},
journal = {IEEE Personal Communications},
year = {2001},
volume = {8},
number = {6},
pages = {18--24},
month = {December},
annote = {The article presents a QoS architecture that comprises both wireline and wireless communication. The QoS architecture only consists of well-established and/or standardized protocol like SIP and RSVP. Consequently, the novely of the QoS architecture is not the included protocols, but how they could interoperate to provide QoS end-to-end.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Friday, June 07, 2002 at 09:50:07 (CEST)}
}
@article{Deng99,
author = {Xing Deng},
title = {Short Term Behaviour of Ping Measurements},
journal = {M. Sc. Thesis, University of Waikato},
year = {1999},
month = {July},
annote = {(Abstract): In this study ICMP (Internet Control Message Protocol) packets were sent at high speed for a short period by the ping utility to various Internet addresses and their round-trip times (RTTs) were measured by ping and the tcpdump utility. The study has two purposes: to evaluate the accuracy of ping and tcpdump in terms of measuring packet wire round-trip times (WRTTs); and to identify short term high density ping traffic patterns and interpret them. A careful comparison was done of different software and hardware configurations for measuring the WRTTs, as well as the repeatability of measurements on similar configurations. The results show that ping's own RTT measurements can contain significant source processing delay and so distort the analysis of traffic patterns; running tcpdump to capture the packet wire times increases the accuracy of WRTT measurement; running tcpdump on separate measurement machines in single-user mode with output to the RAM disks can generate consistent WRTT measurements. Findings in the pattern analysis include: short term high density ping traffic patterns can sometimes be interpreted by buffering and queueing models; the size of measurement packets can significantly influence traffic behaviour; due to source host factors ping request packets may be sent out of sequence. },
url = {papers/Deng99PingMeasurments.ps},
submitter = {Johan Garcia},
bibdate = {Saturday, June 08, 2002 at 01:26:35 (CEST)}
}
@article{tcp-parameters,
author = {The {I}nternet {A}ssigned {N}umbers {A}uthority ({IANA})},
title = {Assigned {TCP} option numbers},
journal = {http://www.iana.org/assignments/tcp-parameters},
annote = {The Transmission Control Protocol (TCP) has provision for optional header fields identified by an option kind field. This document contains the currently assigned numbers for TCP options.},
bibdate = {Tuesday, June 11, 2002 at 14:48:54 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Mathis96,
author = {M. Mathis and J. Mahdavi , August 1996, },
title = {Forward Acknowledgement: Refining TCP Congestion Control},
journal = {Proceedings of SIGCOMM'96, Stanford, CA. },
year = {1996},
month = {August},
annote = {This paper proposes an algorithm called FACK (Forward Acknowledgement), to improve TCP congestion control during recovery. The problem with TCP Reno is that it performs poorly in the case of multiple losses from the same window. The authors argue that this is because it lacks a sufficiently accurate estimate of the data outstanding in the network. The FACK algorithm uses SACK info to keep an explicit measure of the total number of bytes of data outstanding in the network. This is done by using two new state variables, snd.fack and retran_data. snd_fack keeps track of the forward-most data held by the receiver and retran_data is the quantity of outstanding retransmitted data in the network. Therefore, TCP's estimate of the amount of data outstanding in the network during recovery is: outstand_data = snd.nxt - snd.fack + retran_data Congestion is controlled by only allow transmission of data while outstan_data is less than cwnd. In simulations it is shown that FACK is less bursty than Reno +SACK, and can recover from episodes of heavy loss better than Reno+SACK },
submitter = {Katarina Asplund},
bibdate = {Sunday, June 16, 2002 at 20:07:27 (CEST)}
}
@article{Mathis99,
author = {M. Mathis and J. Semke and J. Mahdavi and K. Lahey},
title = {The Rate-Halving Algorithm for TCP congestion Control},
journal = {Draft},
year = {1999},
month = {June},
annote = {One problem with Fast Recovery is that it can add to the overall burstiness in the network. Once the cwnd has been halved (after fast retransmit), TCP waits for additional dupacks to arrive, indicating that half of the data in flight has left the network. This delay leads to that an entire window of data is transmitted in one half of one RTT. This burst of data can also cause additional bursts in successive RTTs. This Rate-halving algorithm proposed in the paper adjusts the cwnd by spacing transmissions at the rate of one data segment per two segments acknowledged over the entire recovery period, thereby sustaining the self-clocking of TCP and avoiding a burst. The algorithm can be used with both ECN, NewReno and SACK. },
bibdate = {Sunday, June 16, 2002 at 20:36:36 (CEST)},
submitter = {Katarina Asplund}
}
@inproceedings{Jiang00,
author = {W. Jiang and H. Schulzrinne},
title = {Modeling of Packet Loss and Delay and Their Effect on Real-time Multimedia Service Quality},
year = {2000},
month = {June},
annote = {The paper discusses factors affecting the perceptual quality in a VoIP application. The first factor is the modeling of network delay and loss, where the authors propose the n-state extended Gilbert model and inter-loss loss distance (ILD) to characterize loss burstiness, and the conditional CDF to capture the temporal dependency in network delays. However, the perceptual quality is ultimately determined by the final loss pattern (FLP), i.e. after playout delay control and optionally FEC. The authors found, by examining packet traces, that the FLP is still burstier than random losses and needs to be described by the extended Gilbert model. Particularly if FEC is not employed and jitter is low, late losses and network losses often merge into longer loss bursts. It is due to the observed inter-dependency between lossand delay, e.g. a loss is often preceded by high delays. },
journal = {NOSSDAV2000, Chapel Hill, North Carolina, USA},
submitter = {Katarina Asplund},
bibdate = {Sunday, June 16, 2002 at 21:16:42 (CEST)}
}
@inproceedings{Jiang00a,
author = {W. Jiang and H. Schulzrinne},
title = {Analysis of On-Off Patterns in VoIP and Their Effect on Voice Traffic Aggregation},
booktitle = {9th International Conference on Computer Communications and Networks (ICCCN)},
year = {2000},
address = {Las Vegas, Nevada, USA},
month = {October},
annote = {This paper analyzes the on-off patterns of voice over IP. In particular, the paper studies the impact of silence detectors. It is shown that although an experimental distribution of gaps and bursts in voice traffic may be adequate as a first-hand estimate, a more heavy-tailed distribution is required for more detailed analysis; Especially this is true for the gap distribution.},
url = {papers/Jiang00.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Herrin00,
author = {G. Herrin},
title = {Linux IP Networking: A guide to the Implementation and Modification of the Linux Protocol Stack},
journal = {Technical report 00-04, Department of Computer Science, University of New Hampshire},
year = {2000},
month = {May},
annote = {This report is intended as a reference for experimenters with overviews, walk-throughs, source code explanations an examples. There are chapters on, for example, initialization, connections, transmitting, receiving, routing, the proc file system, and on how to write and install Linux modules. },
submitter = {Katarina Asplund},
bibdate = {Sunday, June 16, 2002 at 21:31:05 (CEST)}
}
@article{Balan02,
author = {R. K. Balan and B. P. Lee and K. R. R. Kumar and L. Jacob and W. K. G. Seah and A. L. Ananda},
title = {{TCP HACK}: {A} mechanism to improve performance over lossy links},
journal = {Computer Networks},
year = {2002},
volume = {39},
number = {4},
pages = {347-361 },
month = {July},
annote = {A slightly extended version of the Balan01 paper. Both papers present results of a TCP modification that adds a separate header checksum to be able to differentiate congestion losses from corruption losses. They present their implementation and some results. The results show considerable throughput gain over both SACK and evenmore NewReno. Their experiments however did not include either congestion losses or backchannel losses, so their solutions performance in a more realistic setting is not known. Their loss feedback mechanism is sensitive to backchannel losses. },
url = {papers/Balan02TCP_HACK.pdf},
bibdate = {Thursday, June 20, 2002 at 20:16:45 (CEST)},
submitter = {Johan Garcia}
}
@techreport{willig-gilbert-elliot,
author = {Jean-Pierre Ebert and Andreas Willig},
title = {A {G}ilbert-{E}lliot Bit Error Model and the Efficient Use in Packet Level Simulation},
institution = {EE, TU Berlin, Germany},
year = 1999,
number = {TKN-99-002},
url = {http://www-tkn.ee.tu-berlin.de/~ebert/publication.html}
}
@article{Wang95,
author = {H. Wang and N. Moayeri},
title = {Finite-State Markov Channel: A Useful Model for Radio Communication Channels.},
journal = {{IEEE} Trans. on Veh. Tech., 44(1):163--171},
year = {1995},
month = {February}
}
@article{Williamson02,
author = {Carey Williamson and Qian Wu },
title = {A case for context-aware TCP/IP },
journal = {ACM SIGMETRICS Performance Evaluation Review},
year = {2002},
volume = {29},
number = {4},
pages = {11-23},
month = {March},
annote = {ABSTRACT: This paper discusses the design and evaluation of CATNIP, a Context-Aware Transport/Network Internet Protocol for the Web. This integrated protocol uses application-layer knowledge (i.e., Web document size) to provide explicit context information to the TCP and IP protocols. While this approach violates the traditional layered Internet protocol architecture, it enables informed decision-making, both at network endpoints and at network touters, regarding flow control, congestion control, and packet discard decisions.We evaluate the performance of the context-aware TCP/IP approach first using ns-2 network simulation, and then using WAN emulation to test a prototype implementation of CATNIP in the Linux kernel of an Apache Web server. The advantages of the CATNIP approach are particularly evident in a congested Internet with 1-10% packet loss. Simulation results indicate a 10-20% reduction in TCP packet loss using simple endpoint control mechanisms, with no adverse impact on Web page retrieval times. More importantly, using CATNIP context information at IP touters can reduce mean Web page retrieval times by 20-80%, and the standard deviation by 60-90%. The CATNIP algorithm can also interoperate with Random Early Detection (RED) for active queue management. },
url = {papers/Williamson02_Ctxt_aware_tcp.pdf},
bibdate = {Thursday, July 04, 2002 at 01:38:53 (CEST)},
submitter = {Johan Garcia}
}
@article{Gurtov02a,
author = {A. Gurtov and M. Passoja and O. Aalto and M. Raitola},
title = {Multi-Layer Protocol Tracing in a {GPRS} Network},
journal = {Proceedings of the IEEE Vehicular Technology Conference (Fall VTC2002), Vancouver, Canada},
year = {2002},
month = {September},
annote = {Abstract: This paper presents a performance evaluation of GPRS accomplished by combination of measurement at the end hosts and tracing inside the network. The multi-layer tracing approach allows not only observing, but also understanding the network performance. With end-to-end measurements we assess data rates, latency, and buffering experienced by users in a live GPRS network. Comparing the results to our previous measurements shows a notable improvement in the network and terminals over past two years. Mobility tests while driving in the urban environment quantify the interval, duration and data loss caused by cell reselections. In the test lab, multi-layer tracing of radio, link and transport protocols gives a closer picture of GPRS performance. For instance, TCP interacts inefficiently with resource allocation at the RLC layer and fragmentation at the LLC layer. Finally, we illustrate delay spikes and data losses during a cell reselection by tracing of signaling messages during a cell update and routing area update procedures. Annika: The performance of data transmission in Sonera's live and test GPRS networks is investigated in this paper. The GPRS implementation is based on release 97. The coding scheme used was CS-2. The network was configured for LLC in unacknowledged mode and RLC in acknowledged mode. VJ header compression was disabled. Throughput, latency, buffering, and mobility (cell reselections) were investigated. Both TCP and RLC data was captured (with Tcpdump and Nethawk) and analyzed. The maximum throughput was 43kbps in DL and 21kbps in UL. The round trip time in an unloaded network was 0.5-1.1s (with a typical value of 0.7s). Many interesting results are presented, and only some of them are included here. The buffers in the GPRS network are too large (50kbps per user in DL), cell reselections lead to delay spikes at the TCP level that can cause spurious timeouts. Increases in round trip times occurred due to frequent TBF allocations and releases at the RLC layer. The RLC acknowledgment scheme wasted resources in some cases. The authors also found some performance degrading bugs in the GPRS networks.},
url = {papers/Gurtov02_multilayer_TCP_GPRS.pdf},
bibdate = {Sunday, July 07, 2002 at 06:54:44 (CEST)},
submitter = {Johan Garcia}
}
@article{Stone00,
author = {Jonathan Stone and Craig Partridge},
title = {When The {CRC} and {TCP} Checksum Disagree },
journal = {Proc. ACM SIGCOMM, Stockholm, Sweden},
year = {2000},
pages = {309-319 },
month = {September},
annote = {This paper uses traces from the internet to analyze the frequencies and causes for errors that are detected by the TCP checksum. Annika: Packet errors that are not captured by the link level CRC are investigated in this paper. In the traces analyzed the Internet checksum failed in 1 packet of 1100 to 1 packet in 31900. This indicates that there must be other sources of errors than those that can be captured by the CRC check on the link level. Packets were captured with BPF (libpcap) and the Internet checksum was computed and if it failed the packet was further analyzed with an analysis software. For TCP packets the analysis software matched original segments with retransmitted. Errors were found in end-host hardware, in end-host software, in router memory, and at the link level or the network-interface hardware. Examples of the errors analyzed are: errors in IP headers caused by certain Ethernet interface cards and the most common software error was the Windows NT ACK-of-FIN bug which is now fixed. The conclusion is that the Internet checksum is useful, since many errors occur that cannot be captured on the link level. The risk of undetected checksum errors is between 1 packet in 16 million and 1 packet in 10 billion. The authors give some advice on how to reduce the error rates: hardware should not be trusted, owners of hosts and routers generating errors should be informed, and valuable data should be protected with an application level checksum.},
url = {papers/Stone00_CRC_TCPCsum_disagree.ps.gz},
bibdate = {Monday, July 08, 2002 at 20:22:13 (CEST)},
submitter = {Johan Garcia}
}
@article{Barman02,
author = {Barman, Dhiman and Matta, Ibrahim},
title = {Effectiveness of Loss Labeling in Improving {TCP} Performance in Wired/Wireless Networks},
journal = {CS Dept Technical report 2002-016, Boston University},
year = {2002},
month = {May},
annote = {Presents the NewReno-FF scheme which uses applies two RTT estimation filters at the sender to be able to differentiate loss causes. The underlying assumption is that for packets suffering congestion losses, the observed RTT vill vary whereas for random packet loss the RTT will not vary much. },
url = {papers/Barman02_loss_labeling_TCP.ps.gz},
submitter = {Johan Garcia},
bibdate = {Tuesday, July 09, 2002 at 20:14:01 (CEST)}
}
@article{Liu02,
author = {Liu, Jun and Matta, Ibrahim and Crovella, Mark},
title = {End-to-End Inference of Loss Nature in a Hybrid Wired/Wireless Environment},
journal = {CS Dept Technical report 2002-008, Boston University},
year = {2002},
month = {March},
annote = {In this paper they use packet loss pairs and hidden markov modeling to infer a network state based on the most recent RTT at the time of a loss. The paper evaluates a 4-state HMM trained with 10000 loss pairs to obtain RTT distributions for the states. The states are then mapped to loss causes according to some heuristics. },
url = {papers/Liu02_EndToEnd_Loss_inference.ps.gz},
submitter = {Johan Garcia},
bibdate = {Tuesday, July 09, 2002 at 23:19:06 (CEST)}
}
@article{Kim99a,
author = {T. Kim and S. Lu and V. Bharghavan},
title = {Improving Congestion Control Performance Through Loss Differentiation},
journal = {Proceedings International Conference on Computers and Communications Networks (ICCCN99), Boston, USA},
year = {1999},
month = {October},
annote = {This paper suggest the use of Linear Increase/Multiplicative Decrease with History (LIMD/H). Effective transmission rate history and congestion throttling history is maintained. Depending on the current rates relation to the effective rate, losses are classified into one of three categories: Congestion loss, probe loss or non-congestion loss. },
url = {papers/Kim99a_LIMDH_LossDiff.ps.gz},
submitter = {Johan Garcia},
bibdate = {Wednesday, July 10, 2002 at 01:52:37 (CEST)}
}
@article{Wang01a,
author = {Shie Yuan Wang and H. T. Kung},
title = {Use of {TCP} Decoupling in Improving {TCP} Performance over Wireless Networks},
journal = {Wireless Networks},
year = {2001},
volume = {7},
number = {3},
pages = {221-236},
annote = {Abstract: We propose using the TCP decoupling approach to improve a TCP connection's goodput over wireless networks. The performance improvement can be analytically shown to be proportional to $\sqrt{\mathrm{MTU}/\mathrm{HP}\_\mathrm{Sz}}$, where MTU is the maximum transmission unit of participating wireless links and HP_Sz is the size of a packet containing only a TCP/IP header. For example, on a WaveLAN [32] wireless network, where MTU is 1500 bytes and HP_Sz is 40 bytes, the achieved goodput improvement is about 350%. We present experimental results demonstrating that TCP decoupling outperforms TCP reno and TCP SACK. These results confirm the analysis of the performance improvement.},
url = {papers/Wang01a_TCP_Decoupling_wireless.pdf},
bibdate = {Wednesday, July 10, 2002 at 03:07:47 (CEST)},
submitter = {Johan Garcia}
}
@article{Heinzelman00,
author = {Wendi Heinzelman},
title = {Application-Specific Protocol Architectures for Wireless Networks},
journal = {PhD Thesis, Massachusetts Institute of Technology,},
year = {2000},
month = {June},
annote = {Abstract: In recent years, advances in energy-efficient design and wireless technologies have enabled exciting new applications for wireless devices. These applications span a wide range, including real-time and streaming video and audio delivery, remote monitoring using networked microsensors, personal medical monitoring, and home networking of everyday appliances. While these applications require high performance from the network, they suffer from resource constraints that do not appear in more traditional wired computing environments. In particular, wireless spectrum is scarce, often limiting the bandwidth available to applications and making the channel error-prone, and the nodes are battery-operated, often limiting available energy. My thesis is that this harsh environment with severe resource constraints requires an application-specific protocol architecture, rather than the traditional layered approach, to obtain the best possible performance. This dissertation supports this claim using detailed case studies on microsensor networks and wireless video delivery. The first study develops LEACH (Low-Energy Adaptive Clustering Hierarchy), an architecture for remote microsensor networks that combines the ideas of energy-efficient cluster-based routing and media access together with application-specific data aggregation to achieve good performance in terms of system lifetime, latency, and application-perceived quality. This approach improves system lifetime by an order of magnitude compared to general-purpose approaches when the node energy is limited. The second study develops an unequal error protection scheme for MPEG-4 compressed video delivery that adapts the level of protection applied to portions of a packet to the degree of importance of the corresponding bits. This approach obtains better application-perceived performance than current approaches for the same amount of transmission bandwidth. These two systems show that application-specific protocol architectures achieve the energy and latency efficiency and error robustness needed for wireless networks. },
url = {papers/Heinzelman00_AppSpecificProtoArch.ps.gz},
submitter = {Johan Garcia},
bibdate = {Thursday, July 11, 2002 at 01:56:43 (CEST)}
}
@phdthesis{Chengpeng01,
author = {Fu Chengpeng},
title = {{TCP Veno}: End-To-End Congestion Control Over Heterogeneous Networks},
school = {Chinese University of Hong Kong},
year = {2001},
month = {July},
annote = {Describes a TCP variant that uses Vegas like estimation techniques to classify the loss causes and apply a more conservative congestion approach. Has been implemented and tested over real networks and provided gains.},
url = {papers/Chengpeng01_TCPVeno.pdf},
submitter = {Johan Garcia},
bibdate = {Thursday, July 11, 2002 at 05:43:18 (CEST)}
}
@article{Samaraweera99,
author = {N. K. G. Samaraweera},
title = {Non-congestion packet loss detection for {TCP} error recovery using wireless links},
journal = {IEE Proc. Comm.},
year = {1999},
volume = {146},
number = {4},
pages = {222-230},
annote = {Utdrag (papper ej tillgängligt):On detection of a packet loss, the transmitter compares the currently measured round trip delay with a calculated delay threshold (which is equivalent to the delay at the knee point). If it is less than the threshold value, it concludes that the network has been under-utilised. The transmitter, therefore, assumes any packet loss as a non-congestion loss and retransmits the corresponding packet selectively using fast retransmission, but does not involve the congestion avoidance measure (which would have resulted in a decreased window size). Otherwise, the trans-mitter considers it may be a congestion loss and reduces the transmission rate. },
bibdate = {Friday, July 12, 2002 at 05:15:43 (CEST)},
submitter = {Johan Garcia}
}
@inproceedings{Casetti01,
author = {C. Casetti and M. Gerla and S. Mascolo and M. Y. Sanadidi and R. Wang},
title = {{TCP W}estwood: Bandwidth Estimation for Enhanced Transport over Wireless Links},
booktitle = {Proceedings of ACM Mobicom},
year = {2001},
ADDRESS = {Rome, Italy},
pages = {287-297},
month = {July},
annote = {The first paper on TCP Westwood.},
url = {papers/Casetti02_Westwood.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, July 16, 2002 at 09:50:04 (CEST)}
}
@article{Inamura02,
author = {H. Inamura and G. Montenegro and R. Ludwig and A. Gurtov and F. Khafizov},
title = {{TCP over Second (2.5G) and Third (3G) Generation Wireless Networks}},
journal = {PILC Internet draft},
year = {2002},
month = {July},
annote = {This Internet draft from the PILC group gives recommendations on how to optimize TCP over 2.5G and 3G networks. The characteristics of 2.5G and 3G networks are described. The recommendations given conform with IETF standards. For example, both SACK and the timestamp option are recommended optimizations. The timestamp option is recommended because it leads to a more conservative retransmission timer in 2.5G and 3G networks. Header compression should be disabled over the wireless link, since current algorithms do not compress all types of segments (SYNs and FINs) and cannot handle the TCP option fields properly. The draft is concluded with a discussion on open issues such as the initial value of the RTO (3s may be too short) and active queue management.},
url = {http://www.ietf.org/internet-drafts/draft-ietf-pilc-2.5g3g-10.txt},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Wednesday, May 29, 2002 at 14:15:48 (CEST)}
}
@misc{Inamura03,
author = {H. Inamura and G. Montenegro and R. Ludwig and A. Gurtov and F. Khafizov },
title = "{RFC} 3481: {TCP} over Second {(2.5G)} and Third {(3G)} Generation Wireless Networks",
month = {February},
year = {2003},
bibdate = {Monday, April 07, 2003 at 16:12:13 (CEST)},
url = {papers/rfc3481.txt},
submitter = {Annika Wennstr\"{o}m}
}
@article{Ghribi00,
author = {B. Ghribi and L. Logrippo},
title = {Understanding {GPRS}: the {GSM} packet radio service},
journal = {Computer Networks (Amsterdam, Netherlands: 1999)},
year = {2000},
volume = {34},
number = {5},
pages = {763--779},
annote = {The ETSI specifications of the GPRS service description (02.60 and 03.60) are summarized in this paper. The emphasis is on services and architectural aspects rather than on protocols. GPRS terminology is introduced and explained, but no details are given. The paper describes GPRS services, the GPRS architecture, security aspects, mobility management, end-to-end routing, and the GSM/GPRS interaction. The paper is concluded with a discussion about the limitations of GPRS and the evolution to 3G.},
url = {papers/Ghribi_Understanding_GPRS.pdf},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Aug 1 14:42:34 CEST 2002}
}
@article{Ajib01,
author = {Wessam Ajib and Philippe Godlewski},
title = {Acknowledgment Procedures at Radio Link Control Level in {GPRS}},
journal = {Wireless Networks},
year = {2001},
volume = {7},
number = {3},
pages = {237-247},
annote = {The authors propose a new hybrid FEC/ARQ mechanism for GPRS. The idea is to reduce the number of Ack/NAck blocks required at the RLC level. The FEC part works as follows: Information blocks are transmitted first, and in case of errors redundancy blocks are also transmitted. If FEC fails to recover data, then RLC falls back to use its standard Ack/NAck procedures. The proposed mechanism rely on feedback information, but the number of explicit Ack/NAck blocks are reduced, since the feedback is transmitted in unused header fields of control blocks that are used for other signaling purposes (e.g. count down, timing advance). The proposed FEC/ARQ mechanism was evaluated by simulations. The channel was modeled as a two state Gilbert model. The proposed mechanism performed better than standard RLC procedures. The gain was highest for CS-1 and for a channel quality varying over time.},
url = {papers/ajib99acknowledgement.ps.gz},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thu Aug 1 14:42:34 CEST 2002}
}
@article{Stuckmann01,
author = {P. Stuckmann and F. M\"{u}ller},
title = {{Quality of Service Management in GPRS Networks}},
journal = {{Networking - ICN 2001}},
year = {2001},
month = {July},
annote = {The performance gain of GPRS QoS is estimated in comparison to best-effort service. An overview of GPRS QoS is given, and the GPRSim simulator is described. Three service classes are considered: premium, standard, and best-effort. There is a limit on the resources that can be reserved by a class. If the limit is reached the required QoS is not granted. The BSS has one queue per service class, and RLC/MAC blocks are scheduled according to service. The block error probability is 13.5\% which corresponds to a C/I of 12dB. Both LLC and RLC operate in acknowledged mode, and CS-2 is used in all simulations. The result is that the premium service class users can be served with nearly constant throughput and delay also with 40 users in the system.},
url = {papers/Stuckmann01_QoS_GPRS.pdf},
bibdate = {Thu Aug 1 14:42:34 CEST 2002},
submitter = {Annika Wennstr\"{o}m}
}
@article{Chockalingam99,
author = {A. Chockalingam and Michele Zorzi},
title = {{Wireless TCP Performance with Link Layer FEC/ARQ}},
journal = {Proc. IEEE ICC'99},
year = {1999},
month = {June},
annote = {The simulation study presented evaluates the performance of TCP Tahoe and NewReno on wireless links with a FEC/ARQ protocol at the link layer. The parameters investigated were Doppler frequency, TCP segment size, link layer frame size, number of link layer retransmission attempts, and FEC code rate. The ARQ protocol is NAK-based, and after a number of retransmission attempts the link layer gives up and leaves error recovery to TCP. The TCP throughput degrades when the number of error events increase. This implies that few longer error bursts (low Doppler frequencies) are better for TCP performance than shorter errors that occur more often. The TCP throughput is higher with link layer ARQ than without, provided that the link layer recovers data, and the number of error events at the TCP level is kept low. Link layer ARQ, with an appropriate number of retransmission attempts, improve TCP throughput more than FEC. The results were about the same for TCP Tahoe and NewReno.},
url = {papers/chockalingam99wireless.ps.gz},
bibdate = {Thu Aug 1 14:42:34 CEST 2002},
submitter = {Annika Wennstr\"{o}m}
}
@article{Gurtov02b,
author = {A. Gurtov and R. Ludwig},
title = {{Making TCP Robust Against Delay Spikes}},
journal = {INTERNET-DRAFT expires in August 2002, work in progress},
year = {2002},
month = {February},
annote = {Delay spikes occur in wireless WANs due to handovers, preemption, and radio coverage holes. TCP performance is unnecessarily reduced because delay spikes trigger retransmission of data that is not lost but only delayed. The performance of TCP may be enhanced if the retransmission timer is restarted on the arrival on dupAcks. This avoids timeouts and increases the chance of receiving three dupAcks to trigger fast retransmit. The timer should also be restarted when a fast retransmit is transmitted, and dupAcks should be ignored after a timeout. The authors state that the recommendations are permitted by RFC 2581 and RFC 2988.},
url = {papers/draft-gurtov-tsvwg-tcp-delay-spikes-00.txt},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Tue Aug 6 17:03:59 CEST 2002}
}
@inproceedings{Kojo01,
author = {M. Kojo and A. Gurtov and J. Manner and P. Sarolahti and T. Alanko and K. Raatikainen},
title = {Seawind: {A} Wireless Network Emulator},
booktitle = {Proceedings of 11th GI/ITG Conference on Measuring, Modelling and Evaluation of Computer and Communication Systems},
year = {2001},
month = {September},
annote = {The Seawind emulator was developed for evaluating protocols and applications in GPRS. The emulator is however general enough to emulate many networks. The advantage of Seawind is that it emulates characteristics that are typical for wireless networks such as timed events (eg. delay due to resource allocation) and varying link quality. This is not supported by NISTNet and DummyNet. The network model seems rather detailed and it is possible to control many parameters. Seawind also supports post processing of logs. A case study illustrates what can be achieved with Seawind.},
url = {papers/Kojo01_Seawind.pdf},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Tue Aug 6 17:03:59 CEST 2002}
}
@article{Padhye01a,
author = {Jitendra Padhye and Sally Floyd},
title = {On Inferring {TCP} Behavior},
journal = {ACM SIGCOMM},
year = {2001},
pages = {287--298},
annote = {The TCP behavior of major web servers was investigated by a tool that the authors have developed. The tool is called TCP Behavior Inference Tool (TBIT). The source code is downloadable from http://www.icir.org/tbit/. The TBIT architecture is described and the TCP behavior of 4550 remote web servers is presented. The tests performed were initial value of congestion window, congestion control algorithm, conformant congestion control (to check if cwnd is halved after congestion), response to SACK, time wait duration (to check MSL, which is specified to two minutes) and response to ECN. The NMAP tool was used to determine the operating system of the web servers. Each test methodology and its results are described in detail. Most web servers investigated used the NewReno congestion control algorithm. The most commonly used value of the initial value of the congestion window was 2 segments, and most web servers did halve the congestion window. It is interesting to note that more than 25\% of the web servers did not use Fast retransmit. Less than half of the web servers were SACK-enabled. The MSL used by most servers was between 30 seconds and 2 minutes. More than 20 000 web servers were tested for ECN capability. ECN was not widely deployed. Some web servers even denied access to ECN-capable clients. New TBIT tests are under development, and the possibility to generate models for simulators is explored.},
url = {papers/Padhye01a_TCP_behavior.ps.gz},
bibdate = {Wed Aug 14 14:20:52 CEST 2002},
submitter = {Annika Wennstr\"{o}m}
}
@article{Liu02a,
author = {M. Liu and N. Ehsan},
title = {Modeling {TCP} Performance with Proxies},
journal = {International Workshop on Wired/Wireless Internet Communications (WWIC'02), in Proc. International Conference on Internet Computing (IC'02), Las Vegas},
year = {2002},
month = {June},
annote = {In this paper an analytical model of a general TCP proxy is presented. The proxy types that are modeled shorten the TCP feedback loop by slitting or spoofing the end-to-end connection. Examples of proxies that fall into this category are split TCP, TCP spoofing, indirect TCP, and web caches in proxy mode (in case of miss). The mathematical model of lossless links is validated against an Ns2 simulation. For lossy links the model is limited to steady state behavior. The authors plan further development of the lossy link model. The overall result is that the use of TCP proxies gives higher link utilization and lower latency. Performance gain is, however, limited when the proxy becomes congested. In heterogeneous environments, such as wired/wireless networks, the slower link dominates the performance. Performance is maximized if the links have similar properties. This implies that it is important to minimize asymmetry of properties between links, as performance might not improve if only one of the links is made faster. The proxy must of course have sufficient capacity in order not to become congested and drop packets. My interpretation of the results presented in the paper is that proxies improve performance as long as they do not drop packets. This is not explicitly stated in the paper, but different causes to packet loss in the proxy is discussed instead, such as slow or congested proxy and queue build up between links of different speeds.},
url = {papers/Liu02a_modeling_TCP_proxy.ps.gz},
bibdate = {Wed Aug 14 14:20:52 CEST 2002},
submitter = {Annika Wennstr\"{o}m}
}
@article{Wen99,
author = {Jiangtao Wen and John D. Villasenor},
title = {Utilizing Soft Information in Decoding of Variable Length Codes},
journal = {IEEE Data Compression Conference},
year = {1999},
pages = {9},
annote = {Presents a method for utilizing soft information in decoding of variable length codes (VLC). In most applications of VLCs, decoding is performed bit by bit, with the input to the entropy decoder assumed to be a sequence of "hard" bits about which no soft information is available. However, in noisy environments, soft information can be associated with each information bit, either by direct use of channel observations in the case of uncoded transmission, or through soft-output channel decoders when channel coding is used. The paper introduce a soft-in soft-out (SISO) dynamic decoding algorithm for decoding of VLCs. The SISO VLC decoder proposed involves no modification to the encoder, it recieves as input a packet of known length containing VLC data that has been corrupted by Gaussian noise, and produces the codeword sequence which is most likely to have been input to the VLC encoder at the transmitter.},
url = {wen99utilizing.pdf},
submitter = {Hannes Persson},
bibdate = {Friday, August 16, 2002 at 14:32:18 (CEST)}
}
@article{Heier02,
author = {S. Heier, J.-O. Rock, A. Kemper},
title = {Performance Characteristics of {UMTS} for the Mobile Internet Access},
journal = {International Conference on Wireless Networks},
year = {2002},
month = {June},
annote = {The paper describes simulations of TCP performance over UMTS. The simulator was developed using SDL and C++. SDL diagrams illustrating the simulator are presented in the paper. The UMTS radio interface, the radio link control protocol (RLC), and Internet protocols are modeled. Traffic generators for CBR and VBR (web traffic) are also included in the simulator environment. Packet delay and throughput were measured for CBR and VBR over TCP and RLC in acknowledged and unacknowledged mode. As expected and shown by others before, RLC error recovery is more efficient than using only TCP error recovery. RLC in unacknowledged mode is not useful for TCP traffic. VBR and RLC in acknowledged mode give acceptable performance to mobile users, but the performance is not as good as in fixed networks, due to the weak performance of TCP in mobile environments. Hence, higher capacity in the wireless network is not a sufficient solution.},
url = {papers/Heier02_UMTS.pdf},
bibdate = {Sun Aug 18 22:25:58 CEST 2002},
submitter = {Annika Wennstr\"{o}m}
}
@article{Kuhne99,
author = {Gerald Kühne, Christoph Kuhmünch},
title = {Transmitting MPEG-4 Video Streams over the Internet: Problems and Solutions},
journal = {Seventh ACM International Multimedia Conference 1999 Orlando, Florida},
year = {1999},
pages = {8},
month = {November},
annote = {Discuss the transmission of MPEG-4 video streams over lossy packet networks. Due to the efficient compression achieved by the MPEG-4 standard, parts of the bitstream show high inter-dependencies. Consequently, without any precautions, packet loss within the transmission of a video session severely affects the video quality. Based on an error resilient tool video packet (VP), already provided by the MPEG-4 standard, they define an RTP payload format for MPEG-4 video streams and appropriate fragmentation rules. The experimental results show that VPs restricts the influence of errors due to packet loss.},
submitter = {Hannes Persson},
bibdate = {Monday, August 19, 2002 at 08:14:22 (CEST)}
}
@inproceedings{Jacobson88,
author = {Van Jacobson and M. J. Karels},
title = {Congestion Avoidance and Control},
booktitle = {{Proceedings of SIGCOMM '88 Workshop}},
year = {1988},
pages = {314--329},
annote = {this is the historic paper with the reasoning behind the congestion control techniques used today. To remedy the 1986 congestion collapses, seven new algorithms were put into 4BSD TCP; round-trip-time variance estimation, exponential retransmit timer backoff, slow-start, more aggressive receiver ack policy, dynamic window sizing on congestion, karns clamped retransmit backoff and fast retransmit. The cause of the collapses was that TCP flows did not follow the "conservation of packets" principle, i.e. a new packet should not be put into the network until an old packet leaves ("in equilibrium"). The paper further discusses on how to get to the equilibrium (slow start), staying there (round-trip estimator) and how to adapt to changes (congestion avoidance).},
url = {papers/jacobson88congestion.pdf},
bibdate = {Monday, August 19, 2002 at 08:16:48 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Handley01b,
author = {Mark Handley and Vern Paxson},
title = {Network Intrusion Detection: Evasion, Traffic Normalization, and End-to-End Protocol Semantics},
journal = {Proceedings of the {USENIX} Security Symposium},
year = {2001},
annote = {The paper presents the problem with network intrusion detection systems, that it must keep the same state as the monitored receivers. An example of an attack is to use varying TTL's to make some packets reach the destination, while others dont. The NIDS then has difficulties of determining which packets are received by the destination host, and therefore has less chance of discovering an attack. The paper presents a technique known as traffic normalizing, i.e. packet header fields are normalized to safer values. All TTL's could be set to the maximum hop count in the internal network for example. For each of the IP header fields there is a discussion on potential misuse, how to normalize and the effects the change has on semantics. Also includes (without comments) an overview of UDP, TCP and ICMP normalizations.},
url = {papers/handley01_norm-usenix-sec-01.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Monday, August 19, 2002 at 08:27:40 (CEST)}
}
@article{Ghinea99,
author = {G. Ghinea, J.P. Thomas, R.S. Fish},
title = {Multimedia, Network Protocols and Users - Bridging the Gap},
journal = {Seventh ACM International Multimedia Conference 1999 Orlando, Florida},
year = {1999},
pages = {9},
month = {November},
annote = {Present the case for using specifically configured protocol stacks tweaked towards human requirements in the delivery of distributed multimedia. They define Quality of Perception (QoP) as representing the user side of the more technical and traditional QoS. QoP is a term encompassing not only user's satisfaction with the quality of multimedia presentations, but also his/her ability to analyse, synthesis and assimialte the informational content of multimedia. The Dynamically Reconfigurable Protocol Stacks (DRoPS) architecture supports low cost reconfiguration of individual protocol mechanisms in an attempt to best maintain QoP in connections where the provided QoS fluctuates unpredictably. Results show that DRoPS can be used to improve on the QoP (compared to TCP and UDP stacks), especially in the case of dynamic and complex sequences.},
bibdate = {Monday, August 19, 2002 at 08:29:56 (CEST)},
submitter = {Hannes Persson}
}
@article{Amit02,
author = {Amit K. Jain and Sally Floyd},
title = {Quick-Start for {TCP} and {IP}},
journal = {draft-amit-quick-start-00.txt, IETF},
year = {2002},
annote = {This draft outlines an optional Quick-Start mechanism for transport protocols to determine an optional allowed initial congestion window or initial sending rate at the start of a data transfer. By using Quick-Start, a TCP host, say, host A, would indicate its desired initial sending rate in packets per second in a Quick Start Request option in the IP header of the initial TCP SYN or SYN/ACK packet. Each router in turn could either approve the specified initial rate, reduce the specified initial rate, or indicate that nothing above the default initial rate for that protocol would be allowed. The Quick-Start mechanism also can determine if there are routers along the path that do not understand the Quick Start Request option, or have not agreed to the initial rate described in the option. If all of the routers along the path have agreed to the initial rate in the Quick-Start Request, then TCP host B communicates this to TCP host A in a transport-level Quick-Start Response in the answering SYN/ACK or ACK packet. Quick-Start is designed to allow TCP connections to use high initial windows in circumstances when there is significant unused bandwidth along the path, and all of the routers along the path support the Quick-Start Request. },
url = {papers/draft-amit-quick-start-00.txt},
bibdate = {Monday, August 19, 2002 at 08:36:17 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Ma00,
author = {Jian Ma and Jussi Ruutu and Jing Wu},
title = {An enhanced {TCP} mechanism -- {Fast-TCP} in {IP} networks with wireless links},
journal = {Wireless Networks},
year = {2000},
volume = {6},
number = {5},
pages = {375-379},
month = {November},
annote = {This paper presents the result of a study of using Fast-TCP over wireless links with transmission errors. The basic operation of Fast-TCP is for the router/gateway to delay acknowledgements when congestion is about to occur. This causes the sender to slow down its transmission, and ease congestion. It "has been proved that Fast-TCP can speed up TCP flow control time, reduce buffer ocillation, increase bandwidth utilization, increase throughput and reduce packet losses" in IP networks with wireless links, and now they wanted to see how it behaved in a wireless environment. The OPNET radio modeler was used to simulate a wireless link, and from the simulations it was concluded that Fast-TCP improves the performance of TCP over wireless (as well as wired then). Graphs show the buffer occupacy in the router being overloaded when regular TCP is used, but stays below limit with Fast-TCP. Although more buffer space is spent on buffering delayed acknowledgements, the total buffer load is decreased since acks are mostly empty packets. However the simulations seem quite weak, seemingly only doing one experiment with a BER of $10^{-8}$.},
bibdate = {Monday, August 19, 2002 at 08:56:27 (CEST)},
submitter = {Stefan Alfredsson}
}
@article{Hagenauer99,
author = {Joachim Hagenauer, Thomas Stockhammer},
title = {Channel Coding and Transmission Aspects for Wireless Multimedia},
journal = {IEEE, Journal on Special Areas in Communications (JSAC) Special Issue Video Transmission for Mobile Multimedia Applications },
year = {1999},
pages = {1764?1777},
annote = {Multimedia transmission has to handle a variety of compressed and uncompressed source signals such as data, text, image, audio, and video. On wireless channels the error rates are high and joint source/channel coding and decoding methods are advantageous. Also the system architecture has to adapt to the bad channel conditions. Several examples of a joint design are given. Especially we advocate the use of rate-compatible punctured systematic recursive convolutional (RCPRSC) codes which are shown to lead to a straightforward and versatile unequal error protection (UEP) design. In addition, the high-end receiver could use soft-outputs and source-controlled channel decoding for even better performance.},
url = {papers/hagenauer99.ps},
submitter = {Hannes Persson},
bibdate = {Wednesday, September 04, 2002 at 15:09:56 (CEST)}
}
@article{Su01a,
author = {XIAO SU},
title = {ERROR CONCEALMENT FOR ROBUST IMAGE AND VIDEO TRANSMISSIONS ON THE INTERNET},
journal = {University of Illinois at Urbana-Champaign},
year = {2001},
annote = {THESIS: Degree of Doctor of Philosophy in Computer Science},
url = {papers/TP15.pdf},
submitter = {Hannes Persson},
bibdate = {Wednesday, September 04, 2002 at 15:26:03 (CEST)}
}
@article{Taubman01,
author = {Taubman, David; Ordentlich, Erik; Weinberger, Marcelo; Seroussi, Gadiel},
title = {Embedded Block Coding in JPEG2000},
journal = {HP Labs Technical Reports},
year = {2001},
pages = {36},
month = {February},
annote = {Abstract and full paper (ps/pdf) available at http://www.hpl.hp.com/techreports/2001/HPL-2001-35.html},
bibdate = {Monday, September 16, 2002 at 11:21:56 (CEST)},
submitter = {Hannes Persson}
}
@article{Lemmon02,
author = {John J. Lemmon},
title = {Wireless Link Statistical Bit Error Model},
journal = {{US National Telecommunications and Information Administration (NTIA) Report 02-394}},
year = {2002},
month = {June},
annote = {Abstract: A bit error model that enables simulations of the digital error performance of wireless communication links has been developed. The model development has been based on error sequences derived from waveform simulations of wireless link performance with various modems operating under varying propagation, noise, and interference conditions. Values of the model parameters are obtained by analyzing the distributions of the lengths of error bursts and error gaps (error-free intervals). Mathematical expressions have been derived for the means and variances of the error burst and error gap distributions of the model as functions of the model parameters. Constraining the means and variances to the values obtained from waveform simulations uniquely determines values of the model parameters corresponding to a given set of link conditions. Examples of error burst and error gap distributions obtained from waveform simulations are compared with those generated by the model for a land mobile radio system and a wireless local area network. The simulated and model distributions are quite similar; however, the model runs tens of thousands of times faster than the corresponding waveform simulations, enabling rapid determination of link performance. Article origin, sept. 2002: http://www.its.bldrdoc.gov/pub/ntia-rpt/02-394/index.html},
url = {papers/Lemmon02_statistical_bit_error_model.pdf},
bibdate = {Wednesday, September 18, 2002 at 10:10:42 (CEST)},
submitter = {Stefan Alfredsson}
}
@techreport{Chou01,
author = {P. A. Chou and Z. Miao},
title = {Rate-Distortion Optimized Streaming of Packetized Media},
institution = {Microsoft Research},
year = {2001},
number = {MSR-TR-2001-35},
month = {feb},
annote = {This paper addresses the problem of streaming packetized media over a lossy network in a cost-and-quality optimized way. It is shown that an optimal algorithm in terms of cost and quality is one that minimizes the Lagrangian expression D + C*R, where D is an estimate of the quality distortion, C is a constant, and R is an estimate of the transmission cost. The paper also presents how the finding in the paper can be applied to a window-based transport protocol like TCP. Notably, the findings in this paper enables streaming with a transport protocol that uses an AIMD-based congestion control scheme.},
submitter = {Karl-Johan Grinnemo}
}
@article{Loguinov02,
author = {D. Loguinov and H. Radha},
title = {End-to-End Internet Video Traffic Dynamics: Statistical Study and Analysis},
journal = {Proc. IEEE Infocom 2002, New York},
year = {2002},
month = {July},
annote = {Describes the results from large scale experiments using modem dial-up access points in 5more than 600 US cities. MPEG4 videos were transferred using their own UDP NACK-based protocol and the trasfers captured. The packet loss ration, RTT and packet reordering were analyzed. The streams were low-bitrate (16 & 27.4 kbps) so the results may be different from general Internet use. The packet loss rate was ca 0.5%. The average loss burst length were 2 packets, and by communication with operators the authors concluded that most ISP actively disable RED. The average RTT was 700-800 ms and the minimum around 150 ms. Physical layer retransmisisons and buffering were given as explanantions. The reordering was 0.2% of all sent packets, and 85% of all reordering events were reordered by 1 packet. In summary a good paper with useful info.},
url = {papers/Loguinov02E2EVideoDynamics.pdf },
submitter = {Johan Garcia},
bibdate = {Thursday, October 17, 2002 at 15:59:35 (CEST)}
}
@article{Zhang02,
author = {M. Zhang and B. Karp and S. Floyd and L. Peterson},
title = {Improving TCP's Performance under Reordering with DSACK },
journal = {submitted to SIGCOMM 2002},
year = {2002},
month = {February},
annote = {Today TCP reacts to significant reordering by sending false fast retransmits, something which severly degrades the throughput. This paper proposes a number of enhancements that improves the protocol's robustness to reordering. Firstly, the sender uses DSACK information to detect and recover from false fast retransmits. Secondly, dupthresh is varied dynamically in order to avoid false fast retransmits proactively. Simulations show that the proposed enhancements consistently improves TCP's throughput significantly in the face of reordering, as compared to standard SACK TCP.},
bibdate = {Wednesday, October 23, 2002 at 14:34:40 (CEST)},
submitter = {Katarina Asplund}
}
@article{Mohammed02,
author = {A. Mohammed and E. Jones and H. Ogier},
title = {DiffServ Experiments: Analysis of the Premium Service over the Alcatel-NCSU Internet2 testbed},
journal = {ECUMN'2002, Colmar, France},
year = {2002},
month = {April},
annote = {This paper presents results from a DiffServ field trial over a large scale testbed. Results show that DiffServ can deliver the premium service (expedited forwarding, EF) regarding bandwidth (e.g. guarantee a minimum bandwidth), but that it could not offer the same service regarding jitter and delay, at least not without help from other protocols and mechanisms (such as bandwidth brokers and MPLS).},
submitter = {Katarina Asplund},
bibdate = {Wednesday, October 23, 2002 at 14:53:08 (CEST)}
}
@article{Moors02,
author = {T. Moors},
title = {A critical review of "End-to-End arguments in system design"},
journal = {ICC 2002. IEEE International Conference on Communications},
year = {2002},
volume = {2},
pages = {1214-1219},
month = {October},
annote = {This paper reviews the end-to-end arguments, highlighting their subleties, and provides additional arguments for and against end-to-end implementations. It discusses, for example, how to determine if the the end-to-end arguments are applicable to a certain service. The author means that it is important to consider what entity is responsible for ensuring the service, and the extent to which that entity can trust other entities to maintain that service. For example, as congestion is a phenomenon of the network, it is the network that is responsible for isolating endpoints that offer excessive traffic so the network can provide its service to other endpoints. Furthermore, as the network has no reason to trust the endpoints, congestion control should be implemented in the network (and not in transport protocols)},
submitter = {Katarina Asplund},
bibdate = {Wednesday, October 23, 2002 at 15:23:07 (CEST)}
}
@article{Floyd01a,
author = {S. Floyd and R. Gummadi and S. Shenker},
title = {Adaptive RED: An Algorithm for increasing the Robustness of RED's Active Queue Management},
journal = {submitted for publication},
year = {2001},
month = {August},
annote = {One of RED's weaknesses is that the average queueing delay is sensitive to the traffic load and to parameter settings, and is therefore not predictable in advance. Consequently, as network operators cannot estimate average delay a priori, they cannot offer delay as a QoS parameter to their customers. This paper proposes some changes to the RED algorithm, which removes the sensitivity to parameters that affect RED's performance. Simulations show that this Adaptive RED can maintain a predictable average queue size in a wide variety of traffic scenarios.},
bibdate = {Thursday, October 24, 2002 at 10:15:20 (CEST)},
submitter = {Katarina Asplund}
}
@article{Kohler02a,
author = {E. Kohler and M. Handley and S. Floyd and J. Padhye},
title = {Datagram Congestion Control Protocol (DCCP)},
journal = {Internet Draft, draft-kohler-dcp-04.ps, expires december 2002},
year = {2002},
month = {June},
annote = {This document is a new draft of (Kohler02). Changes since that draft (02):Changed name of protocol; renamed options; new state diagram; new Loss Window and Connection Nonce options/features address sequence number validity; Slow Receiver option; DCCP checksum includes a pseudoheader; Move packet format changed; added explanation of acks-of-acks. },
bibdate = {Thursday, October 24, 2002 at 10:32:51 (CEST)},
submitter = {Katarina Asplund}
}
@article{Allman00,
author = {M. Allman},
title = {A Web Server's View of the Transport Layer},
journal = {ACM Computer Communication Review},
year = {2000},
volume = {30},
number = {5},
month = {October},
annote = {This paper presents observations of traffic to and from a particular WWW server over the course a year and a half. Among others, the paper considers the deployment of various TCP and HTTP options, the range of RTTs observed, packet sizes, and the impact of using larger initial congestion window. Some of the key results are that SACK is being steadily deployed, that ca 85% of the average RTTs observed are between 15-500 ms, and that using larger initial values for cwnd is appropriate in most network paths.},
bibdate = {Thursday, October 24, 2002 at 13:39:43 (CEST)},
submitter = {Katarina Asplund}
}
@misc{rfc3168,
author="K. Ramakrishnan and S. Floyd and D. Black",
title={The Addition of Explicit Congestion Notification ({ECN}) to {IP}},
series="Request for Comments",
number="3168",
howpublished="RFC 3168 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2001,
month=sep,
url="http://www.ietf.org/rfc/rfc3168.txt",
}
@article{Ely01,
author = {D. Ely and N. Spring and D. Wetherall and S. Savage and T. Anderson},
title = {Robust Congestion Signaling},
journal = {proceedings of International Conference on Network Protocols, Riverside, CA},
year = {2001},
month = {November},
annote = {One disadvantage with ECN is that it fairly easy to implement a deceitful client, that conceals congestion signals from servers. By doing that, a client can gain up to ten times its fair share of the bandwidth. This paper presents a mechanism that enables a router to signal congestion to the sender without trusting the receiver or other network devices along the signaling path. The mechanism carries one-bit nonces on unmarked IP packets from sender to receiver, and returns one-bit cumulative nonces from the transport receiver to the transport sender. This allows the sender to probabilistically check that congestion signals are not being concealed without trusting any party other than the marking router.},
submitter = {Katarina Asplund},
bibdate = {Thursday, October 24, 2002 at 14:12:00 (CEST)}
}
@article{GSM0360,
author = {ETSI},
title = {{Digital cellular telecommunications system (Phase 2+); General Packet Radio Service (GPRS); Service description; Stage 2 (GSM 03.60 version 6.7.0 Release 1997)}},
journal = {ETSI},
year = {2000},
url = {Loke},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Thursday, October 24, 2002 at 16:17:28 (CEST)}
}
@inproceedings{Ohsaki01,
author = {H. Ohsaki and M. Murata and H. Miyahara},
title = {Modeling End-to-End Packet Delay Dynamics of the {Internet} using System Identification},
booktitle = {17th International Teletraffic Congress},
year = {2001},
month = {December},
annote = {This paper proposes an approach to model the end-to-end packet delay dynamics of the Internet using system identification. The end-to-end delay dynamics is modeled as a SISO system. The input to the system is the packet inter-departure time from the source host, and the output from the system is the end-to-end packet delay variation measured by the destination host. The system is modeled according to the so-called ARX (Auto-Regressive Exogenous) model. Through several numerical examples using MATLAB, they show that the ARX model accurately captures the end-to-end packet-delay dynamics. Furthermore, it is shown that competing traffic can be modeled as white noise.},
submitter = {Karl-Johan Grinnemo}
}
@mastersthesis{Yamashita01,
author = {T. Yamashita and N. Wakamiya and M. Murata and H. Miyahara},
title = {Integrated Resource Allocation Scheme for Real-Time Video Multicast},
school = {Graduate School of Engineering Science, Osaka University},
year = {2001},
address = {Osaka, Japan},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Wakamiya02,
author = {N. Wakamiya and M. Miyabayashi and M. Murata and H. Miyahara},
title = {Dynamic Quality Adaptation Mechanisms for {TCP}-Friendly {MPEG-4} Video Transfer},
booktitle = {{IEEE} International Conference on Multimedia & Expo},
year = {2001},
address = {Lausanne, Switzerland},
month = {February},
annote = {This paper presents a video quality adjustement mechanism that enables stable and TCP-friendly video transfer in a lossy environment.},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Miyabayashi01,
author = {M. Miyabayashi and N. Wakamiya and M. Murata and H. Miyahara},
title = {{MPEG-TFRCP}: Video Transfer with {TCP}-Friendly Rate Control Protocol},
booktitle = {{IEEE} International Conference on Communications ({ICC2001})},
year = {2001},
address = {Helsinki, Finland},
month = {June},
annote = {This paper evaluates the suitability of using TFRCP for MPEG video transfers. It is shown that without any measures taken, TFRCP is not suitable for MPEG. However, the paper also proposes some modifications to TFRCP which dramatically increase the video quality.},
submitter = {Karl-Johan Grinnemo}
}
@article{Hasegawa01,
author = {G. Hasegawa and M. Murata},
title = {Dynamic Threshold Control of {RED} for Establishing Fairness among Thousands of {TCP} Connections},
journal = {{SPIE ITCom}},
year = {2001},
month = {April},
annote = {One approach to deal with Internet congestion is to use Active Queue Management. A particularly interesting AQM algorithm is Random Early Detection (RED). With RED packets are dropped with a certain probability at times of incipient network congestion. This, in contrast to Drop-Tail which simply discards all arriving packets when the router buffer starts to fill upp. A problem with RED is the difficulty of setting its parameters. In fact, if the parameters are misconfigured, the throughput of a RED router can be lower than a Drop-Tail router. This paper, proposes an enhancement to RED called dt-RED (dynamic threshold RED), which dynamically regulates the RED parameters according to the observed behavior of the RED queue. The effectiveness of dt-RED is demonstrated with some simulation experiments. Specifically, the simulations show that dt-RED is able to improve throughput as well as fairness among competing flows as compared to RED.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 03, 2002 at 12:04:57 (CET)}
}
@article{Hasegawa01b,
author = {G. Hasegawa and M. Murata},
title = {Survey on Fairness Issues in {TCP} Congestion Control Mechanisms},
journal = {{IEICE} Transactions on Communications},
year = {2001},
pages = {1461--1472},
month = {June},
annote = {This paper surveys the fairness issues related to the TCP congestion control. First, the congestion control mechanisms of TCP Reno are discussed. The problems with TCP Reno in the cases when we have different propagation delays and link capacities are recognized. As examples of approaches to partly solve these problems Random Early Detection (RED) and Explicit Congestion Notification (ECN) are mentioned. Furthermore TCP Vegas are described. Second, Active Queueing is discussed. Queueing algorithms such as Round Robin and Deficit Round Robin are considered. The paper concludes with a discussion about TCP-friendly rate control.},
bibdate = {Sunday, November 03, 2002 at 12:18:30 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Reinbold01,
author = {P. Reinbold and O. Bonaventure},
title = {A Comparison of {IP} Mobility Protocols},
journal = {Technical Report, Infonet-2001-07, University of Namur},
year = {2001},
month = {June},
annote = {The advent of GPRS and UMTS, and the convergence towards an all-IP network infrastructure, have spawned a demand for effective, scalable, and robust mobility protocols. Generally, mobility support protocols are grouped into two classes: inter-mobility protocols and intra-mobility protocols. Inter-mobility protocols manage the mobility between different domains, i.e., between wireless networks governed by different administrative units, while intra-mobility protocols manage the mobility within a particular domain. This paper surveys inter- as well as intra-domain mobility protocols and tries to point out their strengths and weaknesses. The conclusion of the paper is that neither of the existing mobility support protocols will be able to adequately support GPRS and UMTS, and that this is a field that is in great need of further research.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, November 03, 2002 at 19:22:28 (CET)}
}
@article{Hasegawa01c,
author = {G. Hasegawa and M. Murata},
title = {Analysis of Dynamic Behaviors of Many {TCP} Connections Sharing Tail-Drop/{RED} Routers},
journal = {Proceedings of {IEEE GLOBECOM}},
year = {2001},
month = {November},
annote = {The main contribution of this paper is a new analysis method for determining the distribution of the TCP sender window when several TCP connections share the same bottleneck router. The analysis method is used to evaluate the TCP window size distribution when several TCP connections are passing through a Drop-Tail and RED router, respectively. It is shown that RED does not help improve the throughput of the router even when appropriately configured. However, it is also shown that RED is still useful to provide fairness among competing TCP connections.},
url = {papers/Hasegawa01c.pdf},
bibdate = {Monday, November 04, 2002 at 08:05:53 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Barford01,
author = {P. Barford and M. Crovella},
title = {Critical Path Analysis of {TCP} Transactions},
journal = {{IEEE/ACM} Transactions on Networking},
year = {2001},
volume = {9},
number = {3},
pages = {238--248},
month = {June},
annote = {Improving the performance of data transfers in the Internet, e.g., Web transfers, requires a detailed understanding of when and how delays are introduced. However, doing this is rather difficult, especially since delays in the Internet are often caused by a mix of network and server delays. This paper addresses the problem of finding the root causes of delays in the Internet by employing a method based on critical path analysis. The method has been implemented in a tool called tcpeval. Experiments with tcpeval over nodes at Boston University, University of Denver, and Harvard University, reveals that server load is the major cause of delayed transfer times for small files, while network load is the major cause for large files. },
bibdate = {Sunday, November 10, 2002 at 17:23:17 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Christiansen01,
author = {M. Christiansen, K. Jeffay, D. Ott, F. D. Smith},
title = {Tuning {RED} for Web Traffic},
journal = {{IEEE/ACM} Transactions on Networking},
year = {2001},
volume = {9},
number = {3},
pages = {249--264},
month = {June},
annote = {This paper study the effects of RED for Web traffic. The performance metric studied are response time for HTTP request-response pairs. The performance of RED is evaluated accross a range of parameter settings and traffic loads. The outcome suggest that RED has minimal effect on HTTP response times for traffic loads up to 90%, while it can give some improvement for Web traffic for traffic loads between 90% and 100%. However, the improvements that can be obtained with RED are to a large extent dependent on the RED parameter settings: a suboptimal parameter setting can in fact result in worse performance.},
bibdate = {Sunday, November 10, 2002 at 17:35:25 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Wang02,
author = {S. Y. Wang and H. T. Kung},
title = {A new methodology for easily constructing extensible and high-fidelity {TCP/IP} network simulators},
journal = {Computer Networks},
year = {2002},
volume = {40},
number = {2},
pages = {257-278},
month = {October},
annote = {Presents a simulation approach based on using an actcual FreeBSD stack implementation in one machine to do all protocol processing and using tunnel interfaces to be able to construct a more complex topology. Discusses virtual clocks and event scheduling. This tool might be useful, or some ideas might be reused in our emulation setup.},
url = {papers/Wang02TCPStackSimulation.pdf},
bibdate = {Monday, November 11, 2002 at 12:09:40 (CET)},
submitter = {Johan Garcia}
}
@article{NRC2002,
author = {Committee on Broadband Last Mile Technology, Computer Science and Telecommunications Board, National Research Council},
title = {Broadband: Bringing Home the Bits},
journal = {{ACM/IEEE} Computer Communication Review},
year = {2002},
volume = {32},
number = {2},
pages = {5--29},
month = {April},
annote = {The sudden decline in interest for broadband which started with the recession in 2000, has raised concerns for the importance of this technology; this goes for Europe as well as U.S. In an attempt to analyze the economical remifications for broadband, as well as how to inject a renewed interest in broadband technologies and broadband-dependent application providers in the U.S., the U.S. National Research Council, the research arm of the U.S. Academy of Science, have conducted an extensive study of the broadband market and published this in a report. This paper is an executive summary of this report and well worth reading. First, the paper establishes the present status of broadband in the U.S., and compares the evolution of broadband with that of telephony and computers. Then, it presents a list of the findings of the report. Although, the findings are not directly applicable to Sweden, I still think there are some interesting points noted in them that applies to our situation as well. For example, the paper argues against governmental intervention and for facitilies-based competition (in contrast of using the equipment of a regional/national telecommunication/cable provider). Furthermore, the paper concludes that that there are no such thing as one broadband technology that fits all regions. In densely populated areas, DSL technologies might be beneficial, however in areas with long local loops this is probably not the case. Another, related issue brought up in the paper, is how to spur competition in sparsely populated areas. Contrary to urban areas, this paper argues for govermental intervention when it comes to these areas. However, it is emphasized that the prices for broadband in less populated areas have to be continuously monitored so that it is not over-priced, i.e., follows the prices in the urban areas.},
url = {papers/nrc2002.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Saturday, November 23, 2002 at 12:00:20 (CET)}
}
@article{Belenki02,
author = {S. Belenki},
title = {An Enforced Inter-Admission Delay Performance-Driven Connection Admission Control Algorithm},
journal = {Computer Communication Review},
year = {2002},
volume = {32},
number = {2},
pages = {31--41},
month = {April},
annote = {An important component in QoS aware networks are admission control. The function of an admission control system is to decide whether a new connection should be admitted or not on the basis on available network resources. Frequently, admission control algorithms use measurements to find out about available network resources. However, this paper proposes a QoS architecture independent admission control algorithm that entails no measurements. Instead, the algorithm bases its admission decisions on the packet-loss rate. If the packet-loss rate is below a target packet-loss rate, a new connection is admitted, otherwise not. The performance of the proposed admission algorithm has been evaluated through simulations. The simulations showed that the admission algorithm does not achieve as high link utilization as measurement-based admission control algorithms as measured-sum. However, at the same time it should be noted that the proposed admission control algorithm makes use of much less information than measured-sum.},
url = {papers/belenki02.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Saturday, November 23, 2002 at 18:52:55 (CET)}
}
@article{Q700,
author = {{ITU-T}},
title = {{Q.700}: Specifications of Signalling System No. 7},
journal = {{ITU-T}},
year = {1993},
month = {March},
annote = {This recommendation serves as an introduction to the CS7/SS7 signalling system. It details the objectives of the CS7/SS7 signalling system, and gives a brief overview of the principal protocols of CS7/SS7. Everyone interested in knowing CS7/SS7 should start reading this recommendation. Contrary to many other recommendations, this recommendation is quite easy to read.},
url = {papers/q700.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, November 28, 2002 at 09:19:25 (CET)}
}
@article{Q701,
author = {{ITU-T}},
title = {{Q.701}: Functional Description of the Message Transfer Part ({MTP}) of Signalling System No. 7},
journal = {{ITU-T}},
year = {1993},
month = {March},
annote = {This ITU-T recommendation provides a succinct overview of the design and implementation of the Message Transfer Part of SS7. That is those parts of SS7 which approximately provides the same functionality as layers 1 through 3 in the OSI reference model. It is recommended that readers who intend to read Q.704, which describes MTP-L3, first skims through this recommendation.},
url = {papers/q701.pdf},
bibdate = {Thursday, November 28, 2002 at 17:51:25 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Q702,
author = {{ITU-T}},
title = {{Q.702}: Signalling Data Link},
journal = {{ITU-T}},
year = {1993},
annote = {Signaling System No. 7 (SS7) is designed to work over a data link exclusively dedicated to signaling traffic. This recommendation details the requirements imposed on the data link by SS7. It also discusses solutions for cases in which SS7 have to operate in less-than-ideal environments.},
url = {papers/q702.pdf},
bibdate = {Monday, December 02, 2002 at 11:41:22 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Q704,
author = {{ITU-T}},
title = {{Q.704}: Signalling Network Functions and Messages},
journal = {{ITU-T}},
year = {1996},
month = {July},
annote = {This recommendation gives a detailed description of MTP-L3. A reader used to read RFC's, will probably find this recommendation hard to read. However, for a reader who seeks succinct facts of the different modules comprising MTP-L3, this is the recommendation to read.},
url = {papers/q704.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, December 02, 2002 at 11:59:13 (CET)}
}
@article{Q703,
author = {{ITU-T}},
title = {{Q.703}: Signalling Link},
journal = {{ITU-T}},
year = {1996},
month = {July},
annote = {This recommendation describes the functions and procedures pertaining to level 2 of the Message Transfer Part (MTP) in the SS7 stack. The recommendation includes the following topics: delimitation, alignment, error detection and correction, error monitoring, and flow and congestion control. For everyone looking for information about level 2 issues, this is were to look.},
url = {papers/q703.pdf},
bibdate = {Monday, December 02, 2002 at 15:07:53 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Q706,
author = {{ITU-T}},
title = {{Q.706}: Signalling System No.7 - Message Transfer Part Signalling Performance},
journal = {{ITU-T}},
year = {1993},
month = {March},
annote = {The Message Transfer Part (MTP) of SS7 is designed with the intent of functioning as a joint transport system for all user parts (e.g., ISUP, TUP, DUP). In so doing, MTP has to meet some lower-bound requirements. This recommendation specifies these requirments. When applicable, the recommendation also motivates the requirments by showing how they were derived. The recommendation is very succinctly written which makes some passages of the text unnecessary hard to understand.},
url = {papers/q706.pdf},
bibdate = {Monday, December 02, 2002 at 15:46:03 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Q709,
author = {{ITU-T}},
title = {{Q.709}: Signalling System No. 7 - Hypothetical Signalling Reference Connection},
journal = {{ITU-T}},
year = {1993},
month = {March},
annote = {This recommendation illustrates how the various components of a SS7 network (i.e., signaling points and signaling transfer points) are combined to meet the the service requirements of the user parts, i.e., the requirements of Q.706. Since the recommendation is very brief, it leaves very much to the reader to figure out.},
url = {papers/q709.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, December 02, 2002 at 16:06:46 (CET)}
}
@article{Q761,
author = {{ITU-T}},
title = {{Q.761}: Signalling System No. 7 - ISDN user part functional description},
journal = {{ITU-T}},
year = {1999},
month = {December},
annote = {},
url = {},
submitter = {Johan Eklunf},
bibdate = {August 30, 2007 at 12:06:46 (CET)}
}
@article{Q771,
author = {{ITU-T}},
title = {{Q.771}: Signalling System No. 7 - functional description of transaction capabilities},
journal = {{ITU-T}},
year = {1997},
month = {June},
annote = {},
url = {},
submitter = {Johan Eklund},
bibdate = {August 30, 2007 at 12:16:12 (CET)}
}
@article{Jungmaier00,
author = {Andreas Jungmaier and Michael Schopp and Michael Tuexen},
title = {Performance Evaluation of the Stream Control Transmission Protocol},
journal = {Proceedings of the IEEE Conference on High Performance Switching and Routing},
year = {2000},
pages = {141--148},
month = {June},
annote = {This paper studies the fairness of SCTP when competing with TCP traffic. The conclusion of the paper is that SCTP indeed competes fairly with TCP traffic. Furthermore, it is observed that multiplexing several SCTP streams into a single association results in a less aggressive behavior than opening one association per stream.},
url = {papers/jungmaier00.pdf},
bibdate = {Monday, December 02, 2002 at 18:02:33 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Gradischnig,
author = {K. D. Gradischnig and M. Tuexen},
title = {Signaling Transport over {IP}-Based Networks using {IETF} Standards},
journal = {In Proceedings of the Design of Reliable Communication Networks},
year = {2001},
month = {October},
annote = {This paper analyzes some of the reliability features of the transport layers of the signaling system no. 7 (SS7) and compares them to the reliability and performance achievable with the protocol stack for signaling transport over IP currently being defined in the SIGTRAN working group of the IETF. It identifies parameters which have to be adjusted and restrictions to available addressing options which have to be made in order for the SIGTRAN protocol stack to achieve the reliability and performance of SS7. },
url = {papers/gradischnig.pdf},
bibdate = {Monday, December 02, 2002 at 18:44:59 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Stewart02b,
author = {Randal R. Stewart and Lyndon Ong and Ivan Arias-Rodriguez and Kacheong Roon and Phillip T. Conrad and Armando L. Caro Jr. and Michael Tuexen },
title = {Stream Control Transmission Protocol (SCTP) Implementers Guide - Work in progress},
journal = {http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-07.txt},
year = {2002},
pages = {62},
month = {October},
annote = {I denna draft står det beskrivet om de problem som har uppmärksammats och hur de har lösts. Exempelvis så beskrivs det varför SCTP får överskrida cwnd.},
url = {http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-sctpimpguide-07.txt},
submitter = {Torbj\"{o}rn Andersson},
bibdate = {Wednesday, December 04, 2002 at 14:13:04 (CET)}
}
@article{KUROSE01,
author = {James F. Kurose and Keith W. Ross},
title = {Computer Networking: A Top-Down Approach Featuring the Internet},
journal = {Addison Wesley},
year = {2001},
pages = {688},
bibdate = {Friday, December 06, 2002 at 10:04:16 (CET)},
submitter = {Torbj\"{o}rn Andersson}
}
@techreport{Grinnemo02b,
author = {K-J Grinnemo and A. Brunstrom and J. Garcia},
title = {A Taxonomy and Survey of Retransmission Based Partially Reliable Transport Protocols},
institution = {Dept. of Computer Science, Karlstad University},
year = {2002},
number = {2002:34},
month = {Oct},
url = {papers/grinnemo02b.ps.gz},
submitter = {Karl-Johan Grinnemo}
}
@techreport{Grinnemo02c,
author = {K-J Grinnemo and A. Brunstrom and J. Garcia},
title = {A Simulation Based Performance Evaluation of {PRTP}},
institution = {Dept. of Computer Science, Karlstad University},
year = {2002},
number = {2002:35},
month = {Oct},
url = {papers/grinnemo02c.ps.gz},
submitter = {Karl-Johan Grinnemo}
}
@article{Vardalis02,
author = {D. Vardalis and V. Tsaoussidis},
title = {Efficiency/Fairness Tradeoffs in Networks with Wireless Components and Transient Congestion},
journal = {The Journal of Supercomputing},
year = {2002},
volume = {23},
number = {3},
pages = {281-296},
month = {November},
annote = {Might relate to my wireless results... They test x-kernel implementations of tahoe and reno over channels with wireless and mixed error sources. Abstract:Our work explores the impact of aggressive/conservative congestion control strategies on the fairness and efficiency of reliable transport protocols, in a wired/wireless environment. Based on experiments, we study the behavior of congestion control mechanisms in response to wireless errors, and transient congestion caused by a small number of competing flows. We show that: (i) the traditional TCP algorithm proves to be inadequate in terms of efficiency and fairness when random wireless errors occur in the network and, (ii) an aggressive strategy does not necessarily yield better performance. On the contrary, in the combined presence of transmission errors and transient congestion a conservative strategy appears superior. },
url = {papers/Vardalis02_EffFairnWirelss.pdf},
bibdate = {Tuesday, December 10, 2002 at 18:23:26 (CET)},
submitter = {Johan Garcia}
}
@article{Pan02,
author = {Jianping Pan and Jon W. Mark and Sherman X. Shen },
title = {TCP Performance and Behaviors with Local Retransmissions},
journal = {The Journal of Supercomputing},
year = {2002},
volume = {23},
number = {3},
pages = {225-244},
month = {November},
annote = {Discusses the tradeoff between local retransmissions and the risk of tcp timeouts. Also exmaines the effect of spurious dup-acks and how DSACK fixes this. Abstract: TCP has been the dominant transport protocol over the global Internet, and its performance over a hybrid wireless/wireline network has attracted much attention in recent years. This paper investigates the end-to-end TCP performance, in terms of normalized throughput, effective goodput, and packet delay, over wireless lossy links with local retransmissions. The results reveal that local retransmissions can increase the normalized TCP throughput in different wireless bandwidth, delay, and error settings, at the cost of a decrease in effective goodput and an increased packet delay. The performance observation is explained by the explored TCP endpoint behaviors, including the spurious timeout and duplicated acknowledgment. Analysis shows that spurious timeouts with local retransmissions are rare due to the conservative TCP timeout algorithm. However, spurious duplicated acknowledgments have negative impact and a further improvement with the D-SACK proposal is evaluated. },
url = {/papers/Pan02TCPLocalRxmit.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, December 10, 2002 at 18:37:11 (CET)}
}
@article{Langendorfer2002,
author = { 23(3): 245-260; Nov 2002Peter Langend\"{o}rfer and Michael Methfessel and Horst Frankenfeldt and Irina Babanskaja and Irina Matthaei and Rolf Kraemer},
title = {The Journal of Supercomputing 23(3): 245-260; Nov 2002 },
journal = {The Journal of Supercomputing},
year = {2002},
volume = {23},
number = {3},
pages = {245-26},
month = {November},
annote = {Presents Bones-simulations of TCP and WLAN MAC for uplink transmissions. Used a fixed Internet bw of 200kbps and varied the internet path delay, and the loss characteristics. Maybe not the best paper, but some useful info might be here.... Abstract: A known problem for TCP connections over wireless links is that errors in the wireless channel interfere with the TCP protocol even for minor packet loss. In the first part of this paper we evaluate how the data rate reduction depends on the channel delay. For comparatively short delays in the order of 100 ms, the decrease of the throughout is noticeable but not dramatic. This indicates that the problem is not severe if the communication partners are located in the same WLAN or interact over a fast Internet connection. A significant throughput reduction arises in the case of a large network delay. Simulation results for the uplink transmission are presented as part of an overall strategy in which all improvements are made by optimizing the mobile end device only, an approach which allows performance improvements without any protocol modifications. },
url = {papers/Langendorfer02ShieldingTCPWireless.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, December 10, 2002 at 19:19:02 (CET)}
}
@techreport{Grinnemo02d,
author = {K-J Grinnemo and A. Brunstrom},
title = {A Study of Partially Reliable Transport Protocols for Soft Real-Time Applications},
institution = {Dept. of Computer Science, Karlstad University},
year = {2002},
number = {2002:36},
month = {Nov},
url = {papers/grinnemo02d.ps.gz},
submitter = {Karl-Johan Grinnemo}
}
@article{Lin99,
author = {H-A P. Lin and A. Broscius and C. Huitema},
title = {{VoIP} Signaling Performance Requirements and Expectations},
journal = {{IETF}},
year = {1999},
month = {June},
annote = {This document serves as input to the IETF SIGTRAN requirements process. It includes call setup delay requirements, derived from relevant ISDN and SS7 standards published by ITU-T and generic requirements published by Telcordia Technologies.},
url = {papers/lin99.txt},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, December 17, 2002 at 08:05:42 (CET)}
}
@article{Coene02,
author = {L. Coene and J. Pastor},
title = {Telephony Signalling Transport over {SCTP} Applicability Statement},
journal = {{IETF}},
year = {2002},
month = {November},
annote = {The document describes how adaptation protocols (e.g., M3UA) and SCTP have to be used to transport signaling information over IP.},
url = {papers/Coene02.txt},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, December 17, 2002 at 08:11:48 (CET)}
}
@article{Loguinov02b,
author = {D. Loguinov and H. Radha},
title = {Retransmission Schemes for Streaming {Internet} Multimedia: Evaluation Model and Performance Analysis},
journal = {Computer Communication Review},
year = {2002},
volume = {32},
number = {2},
pages = {70--83},
month = {April},
annote = {Recall that TCP's RTO estimation essentially comprises two algorithms. One for calculating the so-called smoothed RTT estimator (1) and one for calculating the smoothed variance (2): (1) SRTT_0 = RTT_0; (1 - alpha) * SRTT_i-1 + alpha * RTT_i, (2) SVAR_0 = RTT_0 / 2; (1 - beta) * SVAR_i-1 + beta * VAR_i. The RTO is then calculated as (3) RTO(t) = n * SRTT_i + k * SVAR_i This paper performs a thorough analysis of the consequences changing the parameters, alpha, beta, n, and k, has on the transport protocol performance in terms of overestimating the actual round-trip time, and generating duplicate packets because of underestimation of the actual round-trip time. Usually streaming multimedia transport protocols uses a NACK based retransmission scheme, i.e., instead of acknowledge successfully received packets, they only notify the sender of lost or corrupted packets. Consequently, streaming transport protocols have a low frequency sampling of their RTT estimation. In contrast, ACK-based retransmission schemes, such as the one employed by TCP, have a high sampling rate. In this paper it is shown that the Jacobson RTT estimation algorithm is NOT suitable for transport protocols employing a low-frequency RTT estimation algorithm. In particular, it is shown that transport protocols that uses low-frequency sampling of the RTT could improve their performance by using (4) instead of (1), (2) and (3). (4) RTO(t) = n * RTT_i + m * VAR_i, The parameters n and m in (4) have been calculated using an optimization method called "Downhill Simplex Method in Multidimensions" (Lagrange and similar methods cannot be used since they assume the existence of partial derivatives.). Most notably with this result is that the optimal calculation of the RTO at low sampling frequencies of RTT does not include the smoothing factors. The reason to this is of course that the low frequence of the sampling makes the history of RTT's obsolete. Even better performance than with (4) was obtained when the jitter characteristics were included. To be more specific, it is shown that by including the so-called smoothed inter-burst delay variance, a more than 60% performance improvement compared to (4), and consequently even more for (3). When the frequency of the RTT sampling increased, the Jacobson RTO estimation in (3) approached the performance of (4) and the RTT estimation including jitter. That is, at high frquency sampling of RTT, the RTO calculation of TCP is more or less optimal, but degrades drastically when the sampling frequence decreases. },
url = {papers/Loguinov02b.pdf},
bibdate = {Thursday, December 19, 2002 at 12:11:53 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Loguinov02c,
author = {D. Loguinov and H. Radha},
title = {Increase-Decrease Congestion Control for Real-Time Streaming: Scalability},
journal = {Proceedings of {IEEE INFOCOM}},
year = {2002},
month = {June},
annote = {It is well acknowledged that the stability and performance of the current Internet is in large part attributable to the ACK-based congestion control of TCP. However, current real-time streaming applications typically rely on rate-based congestion control with, at best, rudimenatary congestion control. Although, it could be argued that real-time streaming applications do not have to compete fairly with TCP, they still have to compete fairly with each other, and not exhibit a large performance degradation in terms of packet loss when the number of competing flows increases. The latter requirement is in this paper called "scalability". This paper theoretically analyses the performance of rate-based congestion control - especially with respect to scalability. The focus of the paper is on increase-decrease congestion control and then in particular on binomial schemes (c.f. the work of Bansal and Balakrishnan). It is shown that among the TCP-friendly binomial schemes, the Additive-Increase-Multiplicative-Decrease (AIMD) scales best. Furthermore, it is shown that the scalability of an AIMD scheme could be substantially increased if an upper bound of the bottleneck link capacity is known. The paper concludes with an introduction to the concept of Ideally-Scalable Congestion Control, and theoretically demonstrates how values of the bottleneck capacity could be used to improve the scalability of an arbitrary AIMD scheme.},
url = {papers/Loguinov02c.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, January 01, 2003 at 11:07:44 (CET)}
}
@article{Ding01,
author = {Winqing Ding and Abbas Jamalipour},
title = {A New Explicit Loss Notification with Acknowledgement for Wireless TCP},
booktitle={Proceedings of the IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)},
year = {2001},
volume = {1},
pages = {B-65-B-69 },
url = {papers/Ding01_eln_with_ack_for_wireless_tcp.pdf},
bibdate = {Tuesday, January 07, 2003 at 10:01:31 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Trickey02,
author = {A. Trickey},
title = {Simply {SS7}},
journal = {{SS8} Networks},
year = {2002},
annote = {This booklet gives a broad, yet relatively detailed, understanding of the SS7 network architecture and protocols. From a personal perspective, I would say that this text prepares you for reading the ITU standards. Compared to the book "Signaling System #7" by Travis Russel, this book roughly corresponds to the intermediate level. The booklet comprises four sections. After having motivated the existence of SS7 in Section 1, Section 2 elaborates the logical view of the SS7 network architecture. That is, concepts like associated links, cross links etc. are discussed. In Section 3, the SS7 protocol suite are surveyed. The emphasis of the survey is the function of MTP-L2 and MTP-L3. Finally, in Section 4, the formats of the three types of packets found in an SS7 network, MSU, LSSU, FISU, are described. This section also extends the discussion in Section 3 flow and congestion control by briefly mention the use of the signal unit fields "forward indicator bit", "forward sequence number", "backward indicator bit", and "backward sequence number".},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, January 08, 2003 at 15:13:28 (CET)}
}
@article{SS802,
author = {{SS8} Networks},
title = {{SS7} Pocket Guide},
journal = {{SS8} Networks},
year = {2002},
annote = {This pocket guide to SS7 is essntially a summary of the booklet "Simply SS7". Although, it provides an introduction to the SS7 architecture and protocols, I find this guide to be too succinct for its own good. In particular, too much information on the SS7 protocols has been left out to be able to confer a real understanding of how these protocols interwork to provide call/network control and supervision.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, January 09, 2003 at 10:12:36 (CET)}
}
@inproceedings{Garcia02,
author = {Garcia, J. and Brunstrom, A.},
title = {Checksum-based Loss Differentiation},
booktitle = {Proceedings 4th IEEE Conference on Mobile and Wireless Communications Networks (MWCN)},
address={Stockholm, Sweden},
year = {2002},
month = sep,
url = {papers/Garcia02_MWCN.pdf},
bibdate = {Friday, January 10, 2003 at 22:20:21 (CET)},
submitter = {Johan Garcia}
}
@article{Nagarajan99,
author = {R. Nagarajan},
title = {Threshold-Based Congestion Control for the {SS7} Signaling Network in the {GSM} Digital Cellular Network},
journal = {{IEEE} Transactions on Vehicular Technology},
year = {1999},
volume = {48},
number = {2},
pages = {385--396},
month = {March},
annote = {This paper evaluates the SS7 congestion control scheme in the context of GSM. The principal contributions of this paper are a recommendation on the throttling of traffic at the originating signaling point at times of congestion, and a procedure to calculate the size of the congestion thresholds of the MTP-L2 transmission buffers so as to optimize performance during congestion. The paper also briefly considers the consequences of delayed congestion onset and abatement mechanisms.},
url = {papers/Nagarajan99.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, January 16, 2003 at 09:22:14 (CET)}
}
@article{Silberschatz02,
author = {Avi Silberschatz and Peter B. Galvin and Greg Gagne},
title = {Operating System Concepts, Sixth Edition},
journal = {John Wiley & Sons, Inc},
year = {2002},
pages = {803},
submitter = {Torbj\"{o}rn Andersson},
bibdate = {Tuesday, January 28, 2003 at 16:46:35 (CET)}
}
@article{Jungmaier00a,
author = {A. Jungmaier and E. P. Rathgeb and M. Schopp and M. Tuxen},
title = {SCTP - A Multi-link End-to-end Protocol for IP-based Networks},
journal = {AEU International Journal of Electronics and Communications},
year = {2001},
volume = {55},
number = {1},
pages = {46-54},
annote = {This paper gives a nice overview (of some of the features) of the SCTP protocol and highlights some advantages it has over TCP. Measurements for a scenario with a satellite link protected by a terrestrial secondary path show the improvement in delivery delay compared to TCP. In addition, a modification of SCTP is proposed and evaluated which improves the delay performance even more. },
submitter = {Katarina Asplund},
bibdate = {Tuesday, February 04, 2003 at 13:35:02 (CET)}
}
@article{Feamster02,
author = {N. Feamster and H. Balakrishnan},
title = {Packet Loss Recovery for Streaming Video},
journal = {12th International packet video workshop, Pittsburgh, PA},
year = {2002},
month = {April},
annote = {This paper analyzes the effects of packet loss on the quality of MPEG-4 video and proposes a analytical model to explain these effects. It shows that, by recovery of only the most important data in the bitstream, significant performance gains can be achieved without much additional penalty in terms of latency. The authors have designed a system that uses a receiver-driven selective retransmission extension to RTP in conjunction with receiver postprocessing. The system also uses the Congestion Manager to perform TCP-friendly congestion control. Experiments show that the system is adaptive to changing network conditions and that selective reliability is a both feasible and beneficial means of improving the quality of received video. },
submitter = {Katarina Asplund},
bibdate = {Tuesday, February 04, 2003 at 14:01:32 (CET)}
}
@article{Rabinovich01,
author = {M. Rabinovich and H. Wang},
title = {DHTTP: An Efficient and Cache-Friendly Transfer Protocol for Web Traffic},
journal = {INFOCOM'2001},
year = {2001},
pages = {10},
annote = {Beskriver en utökning av HTTP där kontrollkanalen där "Request" skickas körs över UDP. Små dataobjekt som skickas från servern skickas över UDP för att undvika att belasta servern med små korta TCP koppel. I deras tester använder de en trafikgenerator som de kallar för SURGE, Workload generator. En beskrivning går att få av denna bland referenserna till artikeln.},
url = {http://www.research.att.com/~misha/dhttp/infocom01.pdf},
submitter = {Torbj\"{o}rn Andersson},
bibdate = {Wednesday, February 05, 2003 at 13:20:35 (CET)}
}
@article{Schaaf01,
author = {A. van der Schaaf and Langendoen, K. and Lagendijk, R.L.},
title = {Design of an adaptive interface betgween video compression and transmission protocols for mobile communications},
journal = {Proceedings of the 11th International Packet Video Workshop PV-2001, Kyongju, Korea},
year = {2001},
pages = {* 395 - 404},
month = {April},
annote = {Discusses some abstract QoS parameters and the relations these may have to coder, transport protocol and transfer channel. The suitability of bidirectional QoS negotiation is discussed.},
url = {papers/Schaaf01_AdaptiveIF.pdf},
bibdate = {Thursday, February 06, 2003 at 14:57:19 (CET)},
submitter = {Johan Garcia}
}
@article{Elaarag02,
author = {Hala Elaarag},
title = {Improving {TCP} performance over Mobile Networks},
journal = {ACM Computing Surveys},
year = {2002},
volume = {34},
number = {3},
pages = {357-374},
month = {August},
annote = {In this paper, we present how regular TCP is well tuned to react to packet loss in wired networks. We then define mobility and the problems associated with it. We discuss why regular TCP is not suitable for mobile hosts and their wireless links by providing simulation results that demonstrate the effect of the high bit error rates of the wireless link on TCP performance. We discuss and illustrate the problems caused by the mobility of hosts using a graph tracing packets between fixed and mobile hosts. We then present a survey of the research done to improve the performance of TCP over mobile wireless networks. We classify the proposed solutions into three categories: link layer, end-to-end and split. We discuss the intuition behind each solution and present example protocols of each category. We discuss the protocols functionality, their strengths and weaknesses. We also provide a comparison of the different approaches in the same category and on the category level. Additional Key Words and Phrases: standard TCP, mobile TCP, wireless TCP, TCP performance, wired networks, mobile wireless networks, mobility, base station, mobile host, comparison of TCP implementations, link layer, end-to-end, split TCP, snoop, Reno, New-Reno, SACK, I-TCP, WTCP, MTCP, M-TCP, WAP},
url = {papers/elaarag02_mobile_tcp_performance_survey-ACMCOPYRIGHT.pdf},
submitter = {Stefan Alfredsson},
bibdate = {Friday, February 07, 2003 at 16:07:54 (CET)}
}
@inproceedings{Goel02,
author = {Asvin Goel and Charles Krasic and Kang Li and Jonathan Walpole},
title = {Supporting Low Latency {TCP}-Based Media Streams},
booktitle = {Proceedings of the Tenth International Workshop on Quality of Service (IWQoS)},
address = {Miami Beach, Florida, USA},
year = {2002},
month = may,
url = {papers/Goel02_supporting_low_latency_TCP_based_media_streams.pdf},
bibdate = {Sunday, February 09, 2003 at 22:39:47 (CET)},
submitter = {Stefan Alfredsson}
}
@article{Lee02,
author = {D-S. Lee and C-C. Lin},
title = {Window Adaptive {TCP} for {EGPRS} Networks},
journal = {IEEE The 5th International Symposium on Wireless Personal Multi-media Communications},
year = {2002},
annote = {TCP performance over 2.5G and 3G is enhanced by decreasing the buffer occupancy and hence the packet delay. For transmissions in downlink this is achieved if the advertised receiver window in TCP on the mobile terminal is set to a value closer to the bandwidth-delay product over the wireless link. A modification to TCP is proposed. Information about the channel rate is passed from the physical to TCP which uses this information and rtt estimates to dynamically adjust the receiver window. The TCP modification is evaluated by event-driven simulations. The idea is better than the paper, partly because the figures are out of focus.},
url = {paper/WA_TCP_for_EGPRS.pdf},
bibdate = {Monday, February 10, 2003 at 15:40:35 (CET)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Mukhtar01,
author = {Rami G. Mukhtar and Stephen V. Hanly and Milosh Ivanovich and Paul Fitzpatrick and Hai L. Vu},
title = {Analysis of {TCP} Performance over Hybrid "Fast Fixed - to - Slow Wireless" Buffered Links},
journal = {IEEE Globecom},
year = {2001},
annote = {Throughput is increased by setting the advertised receiver window once before a TCP connection is set up. The setting is based on statistics from previous connections. Both theory and simulations are used to try to verify the idea. The simulation result is that there is a range of near optimal settings for live 802.11b, live GPRS, and emulated GPRS. I think it would be more interesting if other performance measures, such as delay and loss ratio, also were considered. The GPRS emulation does not have the same result as the live GPRS.},
url = {papers/Ivanovi_GPRS.pdf},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Monday, February 10, 2003 at 16:00:56 (CET)}
}
@inproceedings{Chakravorty02,
author = {R. Chakravorty and I. Pratt},
title = {{WWW} Performance over {GPRS}},
booktitle = {IEEE International Conference on Mobile and Wireless Communication Networks (MWCN)},
address = {Stockholm, Sweden},
year = {2002},
month = aug,
annote = {HTTP performance is investigated over a live GPRS network. A wired-wireless proxy solution is proposed which improves performance at the transport and application layer. The paper describe the characteristics of the GPRS link and TCP problems over GPRS. },
url = {papers/www_gprs_chakravorty.pdf},
bibdate = {Monday, February 10, 2003 at 16:27:22 (CET)},
submitter = {Annika Wennstr\"{o}m}
}
@inproceedings{Chakravorty02a,
author = {R. Chakravorty and J. Cartwright and I. Pratt},
title = {Practical Experience with {TCP} over {GPRS}},
booktitle = {Proceedings of the IEEE Global Communications Conference (GLOBECOM)},
address={Taipei, Taiwan},
year = {2002},
month = nov,
annote = {Live GPRS experiments are conducted to characterize the GPRS link. TCP problems due to high rtt and excessive queueing are described in detail. The authors propose "TCP cwnd clamping" which is a wired-wireless split connection proxy that avoids slow-start, and modifies cwnd and the advertised receiver window. },
url = {papers/practical_tcp_gprs_chakravorty.pdf},
bibdate = {Monday, February 10, 2003 at 16:39:40 (CET)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Stewart02c,
author = {Randal R. Stewart and M. Ramalho and Q. Xie and M. Tuexen and I. Rytina and M. Belinchon and P. Conrad},
title = {Stream Control Transmission Protocol (SCTP) Dynamic Address Reconfiguration},
journal = {http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-addip-sctp-06.txt},
year = {2002},
pages = {36},
month = {September},
note = {Work in Progress},
annote = {Descibes how .},
url = {http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-addip-sctp-06.txt},
bibdate = {Monday, February 17, 2003 at 10:0:04 (CET)},
submitter = {Torbj\"{o}rn Andersson}
}
@article{Willis03,
author = {D. Willis and B. Campbell},
title = {Session Initiation Protocol Extension to Assure Congestion Safety},
journal = {Internet Draft, draft-ietf-sip-congestsafe-01},
year = {2003},
month = {February},
annote = {The Session Initiation Protocol (SIP) provides application support over multiple transport protocols, including UDP and TCP. The fact that SIP could use UDP as a transport protocol opens up for congestion problems. To date, SIP attempts to deal with this in two ways: retransmission timers with exponential backoffs, and limiting the packet size when transmitting over UDP. The authors of this draft considers these attempts incomplete and introduces the concept "Congestion Safety". The idea is to pace SIP requests when sent over UDP, and avoid packet fragmentation by imposing a maximum size on SIP messages.},
url = {papers/willis03.txt},
bibdate = {Wednesday, February 19, 2003 at 08:42:53 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{tom03,
author = {Tom Dunigan},
title = {TCP auto-tuning zoo},
journal = {http://www.csm.ornl.gov/~dunigan/net100/auto.html},
submitter = {Torbjorn Andersson},
bibdate = {Tue Feb 19 10:44:17 CET 2003}
}
@article{Camarillo02,
author = {G. Camarillo and H. Schulzrinne and R. Kantola},
title = {Signalling Transport Protocols},
journal = {Technical Report, Dept. Computer Science, Columbia University, CUCS-002-02},
year = {2002},
month = {February},
annote = {The limitations present in the existing standard transport protocols of Internet, i.e., UDP and TCP, led to the design of SCTP. Although, it has been argued that SCTP is indeed a better transport protocol for signalling traffic than either of UDP or TCP, no experiments have so far been carried out which support this argument. This paper evaluates SCTP for signaling traffic by comparing SCTP with UDP and TCP. The signaling traffic was generated by a SIP traffic generator. Contrary to common beliefs, the difference in delays between SCTP and TCP due to head-of-line blocking was insignificant for moderate traffic loss (i.e., under network conditions considered suitable for signaling traffic). Not until the packet loss became significant, SCTP showed better delays than TCP, however at these packet-loss rates the delays for SCTP were still too large for many applications. Furthermore, SCTP turned out to perform worse than TCP when fragmentation occurred - an effect of SCTP using message counting instead of byte counting. Still another unpleasant surprise was that the window-based congestion control mechanisms of TCP and SCTP do not work well for signaling traffic. Specifically, these congestion control mechanisms are unable to cope with the burstiness of this traffic.},
url = {papers/Camarillo02.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Wednesday, February 19, 2003 at 17:52:26 (CET)}
}
@article{Rosenberg97,
author = {J. Rosenberg},
title = {Internet Telephony: A (Partial) Research Agenda},
journal = {Technical report, Columbia University},
year = {1997},
month = {October},
annote = {IP telephony has until recently been considered a "toy" application, or at best a complement to traditional PSTN telephony. As a consequence, many of the technologies behind IP telephony is quite immature. This paper, tries to compile the major challenges facing IP telephony. Three issues are considered: transport protocol issues, robustness issues, and signaling and architectual issues. Briefly, these issues consider problems like RTP scaling (e.g., too long intervals between RTCP messages) and multiplexing, QoS (e.g., integrated service solutions vs. application adaptation), inter-IP-PSTN signaling.},
url = {papers/Rosenberg97.pdf},
bibdate = {Saturday, February 22, 2003 at 18:26:24 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@phdthesis{Wang99,
author = {S-Y Wang},
title = {Decoupling Control from Data for {TCP} Congestion Control},
school = {Harward University},
year = {1999},
annote = {This thesis proposes an approach, TCP decoupling, for solving, or at least mitigate, the problems inherent with TCP's error and congestion control being intertwined. Briefly, TCP decoupling entails using two physical connections per logical connection: One connection used for the transmission of data packets and one for controlling the transmission of the data packets. On the control connection, TCP header packets are sent (i.e., no payload data) at a pace governed by the TCP congestion control. Each successfully sent TCP header packet generates a credit for sending a certain amount of data on the data connection. Consequently, the data connection behaves in a TCP-Friendly manner without having to use TCP's error recovery scheme. Three applications that could benefit from using the TCP decoupling approach is presented: a "TCP Trunk" which uses TCP congestion control to probe for bandwidth while keeping the packet-loss rate low, transfer over a wireless link, and transfer of streaming media.},
url = {papers/Wang99.pdf},
submitter = {Karl-Johan Grinnemo}
}
@misc{RFC3465,
author = {M. Allman BBN/NASA GRC},
title = {{RFC 3465: TCP} Congestion Control with Appropriate Byte Counting (ABC)},
journal = {http://www.ietf.org/},
year = {2003},
pages = {10},
month = {February},
annote = {Beskriver en "Byte-counting" algorithm för TCP, som idag är "ACK-counting".},
url = {http://www.ietf.org/rfc/rfc3465.txt?number=3465},
submitter = {Torbj\"{o}rn Andersson},
bibdate = {Monday, February 24, 2003 at 09:35:17 (CET)}
}
@article{Baran02,
author = {P. Baran},
title = {The Beginnings of Packet Switching: Some Underlying Concepts},
journal = {{IEEE} Communications Magazine},
year = {2002},
volume = {40},
number = {7},
pages = {42--48},
month = {July},
annote = {This article was written by Paul Baran, the inventor of packet switching, at a seminar held on the occasion of the Franklin Institute's 2001 Bower Award and Prize for Achievement in Science. It describes the author's work during the beginning of the 60's on packet-switching. In particular, the article considers the rationale behind creating the key concepts of packet switching. Concepts discussed include: availability, routing, logical and physical addressing, and virtual circuits.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Saturday, March 01, 2003 at 10:49:00 (CET)}
}
@article{Andrews02,
author = {F. T. Andrews},
title = {The Telephone Network of the 1960s},
journal = {{IEEE} Communications Magazine},
year = {2002},
volume = {40},
number = {7},
pages = {49--53},
month = {July},
annote = {This article, written by one of the front figures of Bellcore back in the 1960s, puts todays development of an all-IP telephone network infrastructure into perspective. It describes the advancement from analog to digital switching technologies, and elaborates the differences in the objectives of the PSTN network and the packet-switched data network.},
bibdate = {Saturday, March 01, 2003 at 17:15:48 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Maniatis02,
author = {S. I. Maniatis and E. G. Nikolouzou and I. S. Venieris},
title = {{QoS} Issues in the Converged {3G} Wireless and Wired Network},
journal = {{IEEE} Communications Magazine},
year = {2002},
volume = {40},
number = {8},
pages = {44--53},
month = {August},
annote = {This article discusses the mapping of the traffic classes offered by UMTS (i.e., conversational, streaming, interactive, and background), and the so-called RCL (Resource Control Layer) architecture which is a prototypical implementation of the next-generation Internet. Apart from proposing the mapping of the classes of the UMTS to the classes of RCL, the article also discusses a possible methodology to appropriately transform the QoS attributes in between the two architectures. Simulations performed with OPNET suggest that the mapping indeed achieves the end-to-end service differentiation between the traffic classes with only marginal interaction effects in between the classes.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Sunday, March 02, 2003 at 19:09:57 (CET)}
}
@article{Zheng01,
author = {Haitao Zheng, Jill Boyce},
title = {An Improved UDP Protocol for Video Transmission Over Internet-to-Wireless Networks},
journal = {IEEE Transactions on multimedia, vol 3, no 3, September 2001},
year = {2001},
volume = {3},
number = {3},
pages = {10},
month = {September},
annote = {This paper propose a new complete user datagram protocol (CUDP), which utilizes channel error information obtained from the physical and link layers to assist error recovery at the packet level. Theoretic and simulation results show that the video quality can be substantially improved by utilizing the frame error information at UDP and application layer.},
url = {/paper/cudp.pdf},
bibdate = {Wednesday, March 12, 2003 at 17:34:44 (CET)},
submitter = {Hannes Persson}
}
@article{Seth98,
author = {T. Seth and A. Broscius and C. Huitema and H-A P. Lin},
title = {Performance Requirements for Signaling in the {Internet} Telephony},
journal = {Internet Draft, IETF},
year = {1998},
month = {November},
annote = {To allow interoperability between the existing telephone network and Internet Telephony, it is necessary for the signaling performance to be comparable to that of the current standards to avoid introducing degradation in the service. This Internet Draft discusses the problem of providing high-quality signaling across an IP network. The emphasis of the Internet Draft is on the delay and packet-loss requirements of the ISUP and TCAP protocols. However, others are mentioned in passing.},
bibdate = {Monday, March 17, 2003 at 17:08:54 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Zhou00,
author = {D. Zhou},
title = {The Simulation of {TCAP} over {IP}},
journal = {Master's Thesis, North Carolina State University},
year = {2000},
annote = {This Masters Thesis simulates the performance of TCAP over IP using Ns2. Several issues are studied. Among other things the delay requirements of TCAP over IP; the delay of TCAP in DiffServ; and the delay of TCAP over IPSec are studied. The results of the simulations suggest that although IP degrades the performance of TCAP, the performance is still within the bounds of the PSTN standards. Further, it is shown that running TCAP atop IPSec does not impair the performance more than it is possible to use IPSec for security.},
url = {papers/Zhou00.doc.gz},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, March 17, 2003 at 17:33:54 (CET)}
}
@article{Brownlee02,
author = {N. Brownlee and K. Claffy},
title = {Understanding {Internet} Traffic Streams: Dragonflies and Tortoises},
journal = {{IEEE} Communications Magazine},
year = {2002},
volume = {40},
number = {10},
pages = {110--117},
month = {October},
annote = {This article presents a method of measuring the size and lifetime of Internet streams. The presented method is then used at two sites: the University of Auckland and the University of California at San Diego. Briefly, the results of the measurements at the two universities suggest that most streams are very short. At least 45\% of the streams have lifetimes of 2s or less, and about 98\% of the last less than 15 minutes. However, the remaining 1-2\% of the streams have lifetimes of hours to days and can carry a high proportion (50\%-60\%) of the total bytes on a link. Furthermore, it is shown that the size and lifetime of streams are independent dimensions, and therefore should be studied separately.},
bibdate = {Monday, March 24, 2003 at 09:06:56 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Camarillo,
author = {G. Camarillo and H. Schulzrinne and R. Kantola},
title = {A Transport Protocol for SIP},
journal = {Not known},
year = {Not known},
annote = {Current SIP implementations typically use UDP or TCP as their transport protocol. However, in cases with aggregate SIP signaling flows, it could be advantageous to use SCTP. This paper elaborates on the possible advantages of using SIP in these particular cases. Specifically, it studies how Head- Of-Line blocking in between different SIP sessions could be avoided by sending them over different streams, a common flow control per association (e.g., SACKs from one flow could trigger fast retransmits on another flow in the same association), and, of-course, multi-homing and load balancing.},
url = {papers/camarillo.pdf},
bibdate = {Thursday, March 27, 2003 at 10:33:14 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@article{Secondiani02,
author = {L. Secondiani and F. Mazzolini and N. Blefari-Melazzi and M. Femminella and L. Piacentini},
title = { *A Transport Interworking Protocol for the Support of TCP/IP based Applications over the ESW Satellite Network in the GMBS Environment},
journal = {IST Mobile & Wireless Telecommunications Summit 2002, Thessaloniki, Greece},
year = {2002},
month = {June},
annote = {Mentions some of the problems of TCP for satellites and discusses a split-connection approach targeted at ESW (EuroSkyWay) sattelites.},
url = {papers/Secondiani02_Sat_TCP_ESW.pdf},
bibdate = {Thursday, March 27, 2003 at 12:14:50 (CET)},
submitter = {Johan Garcia}
}
@article{Sanchez02,
author = {Sánchez, Rafael and Romero, Javier and Martinez, Julia and J\"{a}rvel\"{a}, Rauli},
title = {TCP/IP Performance over (E)GPRS network},
journal = {Proc. IEEE VTC Fall -2002, Vancouver, Canada},
year = {2002},
month = {September},
annote = {Simulation-based study of FTP and web traffic that suggests that large MSS and a initial window of at least two should be used. Abstract: The Performance of TCP/IP based applications over 2.5 and 3G networks presents problems due to long bandwidth-delay products, link errors and long delay variance. Specific characteristics of underlying technology like link level error recovery algorithms or quality of service procedures have also important influence on the behaviour of TCP layer and affect to the end to end performance. In this paper simulation results for EGPRS network are presented from an end-to-end dynamic simulator that models TCP Reno with some additional features, core network delays and buffering, GPRS RLC acknowledged mode and link level interference calculations. },
url = {papers/Sanchez02TCP_EGPRS.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, March 28, 2003 at 08:59:15 (CET)}
}
@inproceedings{Chan02,
author = {Mun Choon Chan and Ramachandran Ramjee},
title = {{TCP/IP} Performance over {3G} Wireless Links with Rate and Delay Variation},
booktitle = {Proceedings of the ACM Annual International Conference on Mobile Computing and Networking (MOBICOM)},
address = {Atlanta, USA},
year = {2002},
month = sep,
annote = {ABSTRACT: Wireless link losses result in poor TCP throughput since losses are perceived as congestion by TCP, resulting in source throttling. In order to mitigate this effect, 3G wireless link designers have augmented their system with extensive local retransmission mechanisms. In addition, in order to increase throughput, intelligent channel state based scheduling have also been introduced. While these mechanisms have reduced the impact of losses on TCP throughput and improved the channel utilization, these gains have come at the expense of increased delay and rate variability. In this paper, we comprehensively evaluate the impact of variable rate and variable delay on long-lived TCP performance. We propose a model to explain and predict TCP's throughput over a link with variable rate and/or delay. We also propose a network- based solution called Ack Regulator that mitigates the effect of variable rate and/or delay without significantly increasing the round trip time, while improving TCP performance by up to 40%},
url = {papers/Chan02_TCP_3G_varrate.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, March 28, 2003 at 09:22:46 (CET)}
}
@article{Jungmaier02,
author = {A. Jungmaier and E. Rathgeb and M. Tuexen},
title = {On the Use of {SCTP} in Failover Scenarios},
journal = {In Proceedings of the 6th World Multiconference on Systemics, Cybernetics and Informatics},
year = {2002},
pages = {363--368},
month = {July},
annote = {Redundancy in SS7 is accomplished through multiple links between signaling points. When a link failure occurs, MTP-L2 informs MTP-L3 about this. MTP-L3 recovers all messages from MTP-L2 which have not been successfully delivered, and resends them on an alternate link. This procedure is called "changeover" and is also implemented in SIGTRAN. This paper studies the changeover delay in a SIGTRAN network based on M2PA. Specifically, the paper studies the performance of the two principal ways of doing changeover, namely through multi-homed SCTP associations and through several single-homed SCTP associations managed by M2PA. It is shown that for both scenarios, the changeover time is below 800 ms which is the maximum time prescribed by ITU-T (Q.706). However, the scenario using a multi-homed SCTP association exhibited a smoother transition in that it kept the average delay per chunk during the changeover much lower than was the case in the single-homed scenario. Furthermore, the multi-homed scenario allowed for a sufficiently fast changeover, i.e., below 800 ms, for a wider range of parameter settings. Comment from Friday, April 25, 2003 at 14:37:14 (CEST):Currently, the trend in telephone signaling networks is towards IP, and then in particular towards SIGTRAN. To avoid a degraded availability, it is imperative that the failover mechanisms in SIGTRAN have comparable performance to those of traditional SS7. The contribution of this paper is a performance evaluation of the two level 2 failover mechanisms of SIGTRAN, namely multihomed SCTP associations and the built-in failover mechanism of M2PA. The study suggests that both failover mechanisms performs within the limit of ITU-T, i.e., 800 ms. Although, they are not on par with the changeover procedure of MTP-L2 in a traditional SS7 stack},
url = {papers/Jungmaier02.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, April 01, 2003 at 08:43:57 (CEST)}
}
@article{BSSGP00,
author = {ETSI},
title = {{Digital cellular telecommunications system (Phase 2+); General Packet Radio Service (GPRS); Base Station System (BSS) - Serving GPRS Support Node (SGSN); BSS GPRS Protocol (BSSGP) (GSM 08.18 version 6.7.1 Release 1997)})},
journal = {ETSI},
year = {2000},
url = {Loke},
bibdate = {Thursday, April 03, 2003 at 14:08:27 (CEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Ho99,
author = {Ho, J. and Zhu, Y. and Madhavapeddy, S.},
title = {Throughput and buffer analysis for GSM General Packet Radio Service {(GPRS)}},
journal = {IEEE Wireless Communications and Networks Conferance 1999 (WCNC'99)},
year = {1999},
volume = {3},
url = {papers/wcnc99_traffic_models.pdf},
bibdate = {Thursday, April 03, 2003 at 15:24:50 (CEST)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Lin03,
author = {C. Lin and D. Wei and S. H. Low and G. Buhrmaster and J. Bunn and D. H. Choe and R. L. A. Cottrel and J. C. Doyle and W. Feng and O. Martin and H. Newman and F. Paganini and S. Ravot and S. Singh},
title = {{FAST TCP:} From Theory to Experiments},
journal = {Submitted for publication in {IEEE} Communications Magazine},
year = {2003},
annote = {Many research areas require the communication of huge bulk data transfers. One such area is nuclear physics which more or less requires reliable data transfers at speeds of magnitude 1-10 Gbps. To address the communication needs of these research areas, and in particular to solve the problem of obtaininghigh sustainable throughput and bandwidth utilization forso-called long fatpipes, researchers at Caltech under the supervision of prof. S. H. Low have designed a variant of TCP called FAST, Fast AQM Scalable TCP. This paper presents some preliminary results with this transport protocol over a connection between Caltech and CERN, and compares the results with Linux. Among other things, it is shown that while FAST TCP obtained an average throughput of 925 Mbps and a bandwidth utilization of 95%, Linux only obtained an average throughput of 185 Mbps and a utilization of 19%. The paper gives no details on how the protocol works, however the main principle behind the protocol seems to be to view the TCP congestion control as a distributed, constrained optimization problem. The communication sources are viewed as utility functions and the links as link capacity constraints.},
url = {papers/Jin03.pdf},
bibdate = {Thursday, April 10, 2003 at 09:15:40 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{Rosenberg02,
author = {J. Rosenberg and H. Schulzrinne and G. Camarillo},
title = {The Stream Control Transmission Protocol as a Transport for for the Session Initiation Protocol},
journal = {www.ietf.org},
year = {2002},
pages = {8},
month = {July},
url = {http://www.ietf.org/internet-drafts/draft-ietf-sip-sctp-03.txt},
submitter = {Torbj\"{o}rn Andersson},
bibdate = {Monday, May 12, 2003 at 13:23:41 (CEST)}
}
@article{Wydrowski02,
author = {B. Wydrowski and M. Zukerman},
title = {{QoS} in Best-Effort Networks},
journal = {{IEEE} Communications Magazine},
year = {2002},
volume = {40},
number = {12},
pages = {44--49},
month = {December},
annote = {This article argues that the techniques to facilitate a QoS-aware, best-effort, packet-switched network is already available. In particular, it is argued by the authors of this article that the combination of utility-based source flow control toghether with non-backlog based active queueing (Remark: tail-drop and RED are examples of backlog-based queueing mechanisms while GREEN and REM are examples of non-backlog based queueing mechanisms) would indeed solve the bandwidth differentiation required by the majority of non-best-effort applications.},
submitter = {Karl-Johan Grinnemo},
bibdate = {Monday, May 12, 2003 at 18:57:08 (CEST)}
}
@article{Gozdecki03,
author = {Gozdecki, J. and Jajszczyk, A. and Stankiewicz, R.},
title = {Quality of service terminology in {IP} networks},
journal = {IEEE Communications Magazine},
year = {2003},
volume = {41},
number = {3},
pages = {153 -159},
month = {March},
annote = { This article provides an overview of commonly used terminology related to quality of service assurance in IP networks. Several approaches to QoS definition, including those of IETF, ITU, and ETSI, are presented and compared. Terms associated with QoS like class of service, grade of service, service level agreement, as well as service level specification (SLS), traffic conditioning agreement (TCA), and traffic conditioning specification (TCS) are discussed. Terminology used in two QoS architectures, IntServ and DiffServ, is also introduced. },
url = {papers/Gozdecki03_QoSterminology.pdf},
bibdate = {Friday, May 16, 2003 at 15:35:48 (CEST)},
submitter = {Johan Garcia}
}
@article{Vetro03,
author = {Vetro, A. and Christopoulos, C. and Huifang Sun},
title = {Video transcoding architectures and techniques: an overview},
journal = {IEEE Signal Processing Magazine},
year = {2003},
volume = {20},
number = {2},
pages = {18- 29},
month = {March},
annote = { Throughout this article, we concentrate on the transcoding of block-based video coding schemes that use hybrid discrete cosine transform (DCT) and motion compensation (MC). We first provide an overview of the techniques used for bit-rate reduction and the corresponding architectures that have been proposed. Then, we describe the advances regarding spatial and temporal resolution reduction techniques and architectures. Additionally, an overview of error resilient transcoding is also provided, as well as a discussion of scalable coding techniques and how they relate to video transcoding. Finally, the article ends with concluding remarks, including pointers to other works on video transcoding that have not been covered in this article, as well as some future directions. },
url = {papers/Vetro03_Vid_Transcoding.pdf},
submitter = {Johan Garcia},
bibdate = {Friday, May 16, 2003 at 16:43:35 (CEST)}
}
@inproceedings{Ladha03,
author = {S. Ladha and P.D. Amer and J. Iyengar and A. L. Caro},
title = {File Transfer in {FCS} Networks using Transport Layer Multistreaming},
booktitle = {MILCOM 2003},
year = {2003},
month = {April},
annote = {The way FTP is designed makes file transfer with FTP unnecessary inefficient. In particular, the traffic on the FTP control connection is typically periodic in nature and therefore remains in slow start at all times. Furthermore, when multiple files are transferred, the files are sent sequentially. An effect of this is that the cwnd is needlessly decreasing in between consecutive file transfers. This paper addresses this problem and proposes a solution which involves using SCTP instead of TCP. First, the multistreaming feature of SCTP is used to separate the control and data traffic. One stream is used exclusively for control traffic and one for data traffic. Second, pipelining is used to avoid the data stream to become idle in between two file transfers. The pipelining entails starting the file retrieval as soon as the request reaches the server, irrespective of whether a file transfer is already on going or not. Experiments conducted on a FreeBSD implementation of the modified FTP suggest that the transfer latency is cut by more than half. Furthermore, as the loss rate on the connection increases, the relative improvement in transfer latency increases.},
url = {papers/Ladha03.pdf},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Iyengar03,
author = {J. R. Iyengar and P. D. Amer and A. L. Caro and R. R. Stewart},
title = {Transport Layer Load Balancing in {FCS} Networks},
booktitle = {MILCOM 2003},
year = {2003},
month = {April},
annote = {One of the advantages with SCTP is that it provides network layer redundancy. However, this has been introduced at the cost of a more complex congestion control compared to TCP and thus more prone to errors. In this paper, one problem with the SCTP congestion control resulting from the possibility of performing changeover is studied. The problem causes the the sender to unnecessary slow down on the primary path and become too aggressive on the alternate path. Two solutions are presented: the Rhein algorithm, a successor to the Eifel algorithm, and the Changeover Aware Congestion Control (CACC) algorithm.},
url = {papers/Iyengar03.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Vahdat02,
author = {Amin Vahdat and Ken Yocum and Kevin Walsh and Priya Mahadevan and Dejan Kostic and Jeff Chase and David Becker},
title = {Scalability and Accuracy in a Large-Scale Network Emulator},
journal = {Proceedings of 5th Symposium on Operating Systems Design and Implementation (OSDI)},
year = {2002},
month = {December},
annote = {Describes the modelnet architecture where a core cluster of dummynet-based emulation servers are used to emulate a network with a more complex topology. Topologies from Internet traces, BGP dumps and topology generators can be used as input. These topologies can then be reduced to ease emulation hw requirements. Approach seems to be centered around a number of administrative scripts for setting up and some dummynet modifiactions to simplify use of multiple pipes. },
url = {papers/Vahdat02_Modelnet.pdf},
bibdate = {Friday, May 30, 2003 at 11:05:52 (CEST)},
submitter = {Johan Garcia}
}
@article{Kohler03,
author = {Eddie Kohler and Mark Handley and Sally Floyd and Jitendra Padhye},
title = {Datagram Congestion Control Protocol ({DCCP})},
journal = {Internet Draft, draft-ietf-dccp-spec-03.txt},
year = {2003},
month = {May},
annote = {Abstract: This document specifies the Datagram Congestion Control Protocol (DCCP), which implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. },
url = {papers/Kohler03_draft-ietf-dccp-03.txt},
submitter = {Johan Garcia},
bibdate = {Friday, May 30, 2003 at 17:06:22 (CEST)}
}
@article{Iyengar03b,
author = {J. R. Iyengar and A. L. Caro and P.D. Amer and G. J. Heinz},
title = {Making {SCTP} More Robust to Changeover},
journal = {Symposium on Performance Evaluation of Computer and Telecommunication Systems ({SPECTS} 2003) },
year = {2003},
month = {January},
annote = {This paper points out a problem with the congestion control algorithm of SCTP which reveals itself in some cases when an application-imposed changeover takes place. Although, only happening at times when the propagation delay of the alternate path is shorter than the primary path, it still demonstrates a deficiency in the path management of SCTP. Two algorithms which mitigate the problem are presented: C-CACC and SFR-CACC.},
url = {papers/Iyengar03b.pdf},
bibdate = {Sunday, June 01, 2003 at 19:22:57 (CEST)},
submitter = {Karl-Johan Grinnemo}
}
@article{dutta01active,
author = {D. Dutta and Y. Zhang},
title = {An Active Proxy Based Architecture for TCP in Heterogeneous Variable Bandwidth Networks},
journal = {IEEE GLOBECOM},
year = {2001},
url = {citeseer.nj.nec.com/dutta01active.html},
text = {D. Dutta and Y. Zhang, An Active Proxy Based Architecture for TCP in Heterogeneous Variable Bandwidth Networks, IEEE GLOBECOM, November 2001.}
}
@techreport{Caro03b,
author = {A. L. Caro and K. Shah and J. R. Iyengar and P. D. Amer and R. R. Stewart}
}
@inproceedings{Iyengar02,
author = {J. R. Iyengar and A. L. Caro and P. D. Amer and G. J. Heinz and R. R. Stewart},
title = {{STCP} Congestion Window Overgrowth During Changeover},
booktitle = {6th World Multiconference on Systemics, Cybernetics and Informatics ({SCI})},
year = {2002},
address = {Orlando, Florida, USA},
month = jul,
annote = {This paper reveals a problem in the way congestion control works in SCTP during changeover. The problem results in unnecessary retransmissions and TCP-unfriendly growth of the sender's congestion window during certain changeover conditions. The paper details the problem and suggest a solution to it - the Rhein algorithm. The Rhein algorithm enables SCTP to differentiate between transmissions and retransmissions. Although, solving the problem of TCP-unfriendly congestion window growth during changeover, it does not solve the problem of unnecessary retransmissions.},
url = {papers/Iyengar02.pdf},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Ravier01,
author = {T. Ravier and R. Brennan and T. Curran},
title = {Experimental Studies of {SCTP} Multihoming},
booktitle = {First Joint {IEI/IEE} Symposium on Telecommunications Systems Research},
year = {2001},
address = {Dublin, Ireland},
month = {nov},
annote = {The work presented in this paper is a first step in evaluating SCTP multihoming performance. Three scenarios are investigated using the SCTP reference implementation: multi-homed hosts with bandwidth limitation and delay but no packet loss, single-homed hosts with bandwidth limitation, delay and packet loss, and finally multi-homed hosts with bandwidth limitation, delay and packet loss. The scenario with multi-homed hosts and packet loss showed that the secondary path was only used for retransmitted packets, i.e., the SCTP reference implementation only uses multihoming for redundancy and not for load balancing.},
url = {papers/Ravier01.pdf},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Fu03,
author = {S. Fu and M. Atiquzzaman},
title = {Improving End-to-End Throughput of Mobile {IP} using {SCTP}},
booktitle = {Workshop on High Performance Switching and Routing},
year = {2003},
address = {Torino, Italy},
month = {june},
annote = {This paper demonstrates the improvement in throughput that can be achieved using SCTP as transport layer for Mobile IP. The performance of TCP Reno, TCP SACK and SCTP in the precense of handovers is evaluated. It is shown that SCTP benefits from its provision of a large number of SACK blocks than TCP SACK. The paper concludes that when the bottleneck bandwidth is low, SCTP can indeed improve the end-to-end throughput of mobile applications.},
url = {papers/Fu03.pdf},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Fu02,
author = {S. Fu and M. Atiquzzaman and W. Ivanic},
title = {Effect of Delay Spike on {SCTP}, {TCP Reno}, and {Eifel} in a Wireless Mobile Environment},
booktitle = {International Conference on Computer Communications and Networks ({IC3N})},
year = {2002},
address = {Miami, FL, USA},
month = {oct},
annote = {This paper evaluates and compares the impact of delay spikes (sudden increases of the RTT followed by equally sharp decreases) on the performance of SCTP, TCP Reno and Eifel under various wireless scenarios. Simulations with ns-2 suggest that in the presence of delay spikes and no packet losses, SCTP and TCP Reno have similar performance while Eifel has a higher performance. In the case of delay spikes with packet losses, the opposite is true: Eifel suffers from long transmission stalls and TCP Reno and SCTP have better performance than Eifel.},
url = {papers/Fu02.pdf},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Ye02,
author = {G. Ye and T. Saadawi and M. Lee},
title = {{SCTP} Congestion Control Performance in Wireless Multi-Hop Networks},
booktitle = {Military Communications Conference ({MILCOM})},
year = {2002},
address = {Anaheim, CA, USA},
month = {oct},
annote = {This paper evaluates the SCTP throughput performance over IEEE 802.11 using different receiver-side window sizes and different number of hops between the source and the destination. It seems as if the SCTP throughput degrades as the number of hops increases and that increasing the receiver-side window does not alleviate the situation. On the contrary, increasing the receiver-side window only amplifies the problem due to the so-called hidden-node and exposed-node problems. Packet loss may severly degrade the SCTP throughput performance. An algorithm is proposed to overcome this so-called small-window syndrome.},
url = {papers/Ye02.pdf},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Snoeren01,
author = {A. C. Snoeren and D. G. Andersen and H. Balakrishnan},
title = {Fine-Grained Failover Using Connection Migration},
booktitle = {3rd USENIX Symposium on Internet Technologies and Systems ({USITS})},
year = {2001},
address = {San Fransisco, CA, USA},
month = {mar},
annote = {Most replication technologies used in the Internet runs at the granularity of a connection. While this granularity is sufficient for most Web connections, it is not acceptable for long-running connections such as streaming media and IP telephony. This paper proposes a connection-level failover system. The system utilizes a transport-layer connection migration mechanism (a.k.a. stream mapper) and an application-layer, soft-state synchronization mechanism. Experimental results with a Linux prototype show that the performance of the failover system is not severly affected when the connection halts and resumption occurs everly few seconds.},
url = {papers/Snoeren01.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Labovitz01,
author = {C. Labovitz and A. Ahuja and A. Bose and F. Jahanian},
title = {Delayed Internet Routing Convergence},
journal = {{IEEE/ACM} Transactions on Networking},
year = {2001},
volume = {9},
number = {3},
pages = {293--306},
month = {jun},
annote = {Unlike PSTN networks, this paper shows that failover in Internet at present averages several minutes, and sometimes trigger fluctuations lasting as long as fifteen minutes. Furthermore, it is shown that the bound on the failover time is linear with the number of Autonomous Systems in the best case and exponential in the worst case.},
url = {papers/Labovitz01.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Shakkottai01,
author = {S. Shakkotai and A. Kumar and A. Karnik and A. Anvekar},
title = {{TCP} Performance Over End-to-End Rate Control and Stochastic Available Capacity},
journal = {{IEEE/ACM} Transactions on Networking},
year = {2001},
volume = {9},
number = {4},
pages = {377--391},
month = {aug},
annote = {This paper studies the performance of TCP over ATM/ABR. Specifically, the paper studies the effects of having a window-based congestion control on top of a rate-based dito. It is shown that the performance of TCP improves significantly if the network bottleneck bandwidth variations are slow compared to the round-trip delay.},
url = {papers/Shakkottai01.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Mansour01,
author = {Y. Mansour and B. Patt-Shamir},
title = {Jitter Control in QoS Networks},
journal = {{IEEE/ACM} Transactions on Networking},
year = {2001},
volume = {9},
number = {4},
pages = {492--502},
month = {aug},
annote = {This paper studies jitter control in networks with guaranteed QoS. Two types of jitter are studied: delay jitter, the maximum difference in the total delay of different packets, and rate jitter, which measures the difference in packet delivery rates at various times. For delay jitter, a simple algorithm using buffer space B is shown to guarantee the same delay jitter as a theoretical optimal algorithm using a buffer space of B/2. For rate jitter, and algorithm is presented which is shown to guarantee a rate jitter of B/h when a buffer size of 2B+h is used for any h>=1.},
url = {papers/Mansour01.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Li01,
author = {Q. Li and D. L. Mills},
title = {Jitter-Based Delay-Boundary Prediction of Wide-Area Networks},
journal = {{IEEE/ACM} Transactions on Networking},
year = {2001},
volume = {9},
number = {5},
pages = {578--590},
month = {oct},
annote = {In TCP, RTT estimation is conducted using a low-pass filter on RTT samples, a method adhering to a class of predictor processes called ARMA -- Auto-Regressive and Moving Average. This paper proposes a new predictor process based on a model of the RTT variations that captures its multistructure. Experiments showed that the proposed predictor outperforms Jacobson's algorithm by 60%.},
url = {papers/Li01.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Yang02,
author = {Y. Yang and H. Zhang and R. Kravets},
title = {Channel Quality Based Adaptation of TCP with Loss Discrimination},
journal = {IEEE Globecom, Taipei, Taiwan},
year = {2002},
month = {November},
annote = {Assumes the presence of loss differentiation and describes a scheme for congestion handling and retransmissions based on two weighted average measures: average packet loss rate (ALR) and temporary packet loss rate (TLR). NS simulations performed, but I feel several question marks about the approach...},
url = {papers/Yang02_CahnQualTCPadaptation.ps.gz },
submitter = {Johan Garcia},
bibdate = {Monday, August 11, 2003 at 11:07:06 (CEST)}
}
@inproceedings{Sarolahti02,
author = {Pasi Sarolahti and Alexey Kuznetsov},
title = {Congestion Control in {Linux TCP}},
booktitle = {Proceedings of USENIX 2002},
year = {2002},
pages = {49-62},
month = {June},
annote = {The design of the TCP implementation in Linux 2.4 is described in this paper. Focus is placed on congestion control and how the Linux implementation differs from conventional TCP implementations. The state machine for the TCP sender is explained. The Linux implementation supports standardized features (SACK, timestamps, and ECN), as well as more experimental features to enhance TCP performance (undoing window adjustments, D-SACK, FACK, congestion window validation, quick acks, rate-halving). The paper is concluded with a performance study of Linux TCP with and without quick acks, rate-halving and cwnd reverting enabled. In the setup used in the study quick acks, rate-halving and cwnd reverting increased performance.},
url = {papers/sarolahti.pdf},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Monday, August 11, 2003 at 11:39:57 (CEST)}
}
@article{Wennstrom02a,
author = {Wennstrom, A. and Brunstrom, A. and Rend\'{o}n, J. and Gustafsson, J.H.},
title = {A {GPRS} Testbed for {TCP} Measurements},
journal = {Proceedings 4th IEEE Conference on Mobile and Wireless Communications Networks (MWCN 2002), Stockholm, Sweden},
year = {2002},
month = {September},
url = {papers/Wennstrom02a_MWCN.pdf},
submitter = {Annika Wennstrom},
bibdate = {Thu Aug 14 15:31:16 CEST 2003}
}
@article{Smith95,
author = {D. E. Smith},
title = {Ensuring Robust Call Throughput and Fairness for {SCP} Overload Controls},
journal = {{IEEE/ACM} Transactions on Networking},
year = {1995},
volume = {3},
number = {5},
pages = {538--548},
month = {October},
annote = {This paper presents an analysis of two different types of Service Conbtrol Point (SCP) overload control algorithms: table-driven and adaptive. It is shown that even when the parameters are well matched to source characteristics, table-driven control do not perform as well as the adaptive control. In addition, the adaptive control is more robust to changes in traffic patterns.},
url = {papers/Smith95.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Thursday, August 21, 2003 at 08:19:57 (CEST)}
}
@article{Garcia02a,
author = {Johan Garcia},
title = {Application and Transport Layer Flexibility: An Image Transfer Example},
journal = {Karlstad University Studies 2002:11, Karlstad, Sweden},
year = {2002},
annote = {JG Lic thesis},
bibdate = {Friday, August 29, 2003 at 15:23:27 (CEST)},
submitter = {Johan Garcia}
}
@inproceedings{Kist02,
author = {A. Kist and R. J. Harris},
title = {A Simple Model for Calculating {SIP} Signalling Flows in 3GPP IP Multimedia Subsystems},
booktitle = {2nd IFIP-TC6 Networking Conference},
year = {2002},
address = {Pisa, Italy},
month = {May},
annote = {The 3rd Generation Partnership Project (3GPP) uses SIP as their signaling protocol in the IP Multimedia Subsystem for 3rd generation UMTS networks. To this end, this paper presents a methodology for calculating flows on SIP connections. The rationale for the paper was to provide an overall planning methodology to enable QoS for the signaling system of the UMTS IP Multimedia Subsystem.},
url = {papers/Kist02.pdf},
submitter = {Karl-Johan Grinnemo}
}
@article{Rumsewicz93,
author = {M. P. Rumsewicz},
title = {Analysis of the Effects of SS7 Message Discard Schemes on Call Completion Rates During Overload},
journal = {{IEEE/ACM} Transactions on Networking},
year = {1993},
volume = {1},
number = {4},
pages = {491--502},
month = {August},
annote = {This paper provides an analysis of call completion rates in SS7 networks during periods of MTP-L3 overload for a signaling transfer point (STP). The analysis shows that call completion is superior to message throughput, the traditional performance metric, for estimating the service offered to customers. Also, the analysis shows that message discard schemes result in extremely poor call completion performance, and to maintain reasonable customer server it is of necessity to have some form of feedback mechanism.},
url = {papers/Rumsewicz93.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, September 02, 2003 at 08:35:00 (CEST)}
}
@article{Alamgir02,
author = {R. Alamgir and M. Atiquzzaman and W. Ivancic},
title = {Effect of Congestion Control on the Performance of TCP and SCTP over Satellite Networks},
journal = {NASA Earth Science Technology Conference, Pasadena, CA},
year = {2002},
month = {June},
annote = {The paper examines SCTP and TCP over a Satelite link with 250ms delay and 1.5% packet error losses using ns nimulations. The results show that SCTP has some slight advantage due to to the different congestion control. In the experiments SCTP uses limited transmit whereas the tested TCP did not. Also the bytecounting of SCTP versus the segment counting of TCP helped SCTP.},
url = {papers/Alamgir02_SCTPoverSat.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, September 23, 2003 at 08:58:44 (CEST)}
}
@article{Atiquzzaman03,
author = {M. Atiquzzaman and W. Ivancic},
title = {Evaluation of SCTP Multistreaming over Satellite Links},
journal = {12th International Conference on Computer Communications and Networks, Dallas, TX,},
year = {2003},
month = {October},
annote = {Examines multistreaming of SCTP over satellite links. Both without errors and with 1,3, and 5% errors. The results show that for a constrained receiver buffer, SCTP performs better. Alternatively, the buffer can be dimensioned smaller. This is said to be useful for wireless handheld devices, although I do not find the overall argumentation totally convincing.},
url = {papers/Atiquzzaman03_SCTP_SatMultistream.pdf},
submitter = {Johan Garcia},
bibdate = {Tuesday, September 23, 2003 at 09:10:31 (CEST)}
}
@article{23107,
author = {3GPP},
title = {TS 23.107: {Q}uality of {S}ervice ({QoS}) concept and architecture},
journal = { },
year = {2003},
month = {September},
url = {papers/23107-5a0.doc},
bibdate = {Tuesday, October 07, 2003 at 21:16:56 (CEST)},
submitter = {Johan Garcia}
}
@article{Kohler03a,
author = {Eddie Kohler and Mark Handley and Sally Floyd and Jitendra Padhye},
title = {Datagram Congestion Control Protocol ({DCCP})},
journal = {draft-ietf-dccp-spec-04.txt, Work in progress},
year = {2003},
month = {June},
annote = {Abstract: This document specifies the Datagram Congestion Control Protocol (DCCP), which implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. },
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-spec-04.txt},
bibdate = {Tuesday, October 07, 2003 at 21:24:34 (CEST)},
submitter = {Johan Garcia}
}
@article{Kohler04a,
author = {Eddie Kohler and Mark Handley and Sally Floyd and Jitendra Padhye},
title = {Datagram Congestion Control Protocol ({DCCP})},
journal = {draft-ietf-dccp-spec-06.txt, Work in progress},
year = {2004},
month = {February},
annote = {Abstract: This document specifies the Datagram Congestion Control Protocol (DCCP), which implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. },
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-spec-04.txt},
bibdate = {Feb 29, 2004 at 21:24:34 (CEST)},
submitter = {Johan Garcia}
}
@article{Kohler04b,
author = {Sally Floyd and Eddie Kohler},
title = {Profile for {DCCP} Congestion Control ID 2: {TCP}-like Congestion Control},
journal = {draft-ietf-dccp-ccid2-05.txt, Work in progress},
year = {2004},
month = {February},
annote = { Abstract: This document contains the profile for Congestion Control Identifier 2, TCP-like Congestion Control, in the Datagram Congestion Control Protocol (DCCP) [DCCP]. DCCP implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. The TCP-like Congestion Control CCID is used by senders who are able to adapt to the abrupt changes in the congestion window typical of TCP's AIMD (Additive Increase Multiplicative Decrease) congestion control. TCP-like Congestion Control is particularly useful for senders who would like to take advantage of the available bandwidth in an environment with rapidly changing conditions.},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-ccid2-05.txt},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 07, 2003 at 21:27:02 (CEST)}
}
@article{Kohler03b,
author = {Sally Floyd and Eddie Kohler},
title = {Profile for {DCCP} Congestion Control ID 2: {TCP}-like Congestion Control},
journal = {draft-ietf-dccp-ccid2-03.txt, Work in progress},
year = {2004},
month = {February},
annote = { Abstract: This document contains the profile for Congestion Control Identifier 2, TCP-like Congestion Control, in the Datagram Congestion Control Protocol (DCCP) [DCCP]. DCCP implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. The TCP-like Congestion Control CCID is used by senders who are able to adapt to the abrupt changes in the congestion window typical of TCP's AIMD (Additive Increase Multiplicative Decrease) congestion control. TCP-like Congestion Control is particularly useful for senders who would like to take advantage of the available bandwidth in an environment with rapidly changing conditions.},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-ccid2-03.txt},
submitter = {Johan Garcia},
bibdate = {Feb 29, 2004 at 21:27:02 (CEST)}
}
@article{Kohler03c,
author = {Sally Floyd and Eddie Kohler and Jitendra Padhye},
title = {Profile for {DCCP} Congestion Control ID 3: {TFRC} Congestion Control},
journal = {draft-ietf-dccp-ccid3-03.txt, Work in progress},
year = {2003},
month = {June},
annote = { Abstract: This document contains the profile for Congestion Control Identifier 3, TCP-friendly rate control (TFRC), in the Datagram Congestion Control Protocol (DCCP). DCCP implements a congestion-controlled unreliable datagram flow suitable for use by applications such as streaming media. The TFRC CCID is used by applications that want a TCP-friendly send rate, possibly with Explicit Congestion Notification (ECN), while minimizing abrupt rate changes.},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-ccid2-03.txt},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 07, 2003 at 21:28:57 (CEST)}
}
@article{Kohler04c,
author = {Sally Floyd and Eddie Kohler and Jitendra Padhye},
title = {Profile for {DCCP} Congestion Control ID 3: {TFRC} Congestion Control},
journal = {draft-ietf-dccp-ccid3-05.txt, Work in progress},
year = {2004},
month = {February},
annote = { Abstract: This document contains the profile for Congestion Control Identifier 3, TCP-friendly rate control (TFRC), in the Datagram Congestion Control Protocol (DCCP). DCCP implements a congestion-controlled unreliable datagram flow suitable for use by applications such as streaming media. The TFRC CCID is used by applications that want a TCP-friendly send rate, possibly with Explicit Congestion Notification (ECN), while minimizing abrupt rate changes.},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-ccid2-05.txt},
submitter = {Johan Garcia},
bibdate = {Feb 29, 2004 at 21:28:57 (CEST)}
}
@article{Blanton03,
author = {E. Blanton and M. Allman and K. Fall and L. Wang},
title = {{RFC} 3517: A Conservative Selective Acknowledgment ({SACK})-based Loss Recovery Algorithm for {TCP}},
journal = { },
year = {2003},
month = {April},
annote = {Abstract: This document presents a conservative loss recovery algorithm for TCP that is based on the use of the selective acknowledgment (SACK) TCP option. The algorithm presented in this document conforms to the spirit of the current congestion control specification (RFC 2581), but allows TCP senders to recover more effectively when multiple segments are lost from a single flight of data.},
url = {http://www.faqs.org/rfcs/rfc3517.html},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 07, 2003 at 21:50:39 (CEST)}
}
@article{Handley03,
author = {M. Handley and S. Floyd and J. Padhye and J. Widmer},
title = {{RFC} 3448: {TCP} Friendly Rate Control ({TFRC)}: Protocol Specification},
journal = { },
year = {2003},
month = {January},
annote = {Abstract: This document specifies TCP-Friendly Rate Control (TFRC). TFRC is a congestion control mechanism for unicast flows operating in a best- effort Internet environment. It is reasonably fair when competing for bandwidth with TCP flows, but has a much lower variation of throughput over time compared with TCP, making it more suitable for applications such as telephony or streaming media where a relatively smooth sending rate is of importance.},
url = {http://www.faqs.org/rfcs/rfc3448.html},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 07, 2003 at 21:55:02 (CEST)}
}
@article{Floyd00a,
author = {Sally Floyd and Mark Handley and Jitendra Padhye and Joerg Widmer},
title = {Equation-Based Congestion Control for Unicast Applications: the Extended Version},
journal = { International Computer Science Institute tech report TR-00-03},
year = {2000},
month = {March},
annote = {Describes TFRC and provides simulation results and some numerical analysis.},
url = {papers/Floyd00aTFRCext.ps.gz},
submitter = {Johan Garcia},
bibdate = {Tuesday, October 07, 2003 at 23:19:21 (CEST)}
}
@book{Smirnov03,
title = {Quality of Future Internet Services},
publisher = {Springer},
year = 2003,
editor = {Michael Smirnov},
volume = 2856,
series = {LNCS},
annote = {The final report from COST action 263. Provides a state-of-the-art survey of technologies, algorithms, models, and experiments in the area of Internet QoS. The chapters included are: roadmap, traffic management, QoS Routing, Internet traffic engineering, mobile networking, algorithms for scalable content distribution, and pricing and QoS.}
}
@techreport{Garcia02b,
author = {Johan Garcia},
title = {{JPEG} Transcoding - Efficiency and Robustness Aspects},
institution = {Karlstad University Studies 2002:10, Karlstad, Sweden},
year = {2002},
annote = {Report covering the JPEG transcoding work.},
submitter = {Johan Garcia},
bibdate = {Friday, October 24, 2003 at 13:16:09 (CEST)}
}
@article{Sagfors03,
author = {S{\aa}gfors, M. and Ludwig, R. and Meyer, M. and Peisa, J.},
title = {Queue management for {TCP} traffic over {3G} links},
journal = {Proceedings of Wireless Communications and Networking (WCNC)},
year = {2003},
pages = {1663-1668},
month = mar,
annote = {An AQM scheme, Packet discard prevention counter (PDPC), for 3G is proposed and evaluated. Queues in 3G and other cellular networks differ from router queues in fixed networks. Data is buffered per terminal, which gives a low statistical multiplexing. High link utilization is more important, since radio resources are scarce. PDPC uses deterministic dropping (suitable for TCP) and avoids over-buffering. PDPC is compared to RED, drop-tail, and drop-front through simulation.},
url = {papers/sagfors_wcnc03.pdf},
bibdate = {Friday, November 14, 2003 at 15:09:16 (CET)},
submitter = {Annika Wennstr\"{o}m}
}
@inproceedings{Sagfors03a,
author = {S{\aa}gfors, M. and Ludwig, R. and Meyer, M. and Peisa, J.},
title = {Buffer management for rate-varying {3G} wireless links supporting {TCP} traffic},
booktitle={Proceedings of IEEE Vehicular Technology Conference (VTC-spring)},
year = {2003},
pages = {675-679},
month = apr,
annote = {In this paper the performance of PDPC, the AQM scheme presented in Sagfors03, is analyzed with respect to link rates and rate variations typical for a 3G link. The analysis is verified through simulation.},
url = {papers/sagfors_vtc_spring_03.pdf},
bibdate = {Friday, November 14, 2003 at 15:32:23 (CET)},
submitter = {Annika Wennstr\"{o}m}
}
@article{Chen02a,
author = {Wei-Yeh Chen and Jean-Lien C. Wu and Hung-Huan Liu},
title = {Performance Analysis of Radio Resource Allocation in {GSM/GPRS} Networks},
journal = {Vehicular Technology Conference, 2002. IEEE VTC'2002.},
year = {2002},
pages = {1461-1465},
month = {September},
annote = {Preemptive priority for voice calls is usually applied in GSM/GPRS, since it is important with a low blocking probability for voice calls. In this paper three alternatives are considered when GPRS data is preempted by voice calls: 1. no buffer for GPRS packets, 2. buffer only for preempted GPRS packets, and 3. buffer for all GPRS packets. An analytical model is developed and validated through simulation. Buffering of GPRS packets (2 and 3) reduces the blocking probability for GPRS. The authors suggest that 2 is sufficient even for real-time, since the queuing delay is rather low. },
url = {papers/chen02_preemption.pdf},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Friday, November 14, 2003 at 16:26:27 (CET)}
}
@article{Grinnemo03,
author = {Karl-Johan Grinnemo and Anna Brunstrom},
title = {Impact of SCTP-controlled Failovers for M3UA Users in a Dedicated SIGTRAN Network},
journal = {First Swedish National Computer Networking Workshop ({SNCNW2003})},
year = {2003},
month = {September},
annote = {Papper presented by Karl-Johan Grinnemo at the SNCNW2003 conference in Kista, Stockholm.},
url = {papers/Grinnemo03.pdf},
bibdate = {Monday, November 17, 2003 at 08:32:15 (CET)},
submitter = {Karl-Johan Grinnemo}
}
@inproceedings{Cavin02,
author = {D.Cavin and Y.Sasson and A.Schiper},
title = {On the accuracy of {MANET} simulators},
booktitle = {Proceedings of the Workshop on Principles of Mobile Computing (POMC'02) ACM},
year = {2002},
month = {October},
annote = {This paper illustrates how hard it is simulate wireless networks and achieve accurate results. Three popular network simulators (ns-2, opnet, GloMoSim) are compared. A simple flooding algorithm for ad hoc networks is implemented on top of of the MAC protocol in each of the simulators. The result is frightening; The simulators give different results, some of which are hardly comparable. A hybrid approach is recommended as an alternative to pure simulation. Lower layers could be simulated and higher layer protocols run on real machines. This seems strange to me, since one of the problems is to model the physical layer. However, the authors point out that more real experiments should be used to evaluate wireless protocols.},
url = {papers/cavin02accuracy.pdf},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Wednesday, November 19, 2003 at 12:25:46 (CET)}
}
@inproceedings{yu03,
author = {Liu Yu, Ye Min-hua, Zhang Hui-min},
title = {Improve TCP Performance over Wireless Link},
booktitle={Proceedings of the IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)},
year = {2003},
pages = {5},
annote = {This paper introduces a new scheme to improve TCP performance over Wireless links in the direction from the mobile host (MH) to the wired network. The scheme implements a TCP-agent at the base station (BS) to monitor (a) the segments related to the mobile host and (b) the corresponding ACKs from the fixed host (FH). When a segment is lost over the wireless link, the agent is capable of triggering a TCP fast retransmit on the MH by replicating and forwarding duplicate ACKs for the lost segment. Thus by invoking TCP fast retransmit, this scheme can avoid timeout retransmissions.},
bibdate = {Thursday, November 20, 2003 at 16:34:50 (CET)},
submitter = {Hannes Persson}
}
@inproceedings{ZZ03,
author = {Zhang Zhi, Qi Binghua, Zhang Ping},
title = {Low-Density Parity-Check Codes and High Spectral Efficiency Modulation},
booktitle={Proceedings of the IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)},
year = {2003},
pages = {5},
annote = {This paper investigates a coding and modulation scheme to improve transmission capacity in bandwidth limited channels. The scheme combines low-density parity-check (LDPC) codes with spectral efficient modulation. Recent advances in error-correcting codes have proven LDPC codes to posses good bit-error performance. At the transmitter the source bits are encoded to the codeword bits with a LDPC code and then converted into QAM symbols. Before the signal mapping a bit-by-bit interleaving is performed. At the receiver a soft output demapper first demodulates the QAM symbols. The basic idea is to demap the signals into soft bit values that have the same sign as a hard detector and whose absolute values indicate the reliability of the decision. In the LDPC decoder, who utilises the soft bits, a repeated decoding process of the LDPC code is undertaken. Simulation results over Gaussian and Rayleigh channels show that the proposed scheme attains good performance in a spectral efficiency sense. },
submitter = {Hannes Persson},
bibdate = {Thursday, November 20, 2003 at 16:37:43 (CET)}
}
@misc{rfc3550,
author="H. Schulzrinne and S. Casner and R. Frederick and V. Jacobson",
title={{RTP}: {A} Transport Protocol for Real-Time Applications},
series="Request for Comments",
number="3550",
howpublished="RFC 3550 (Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2003,
month=jul,
url="http://www.ietf.org/rfc/rfc3550.txt",
}
@misc{RFC2861,
author = {M. Handley and J. Padhye and S. Floyd},
title = {{TCP} Congestion Window Validation},
year = {2000},
month = {June},
annote = {In this experimental RFC the authors propose an extension to TCP's congestion control algorithm. After an idle period the cwnd may not reflect the congestion state in the network. Therefore a TCP sender should reduce the cwnd after an idle period (e.g. after a rwnd limited period).},
url = {papers/RFC2861.txt},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Friday, December 05, 2003 at 13:04:41 (CET)}
}
@inproceedings{Grinnemo04,
author = {K-J Grinnemo and A. Brunstrom},
title = {Performance of {SCTP}-controlled Failovers in {M3UA}-based {SIGTRAN} Networks},
booktitle = {Advanced Simulation Technologies Conference 2004 ({ASTC'04})},
year = {2004},
address = {Hyatt Regency Crystal City, Arlington, Virginia, USA},
month = {apr},
url = {papers/Grinnemo04.pdf}
}
@article{Grinnemo04e,
author = {Karl-Johan Grinnemo and Johan Garcia and Anna Brunstrom},
title = {Taxonomy and Survey of Retransmission Based Partially Reliable Transport Protocols},
year = 2004,
journal = {Computer Communications},
publisher = {Elsevier},
volume = 27,
number = 15,
pages = {1441-1452}
}
@inproceedings{Brennan01,
author = {R. Brennan and T. Curran},
title = {{SCTP} Congestion Control: Initial Simulation Studies},
booktitle = {17th International Teletraffic Congress},
year = {2001},
address = {Salvador, Brazil},
month = {December},
annote = {This paper uses OPNET simulations to study the behavior of the SCTP congestion control. Four different versions of SCTP were studied: A standard SCTP implementation which implements everything in the RFC 2960 marked as \"MUST\" or \"SHOULD\"; an SCTP version not implementing gap acks; an SCTP version which waits to reset the gap-ack counter until a TSN has been retransmitted, and thus prolongs the Fast Retransmit procedure; and finally, an SCTP version with improved detection of Fast Retransmit with faster invokation and which avoids false dupacks. The simulation study suggests that the ' performance of SCTP very much depends on the use of gap acks. To this end, the authors recommend gap-ack reports to become mandatory. Furthermore, the study recommends that retransmission during Fast Retransmit should be done irrespective of the size of the cwnd.},
url = {papers/Brennan01.pdf},
submitter = {Karl-Johan Grinnemo}
}
@misc{rfc3522,
author="R. Ludwig and M. Meyer",
title={The {Eifel} Detection Algorithm for {TCP}},
series="Request for Comments",
number="3522",
howpublished="RFC 3522 (Experimental)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2003,
month=apr,
url="http://www.ietf.org/rfc/rfc3522.txt",
annote = {This {RFC} defines the Eifel detection algorithm. This algorithm has been brought
forward in order to remedy the performance degradation that occurs in {TCP} due to
network anomalies such as packet reordering and delay spikes. In short, the Eifel
detection algorithm solves the retransmission ambiguity problem by using timestamps
(i.e., discerns between an ACK of an original packet and an ACK of a retransmitted
packet): If the timestamp of the first acceptable ACK (an ACK that acknowledges
previously unsent data) that arrives after a retransmission is smaller than the
timestamp of the retransmitted packet, then that ACK must have been sent in response
to an original packet. Thus, the {TCP} sender must have entered loss recovery
unnecessarily.},
}
@article{Ludwig03,
author = {R. Ludwig and A. Gurtov},
title = {The {Eifel} Response Algorithm for {TCP}},
journal = {Work in progress, {IETF}},
year = {2003},
annote = {This Internet draft comprises a complement to RFC 3522. In particular, the
Eifel algorithm as originally described by Reiner Ludwig has been partitioned
into two separate algorithms: a detection algorithm which is described in RFC
3522, and a response algorithm. The reason to this separation is that IETF
requires the detection algorithm as described in RFC 3522, but permits variations
in the response algorithm. The response algorithm described in this Internet
draft is pretty straighforward and entails that the state of the cwnd and sstresh
variables are reset to the values before the retransmission.},
url = {papers/draft-ietf-tsvwg-tcp-eifel-response-04.txt},
submitter = {Karl-Johan Grinnemo},
bibdate = {Tuesday, February 24, 2004 at 21:27:25 (CEST)}
}
@unpublished{Ladha03b,
author = "S. Ladha and S. Baucke and R. Ludwig",
title = "On Making {SCTP} Robust to Network Anomalies",
year = "2003",
note = "Submitted to {ACM} Computer Communication Review",
submitter = "Karl-Johan Grinnemo",
url = "papers/Ladha03b.pdf",
annote = "Network anomalies such as packet reordering and delay
spikes can result in spurious retransmissions and hurt
performance of SCTP. This paper shows that spurious
timeouts in SCTP with multihomed endpoints may result
in a problem of congestion window overgrowth on the
alternate destination (i.e., the sender is increasing
the congestion window of the alternate path on receiving
ACKs for packets sent on the primary path). Furthermore,
this paper proposes an extension to the Eifel algorithm
for SCTP, and shows, through ns2 simulations, that the
Eifel extension significantly improves the SCTP throughput
during reordering events. As a matter of fact, already at
a reordering rate of 1\%, SCTP with Eifel increases throughput
with 60\% compared to standard SCTP."
}
@Article{Braden03,
author = {Robert Braden and Ted Faber and Mark Handley},
title = {From protocol stack to protocol heap: role-based architecture},
journal = {Computer Communication Review},
volume = {33},
number = {1},
pages = {17-22},
month = {January},
year = {2003},
bibdate = {Monday, February 23, 2004 at 15:55:50 (CET)},
url = {papers/Braden03_Protocolheap.pdf},
annote = {Discusses an alternative to the traditional layering protocol orgaization. Partly motivated by the need for cross-layer information flow.
Abstract: Questioning whether layering is still an adequate foundation for networking architectures, this paper investigates non-layered approaches to the design and implementation of network protocols. The goals are greater flexibility and control with fewer feature interaction problems. The paper further proposes a specific non-layered paradigm called role-based architecture.
},
submitter = {Johan Garcia}
}
@Article{Tscudin03,
author = {Christian Tschudin and Richard Gold},
title = { Network pointers},
journal = {Computer Communication Review},
volume = {33},
number = {1},
pages = {23-28},
month = {January},
year = {2003},
bibdate = {Monday, February 23, 2004 at 16:02:57 (CET)},
url = {papers/Tscudin03_NetworkPointers.pdf},
annote = {Suggests the use of network pointers maikng it possible to do packet processing based on information contained in multiple layers. An abstraction making use of multiple-layer information.
Abstract:The Internet architecture can be characterized as having a rather coarse grained and imperative style of network packet handling: confronted with an IP packet and its source and destination addresses, the infrastructure almost blindly and unalterably executes hundreds of resolution, routing and forwarding decisions. There are numerous attempts that try to "extend" the Internet in order to either reduce the immediate impact an arbitrary packet can have (e.g., NAT), or to insert diversions from the normal processing paths in order to better use the existing resources (e.g., content delivery). In this paper we argue that we need a more fine grained control, in the hands of end nodes, over how packets are handled. The basic abstraction presented here is that of networking pointers, which we show to relate to low level concepts like ARP caches, but also high level routing decisions for terminal mobility, content delivery networks, or peer-to-peer overlay forming. We report on first implementation experiences of an "underlay" networking approach which uses pointer tricks underneath IP in order to provide new network layer services.},
submitter = {Johan Garcia}
}
@Article{Sarolahti03,
author = {Pasi Sarolahti and Markku Kojo and Kimmo Raatikainen},
title = {F-RTO: an enhanced recovery algorithm for TCP retransmission timeouts},
journal = {Computer Communication Review},
volume = {33},
number = {2},
pages = {51-63},
month = {April},
year = {2003},
bibdate = {Monday, February 23, 2004 at 17:14:29 (CET)},
url = {papers/Sarolahti03_FRTOforTCP.pdf},
annote = {Presents F-RTO which has the same objective as eifel and DSACK, namely to avoid retransmissions after spurious timeouts. Seems to work generally well, some later detection than eifel in some cases, but still earlier than dsack.
---Abstract:Spurious TCP retransmission timeouts (RTOs) have been reported to be a problem on network paths involving links that are prone to sudden delays due to various reasons. Especially many wireless network technologies contain such links. Spurious retransmission timeouts often cause unnecessary retransmission of several segments, which is harmful for TCP performance. Recent proposals for avoiding unnecessary retransmissions after a spurious RTO require use of TCP options which must be implemented and enabled at both ends of teh connection. We introduce a new TCP sender algorithm for recovery after a retransmission timeout and show that unnecessary retransmissions after a spurious retransmission timeout, improving the TCP performance considerably. The algorithm is friendly towards other TCP connections, because it follows the congestion control principles and injects packets to the network at same rate as a conventional TCP sender. We implemented the algorithm and compared its performance to conventional TCP and Eifel TCP when RTOs occurred either due to sudden delays or due to packet losses. The results show that our algorithm either improves performance or gives similar througput as the other TCP variants evaluated in different test cases.},
submitter = {Johan Garcia}
}
@Article{Khalifa04,
author = {I. Khalifa and L. Trajkovic},
title = {An overview and comparison of analytical TCP models},
journal = {Proc. IEEE Int. Symp. Circuits and Systems, Vancouver, British Columbia},
month = {May},
year = {2004},
bibdate = {Tuesday, February 24, 2004 at 09:50:37 (CET)},
url = {papers/Khalifa04_TCPmodels.pdf},
annote = {Provides a short overview (4pg) of six different TCP modelling efforts and discusses model evaluation and validation.},
submitter = {Johan Garcia}
}
@Article{Wu99,
author = {G. Wu and Y. Bai and J. Lai and A. Ogielski},
title = {Interactions between TCP and RLP in wireless Internet},
journal = {Proc GLOBECOM '99, Rio de Janeireo, Brazil },
volume = {1B},
pages = {661-666},
month = {December},
year = {1999},
bibdate = {Tuesday, February 24, 2004 at 10:17:06 (CET)},
url = {papers/Wu99_TCP_RLP_ISP.pdf},
annote = {Interseting since it mentions interlayer signaliing pipe. Not much detail on this though.
Abstract:
The Internet is implemented on the basis of the ISO-OSI hierarchy architecture where the protocols for different layers are independent of each other. For wireless Internet, however, information of other layers may be required in order to improve the overall system performance. We propose a new protocol stack including an interlayer signaling pipe (ISP) across layers for the wireless Internet access scenario. We investigate the interactions of two distinct loss recovery mechanisms employed by TCP and RLP (radio link protocol) during a low-speed TCP connection over a radio channel with correlated losses to show the necessity of the ISP. It is shown by simulations that the proposed ISP can be used to support such a TCP-RLP coordination mechanism
},
submitter = {Johan Garcia}
}
@Article{Tsaoussidis99,
author = {Vassilios Tsaoussidis and Songbin Wei },
title = {Reliability/Throughput/Jitter Tradeoffs for Real-Time Transport Protocols},
journal = {Proc 20th IEEE Real Time Systems Symposium, Phoenix, Arizona},
month = {December},
year = {1999},
bibdate = {Tuesday, February 24, 2004 at 11:46:42 (CET)},
url = {papers/Tsaoussidis99_Reliability_tradeoffs.pdf},
annote = {Interesting since it suggest the MTP partially reliable protocol. THey did not seem to continue with this work, although Tsoussidis continued with some interesting work on wireless & wave and wait (WWP).
---Abstract: Reliability, Throughput and Jitter are Quality of Service parameters of significance for multimedia applications. Although each packet retransmission has a certain cost to the overall throughput performance, each missing packet does not have the same impact on the quality of multimedia applications. In this paper we report preliminary results on the impact of retransmission-based error recovery on multimedia data transmission over IP, using an experimental Multimedia Transmission Protocol (MTP) ... },
submitter = {Johan Garcia}
}
@Article{Dong04,
author = {Hui Dong and Ian D. Chakeres and Allen Gersho and Elizabeth M. Belding-Royer and J. D. Gibson},
title = {Selective Bit-error Checking at the {MAC} layer for Voice Over Mobile Ad hoc Networks with {IEEE} 802.11},
journal = {Proc. of the IEEE Wireless Communications and Networking Conference (WCNC), Atlanta, Georgia},
month = {March},
year = {2004},
bibdate = {Sunday, February 29, 2004 at 23:11:46 (CET)},
url = {papers/Dong04_SelErrMAC80211.pdf},
annote = {This paper proposes a modified MAC sheme for 802.11 that only checksums parts of the data, in this case the first 75 bits of the NB-AMR voice codec frame. Simulation results show better performance for packet loss rate, delay and subjective voice quality.},
submitter = {Johan Garcia}
}
@misc{rfc3366,
author="G. Fairhurst and L. Wood",
title={Advice to link designers on link Automatic Repeat reQuest ({ARQ})},
series="Request for Comments",
number="3366",
howpublished="RFC 3366 (Best Current Practice)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2002,
month=aug,
url="http://www.ietf.org/rfc/rfc3366.txt",
}
@misc{ECSE,
title = {{National Research Council (USA), Academic Careers for Experimental Computer Scientists and Engineers}},
organization = {National Research Council (USA)},
howpublished = {National Academy Press},
address = {Washington, D.C.},
year = {1994},
url = {http://www.nap.edu/html/acesc/},
submitter = {Annika Wennstr\"{o}m},
bibdate = {Fri Mar 19 11:04:57 CET 2004}
}
@misc{rfc3390,
author="M. Allman and S. Floyd and C. Partridge",
title={Increasing {TCP}'s Initial Window},
series="Request for Comments",
number="3390",
howpublished="RFC 3390 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2002,
month=oct,
url="http://www.ietf.org/rfc/rfc3390.txt",
}
@article{Mishra03,
author = {Arunesh Mishra and Minho Shin and William Arbaugh},
title = {An empirical analysis of the {IEEE 802.11 MAC} layer handoff process},
journal = {ACM Computer Communication Review},
volume = {33},
number = {2},
year = {2003},
issn = {0146-4833},
pages = {93--102},
publisher = {ACM Press},
url = {papers/Mishra_Shin_Arbaugh.pdf},
submitter = {Annika Wennstr\"{o}m},
bibdate = "Wed Apr 7 10:34:32 CEST 2004"
}
@Article{Gurtov04,
author = {Andrei Gurtov and Sally Floyd},
title = {Modeling Wireless Links for Transport Protocols},
journal = {To appear in ACM Computer Communications Review},
year = {2004},
bibdate = {Wednesday, April 07, 2004 at 11:07:35 (CEST)},
url = {papers/gurtov04_modelling.pdf},
submitter = {Annika Wennstr\"{o}m}
}
@book{Lin00,
author = {Yi-Bing Lin and Imrich Chlamtac},
title = {Wireless and Mobile Network Architectures},
year = {2000},
isbn = {0471394920},
publisher = {John Wiley \& Sons, Inc.},
url = {dvbiblioteket},
submitter = {Annika Wennstr\"{o}m},
bibdate = "Wed Apr 7 10:34:32 CEST 2004"
}
@Article{Meyer03,
author = {Meyer, M. and Sachs, J. and Holzke, M.},
title = {Performance Evaluation of a {TCP} Proxy in {WCDMA} Networks},
journal = {IEEE Wireless Communications},
volume = {10},
number = {5},
pages = {70--79},
month = {October},
year = {2003},
bibdate = {Tuesday, April 13, 2004 at 11:50:10 (CEST)},
url = {papers/Meyer03_tcp_wcdma.pdf},
annote = {
},
submitter = {Annika Wennstr\"{o}m}
}
@book{Stallings01a,
author = {William Stallings},
title = {Wireless Communications and Networks},
year = {2001},
isbn = {0130408646},
publisher = {Prentice Hall},
url = {dvbiblioteket},
submitter = {Annika Wennstr\"{o}m},
bibdate = "Tue Apr 20 09:51:27 CEST 2004"
}
@Article{Gerla04,
author = {M. Gerla and B. K. F. Ng and M. Y. Sanadidi and M. Valla and R. Wang},
title = {TCP Westwood with adaptive bandwidth estimation to improve efficiency/friendliness tradeoffs},
journal = {Computer Communications},
volume = {27},
number = {1},
pages = {41-58},
month = {January},
year = {2004},
bibdate = {Thursday, June 10, 2004 at 12:35:50 (CEST)},
url = {papers/Gerla04_TCPWestwoodFairness.pdf},
annote = {Interesting since it discusses fairness metrics for the case when one flow using TCPW is working better than some competing regular TCP flow that are restricted by biterrors. Relevant for JG ICETE paper.
Abstract:In this paper, we propose an extension of Transmission Control Protocol (TCP) Westwood allowing the management of the Efficiency/Friendliness-to-NewReno tradeoffs. We show that the extended TCP Westwood is able to achieve higher total link utilization, yet at the same time maintain friendliness. TCP Westwood (for short, TCPW) implements a novel window congestion control algorithm based on eligible rate estimation (RE). The performance of TCPW has been promising, exceeding that of TCP NewReno in ‘large leaky pipes’; i.e. network paths with high bandwidth-delay product and non-negligible random error rate. Consider the situation where TCPW and TCP NewReno connections coexist and share common bottlenecks. Friendliness in this shared environment is paramount. Under certain conditions TCP NewReno may experience some performance degradation since TCPW ‘learns’ more about connection performance and thus can take better advantage of available bandwidth. To manage the efficiency/friendliness tradeoffs, we propose to combine the original TCPW Bandwidth Estimation (BE) strategy with a new RE strategy. One finds that BE provides significantly higher utilization, but may, under certain conditions, overestimates a connection fair share. RE, on the other hand, tends to be closer to the achieved rate of a connection, but it may underestimate the connection fair share. The question is: which estimate—RE or BE—yields better throughput/friendliness tradeoffs? Our studies show that RE works best when packet loss is mostly due to congestion. If, on the other hand, packet loss is mostly due to link errors, BE gives better performance. To achieve the ‘best of all worlds’, we introduce a method we call Combined Rate and Bandwidth estimation (CRB.) A connection first infers the predominant cause of packet loss (buffer congestion or random error) and then uses the more appropriate estimation method. Simulation shows that the adaptive CRB provides a very effective compromise between efficiency and friendliness.},
submitter = {Johan Garcia}
}
@Article{Zhang04,
author = {Chi Zhang and Vassilis Tsaoussidis},
title = {Error differentiation with measurements based on wave patterns},
journal = {Computer Communications},
volume = {27},
number = {10},
pages = { 989-1000},
month = {June},
year = {2004},
bibdate = {Thursday, June 10, 2004 at 15:18:20 (CEST)},
url = {papers/Zhang04_TCPRealErrorDiff.pdf},
annote = {Abstract: We investigate the efficiency and fairness potential of an error differentiation mechanism with measurements based on wave patterns. Our work is focused on a receiver-oriented protocol, TCP-Real that allows for measurement-based congestion detection and error recovery instead of a ‘blind’ increase/decrease window adjustment. We conduct a comparative evaluation of congestion control mechanisms of standard TCP, TCP-Real, and TCP-friendly protocols in multiplexed wired/wireless networks for both delay-tolerant and delay-sensitive applications. We show that the receiver-oriented communication of TCP-Real (i) diminishes the impact of reverse path asymmetry on the transmission rate, (ii) allows for error recovery tactics that are responsive to the nature of the error detected (i.e. congestion versus bit corruption), and (iii) dictates transmission rate adjustments that conform to the level of present contention. As a direct impact of this design, we realize the inherent properties of our protocol to produce comprehensive dynamics in heterogeneous environments, in contrast to the occasionally limited efficiency of standard TCP and the application-specific design of TCP-friendly protocols.},
submitter = {Johan Garcia}
}
@Article{Beritelli03,
author = {F. Beritelli and G. Ruggeri and G. Schembra},
title = {{TCP}-friendly transmission of voice over {IP}},
journal = {European Transactions on Telecommunications},
volume = {14},
number = {3},
pages = {193 - 203},
month = {July},
year = {2003},
bibdate = {Thursday, June 10, 2004 at 15:22:45 (CEST)},
url = {papers/Beritelli03_TCPfriendlyVoIPTFRC.pdf},
annote = {Abstract:In the last few years an increasing amount of attention has been paid to technologies for the transmission of voice over IP (VoIP). At present, the UDP transport protocol is used to provide this service. However, when the same bottleneck link is shared with TCP flows, and in the presence of a high network load and congestion, UDP sources capture most of the bandwidth, strongly penalizing TCP sources. To solve this problem some congestion control should be introduced for UDP traffic as well, in such a way that this traffic becomes TCP-friendly. In this perspective, several TCP-friendly algorithms have been proposed in the literature. Among them, the most promising candidates for the immediate future are RAP and TFRC. However, although these algorithms were introduced to support real-time applications on the Internet, up to now the only target in optimizing them has been that of achieving fairness with TCP flows in the network. No attention has been paid to the applications using them, and in particular, to the quality of service (QoS) perceived by their users. The target of this paper is to analyze the problem of transmitting voice over IP when voice sources use one of these TCP-friendly algorithms. With this aim, a VoIP system architecture is introduced and the characteristics of each its elements are discussed. To optimize the system, a multirate voice encoder is used so as to be feasible to work over a TCP layer, and a modification of both RAP and TFRC is proposed. Finally, in order to analyze the performance of the proposed system architecture and to compare the modified RAP and TFRC with the original algorithms, the sources have been modeled with an arrival process modulated by a Markov chain, and the model has been used to generate traffic in a simulation study performed with the ns-2 network simulator.},
submitter = {Johan Garcia}
}
@Article{Buchholcz03,
author = {Gerg\"{o} Buchholcz and Adam Gricser and Thomas Ziegler and Tien Van Do},
title = {Explicit Loss Notification to Improve {TCP} Performance over Wireless Networks},
journal = {Lecture Notes in Computer Science},
volume = {2720},
pages = { 481 - 492},
month = {June},
year = {2003},
bibdate = {Thursday, June 10, 2004 at 16:03:22 (CEST)},
url = {papers/Buchholcz03_TCPLossDiff.pdf},
annote = {Discusses the applicatbility of checksum-based loss differentiation and does some simultiaons that show considerable improvement of throughput.
Abstract:In this paper we propose a novel TCP congestion control algorithm to overcome the weakness of the original TCP mechanism in wireless environments. The primary aim of the new algorithm is to cope with packet losses due to bit errors in the radio interface. Our solution is based on the idea of Explicit Loss Notification (ELN) to notify the sender about packet losses in the wireless channel. We also performed extensive simulations with different kind of traffic and error models to demonstrate the performance improvement of the proposed algorithm compared to the original TCP.},
submitter = {Johan Garcia}
}
@Article{Hacker02,
author = {Thomas J. Hacker and Brian D. Noble and Brian D. Athey},
title = {The effects of systemic packet loss on aggregate {TCP} flows},
journal = {Proceedings of the 2002 ACM/IEEE conference on Supercomputing, Baltimore, Maryland},
pages = {1--15},
month = {November},
year = {2002},
bibdate = {Thursday, June 10, 2004 at 16:52:38 (CEST)},
url = {papers/Hacker02_NoncongestionTCPlossEffects.pdf},
annote = {Abstract:The use of parallel TCP connections to increase throughput for bulk transfers is common practice within the high performance computing community. However, the effectiveness, fairness, and efficiency of data transfers across parallel connections is unclear. This paper considers the impact of systemic non-congestion related packet loss on the effectiveness, fairness, and efficiency of parallel TCP transmissions. The results indicate that parallel connections are effective at increasing aggregate throughput, and increase the overall efficiency of the network bottleneck. In the presence of congestion related losses, parallel flows steal bandwidth from other single stream flows. A simple modification is presented that reduces the fairness problems when congestion is present, but retains effectiveness and efficiency.},
submitter = {Johan Garcia}
}
@Article{Cen03,
author = { Song Cen and Pamela C. Cosman and Geoffrey M. Voelker},
title = { End-to-end differentiation of congestion and wireless losses},
journal = { IEEE/ACM Transactions on Networking},
volume = {11},
number = {5},
pages = {703-717},
month = {October},
year = {2003},
bibdate = {Thursday, June 10, 2004 at 17:12:29 (CEST)},
url = {papers/Cen03_LossDiff.pdf},
annote = {This is an improved/extended version of the Cen02 paper evaluating a number of non-checksum receiver based LDAs different
Abstract:In this paper, we explore end-to-end loss differentiation algorithms (LDAs) for use with congestion-sensitive video transport protocols for networks with either backbone or last-hop wireless links. As our basic video transport protocol, we use UDP in conjunction with a congestion control mechanism extended with an LDA. For congestion control, we use the TCP-Friendly Rate Control (TFRC) algorithm. We extend TFRC to use an LDA when a connection uses at least one wireless link in the path between the sender and receiver. We then evaluate various LDAs under different wireless network topologies, competing traffic, and fairness scenarios to determine their effectiveness. In addition to evaluating LDAs derived from previous work, we also propose and evaluate a new LDA, ZigZag, and a hybrid LDA, ZBS, that selects among base LDAs depending upon observed network conditions.We evaluate these LDAs via simulation, and find that no single base algorithm performs well across all topologies and competition. However, the hybrid algorithm performs well across topologies and competition, and in some cases exceeds the performance of the best base LDA for a given scenario. All of the LDAs are reasonably fair when competing with TCP, and their fairness among flows using the same LDA depends on the network topology. In general, ZigZag and the hybrid algorithm are the fairest among all LDAs.},
submitter = {Johan Garcia}
}
@Article{Mascolo04,
author = {S. Mascolo and L. A. Grieco and R. Ferorelli and P. Camarda and G. Piscitelli},
title = {Performance evaluation of {Westwood+} {TCP} congestion control},
journal = {Performance Evaluation },
volume = {55},
number = {1-2},
pages = {93-111 },
month = {January},
year = {2004},
bibdate = {Thursday, June 10, 2004 at 20:07:46 (CEST)},
url = {papers/Mascolo04_WestwoodPlusEval.pdf},
annote = {This paper describes internet experiments, emulation and ns2 simulation evaluation of westwood+. Abstract:Westwood+ TCP is a sender-side only modification of the classic Tahoe/Reno TCP that has been recently proposed to improve fairness and efficiency of TCP. The key idea of Westwood+ TCP is to perform an end-to-end estimate of the bandwidth available for a TCP connection by properly counting and filtering the stream of ACK packets. This estimate is used to adaptively decrease the congestion window and slow-start threshold after a congestion episode. In this way, Westwood+ TCP substitutes the classic multiplicative decrease paradigm with the adaptive decrease paradigm. In this paper we report experimental results that have been obtained running Linux 2.2.20 implementations of Westwood+, Westwood and Reno TCP to ftp data over an emulated WAN and over Internet connections spanning continental and intercontinental distances. In particular, collected measurements show that the bandwidth estimation algorithm employed by Westwood+ nicely tracks the available bandwidth, whereas the TCP Westwood bandwidth estimation algorithm greatly overestimates the available bandwidth because of ACK compression. Live Internet measurements also show that Westwood+ TCP improves the goodput w.r.t. TCP Reno. Finally, computer simulations using ns-2 have been developed to test Westwood, Westwood+ and Reno in controlled scenarios. These simulations show that Westwood+ improves fairness and goodput w.r.t. Reno.},
submitter = {Johan Garcia}
}
@InProceedings{Liu04,
author = {Qingwen Liu, Shengli Zhou, Georgios Giannakis},
title = {TCP Performance in Wireless Access with Adaptive Modulation and Coding},
year = {2004},
location = {ICC 2004},
submitter = {Stefan Alfredsson},
bibdate = {August 30, 2004, 2:30 pm}
}
@InProceedings{Sachs03,
author = {Sachs, J.},
title = {A generic link layer for future generation wireless networking},
year = {2003},
address = {Paris, France},
organization = {Proceedings of the IEEE International Conference on Communication (ICC)},
month = {May},
submitter = {Annika Wennström},
bibdate = {September 9, 2004, 2:33 pm},
annote = {This paper proposes the concept of a generic link layer (GLL) for wireless networks.GLL would enable efficient cooperation between systems with different radio access techniques, such as efficient and lossless inter-system handover.}
}
@MastersThesis{Berlin02,
author = {M. Berlin},
title = {Channel Modeling and Simulation for the Mobile Internet},
year = {2002},
school = {Uppsala University},
submitter = {Stefan Alfredsson},
URL = {http://www.signal.uu.se/Courses/Semabstracts/s0207.html},
bibdate = {September 10, 2004, 12:16 am}
}
@inproceedings{Sternad02,
author = {Mikael Sternad},
title = {The Wireless {IP} Project},
year = {2002},
address = {Stockholm, Sweden},
booktitle = {Proceedings of {Radiovetenskap och Kommunikation}},
month = {June},
URL = {http://www.signal.uu.se/Publications/pdf/c0211.pdf},
bibdate = {September 10, 2004, 12:23 am},
submitter = {Stefan Alfredsson}
}
@Article{Widmer04,
author = {Jörg Widmer and Catherine Boutremans and Jean-Yves Le Boudec },
title = {End-to-end congestion control for TCP-friendly flows with variable packet size},
journal = {ACM SIGCOMM Computer Communication Review},
year = {2004},
number = {2},
volume = {34},
month = {April},
submitter = {Johan Garcia},
URL = {../papers/Widmer04_e2eCongTFRCvwrSize.pdf},
bibdate = {September 26, 2004, 7:50 pm},
annote = {Good paper that highlights the packet size unfairness problem. Suggest some solutions to the overagressiveness problems that appears for the simple fix of setting s to MSS. Only evlauated for a single network setup. The severity of the problem probably vary according to the delay and bw and # of competeing flows etc.
Mentions the fundametal problem of Bytemode RED for smaller than MSS flows.}
}
@misc{RFC3714,
author = {S. Floyd and J. Kempf},
title = {{RFC 3714}: {IAB} Concerns Regarding Congestion Control for Voice Traffic in the {I}nternet},
year = {2004},
month = {March},
annote = {},
submitter = {Johan Garcia},
bibdate = {sun sep 26 21:56:40 CET 2004}
}
@TechReport{Floyd97,
author = { Floyd, S., and Fall, K.},
title = {Router Mechanisms to Support End-to-End Congestion Control},
year = {1997},
institution = {Lawrence Berkeley National Laboratory},
number = { },
adress = {Berkeley, CA},
month = {February},
submitter = {Johan Garcia},
bibdate = {September 26, 2004, 11:01 pm},
annote = {Mentions among other things RED byte mode vs RED packet mode.}
}
@article{Kohler04d,
author = {Eddie Kohler and Mark Handley and Sally Floyd and Jitendra Padhye},
title = {Datagram Congestion Control Protocol ({DCCP})},
journal = {draft-ietf-dccp-spec-07.txt, Work in progress},
year = {2004},
month = {July},
annote = {Abstract: This document specifies the Datagram Congestion Control Protocol (DCCP), which implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. },
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-spec-07.txt},
bibdate = {Oct 10, 2004 at 21:24:34 (CEST)},
submitter = {Johan Garcia}
}
@article{Kohler04e,
author = {Sally Floyd and Eddie Kohler},
title = {Profile for {DCCP} Congestion Control ID 2: {TCP}-like Congestion Control},
journal = {draft-ietf-dccp-ccid2-06.txt, Work in progress},
year = {2004},
month = {July},
annote = { Abstract: This document contains the profile for Congestion Control Identifier 2, TCP-like Congestion Control, in the Datagram Congestion Control Protocol (DCCP) [DCCP]. DCCP implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. The TCP-like Congestion Control CCID is used by senders who are able to adapt to the abrupt changes in the congestion window typical of TCP's AIMD (Additive Increase Multiplicative Decrease) congestion control. TCP-like Congestion Control is particularly useful for senders who would like to take advantage of the available bandwidth in an environment with rapidly changing conditions.},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-ccid2-06.txt},
submitter = {Johan Garcia},
bibdate = {Sunday, October 10, 2004 at 21:27:02 (CEST)}
}
@article{Kohler04f,
author = {Sally Floyd and Eddie Kohler and Jitendra Padhye},
title = {Profile for {DCCP} Congestion Control ID 3: {TFRC} Congestion Control},
journal = {draft-ietf-dccp-ccid3-06.txt, Work in progress},
year = {2004},
month = {July},
annote = { Abstract: This document contains the profile for Congestion Control Identifier 3, TCP-friendly rate control (TFRC), in the Datagram Congestion Control Protocol (DCCP). DCCP implements a congestion-controlled unreliable datagram flow suitable for use by applications such as streaming media. The TFRC CCID is used by applications that want a TCP-friendly send rate, possibly with Explicit Congestion Notification (ECN), while minimizing abrupt rate changes.},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-ccid2-06.txt},
submitter = {Johan Garcia},
bibdate = {Oct 10, 2004 at 21:28:57 (CEST)}
}
@article{Kohler04d,
author = {Eddie Kohler and Mark Handley and Sally Floyd and Jitendra Padhye},
title = {Datagram Congestion Control Protocol ({DCCP})},
journal = {draft-ietf-dccp-spec-07.txt, Work in progress},
year = {2004},
month = {July},
annote = {Abstract: This document specifies the Datagram Congestion Control Protocol (DCCP), which implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. },
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-spec-04.txt},
bibdate = {Oct 10, 2004 at 21:24:34 (CEST)},
submitter = {Johan Garcia}
}
@misc{rfc3828,
author="L-A. Larzon and M. Degermark and S. Pink and L-E. Jonsson and G. Fairhurst",
title="{The Lightweight User Datagram Protocol (UDP-Lite)}",
series="Request for Comments",
number="3828",
howpublished="RFC 3828 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2004,
month=jul,
url="http://www.ietf.org/rfc/rfc3828.txt",
}
@TechReport{Melander02,
author = {B. Melander and M. Björkman},
title = {Trace-Driven Network Path Emulation},
year = {2002},
institution = {Department of Information Technology, Uppsala University, Sweden},
number = {2002-037},
adress = {Uppsala University, Uppsala},
type = {Technical report},
submitter = {Stefan Alfredsson},
bibdate = {November 9, 2004, 4:35 pm},
annote = {This paper reports on on-going work where a trace-driven approach to network path emulation is investigated. Time stamped probe packets are sent along a network path whereby a probe packet trace can be generated. It basically contains the send times and the one-way delays/loss indications of the probe packets. Inside the emulator, the probe packet trace is used by a loss model and a delay model. These determine if a packet should be dropped or what the delay of the packet should be. Three loss models and three delay models are evaluated. For non-responsive UDP-based flows, the trace-driven loss and delay models that determine loss and delay based on loss-rates and delay distribution parameters calculated across the probe packet trace using a small gliding window are found to perform best. For adaptive TCP flows, none of the evaluated trace-driven models performs well. Instead, the Bernoulli loss model and an independent average delay model performs best.}
}
@mastersthesis{Yeom98,
author = {I. Yeom},
title = {{ENDE}: An End-to-end Network Delay Emulator},
school = {Texas A\&M University, College Station, Texas, USA},
year = {1998},
url = {papers/Yeom98-ENDE.ps.gz}
}
@TechReport{Allman97b,
author = {M. Allman and S. Ostermann},
title = {{ONE:} {T}he {O}hio {N}etwork {E}mulator},
year = {1997},
month=aug,
institution = {Ohio University},
number = {TR-19972},
type = {Technical Report},
submitter = {Stefan Alfredsson},
URL = {http://roland.lerc.nasa.gov/~mallman/papers/em.ps},
bibdate = {November 9, 2004, 4:51 pm},
annote = {Studying network protocols and distributed applications in real networks can be difficult due to the need for complex topologies, hard to find physical channels (e.g., satellite channels), and conditions beyond the control of a researcher (e.g., queue sizes). Network emulators can provide a controlled and reproducible environment for network testing. This paper discusses ONE, a network emulator we have written and tested.}
}
@article{Ingham94,
author = "D. B. Ingham and G. D. Parrington",
title = "Delayline: A Wide-Area Network Emulation Tool",
journal = "Computing Systems",
volume = "7",
number = "3",
pages = "313--332",
year = "1994",
url = "citeseer.ist.psu.edu/ingham94delayline.html"
}
@misc{rfc2041,
author="B. Noble and G. Nguyen and M. Satyanarayanan and R. Katz",
title={Mobile Network Tracing},
series="Request for Comments",
number="2041",
howpublished="RFC 2041 (Informational)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1996,
month=oct,
url="http://www.ietf.org/rfc/rfc2041.txt",
}
@article{Nistnet04,
author = "{NIST Internetworking Technology Group}",
title = {{NIST Net} network emulation package},
journal = {http://snad.ncsl.nist.gov/itg/nistnet/},
year = {2004},
month = {November},
submitter = {Stefan Alfredsson},
bibdate = {tis nov 9 22:22:25 CET 2004}
}
@article{Kohler04h,
author = {Sally Floyd and Eddie Kohler},
title = {Profile for {DCCP} Congestion Control ID 2: {TCP}-like Congestion Control},
journal = {draft-ietf-dccp-ccid2-07.txt, Work in progress},
year = {2004},
month = {October},
annote = { Abstract: This document contains the profile for Congestion Control Identifier 2, TCP-like Congestion Control, in the Datagram Congestion Control Protocol (DCCP) [DCCP]. DCCP implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. The TCP-like Congestion Control CCID is used by senders who are able to adapt to the abrupt changes in the congestion window typical of TCP's AIMD (Additive Increase Multiplicative Decrease) congestion control. TCP-like Congestion Control is particularly useful for senders who would like to take advantage of the available bandwidth in an environment with rapidly changing conditions.},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-ccid2-07.txt},
submitter = {Johan Garcia},
bibdate = {Tuesday, November 07, 2003 at 21:27:02 (CEST)}
}
@article{Kohler04g,
author = {Eddie Kohler and Mark Handley and Sally Floyd},
title = {Datagram Congestion Control Protocol ({DCCP})},
journal = {draft-ietf-dccp-spec-08.txt, Work in progress},
year = {2004},
month = {October},
annote = {Abstract: This document specifies the Datagram Congestion Control Protocol (DCCP), which implements a congestion-controlled, unreliable flow of datagrams suitable for use by applications such as streaming media. },
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-spec-08.txt},
bibdate = {Nov 11, 2004 at 21:24:34 (CEST)},
submitter = {Johan Garcia}
}
@article{Kohler04i,
author = {Sally Floyd and Eddie Kohler and Jitendra Padhye},
title = {Profile for {DCCP} Congestion Control ID 3: {TFRC} Congestion Control},
journal = {draft-ietf-dccp-ccid3-07.txt, Work in progress},
year = {2004},
month = {October},
annote = { Abstract: This document contains the profile for Congestion Control Identifier 3, TCP-friendly rate control (TFRC), in the Datagram Congestion Control Protocol (DCCP). DCCP implements a congestion-controlled unreliable datagram flow suitable for use by applications such as streaming media. The TFRC CCID is used by applications that want a TCP-friendly send rate, possibly with Explicit Congestion Notification (ECN), while minimizing abrupt rate changes.},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-ccid2-07.txt},
submitter = {Johan Garcia},
bibdate = {Nov 10, 2004 at 21:28:57 (CEST)}
}
@TechReport
{LeBoudec2002,
author = "J. Widmer, C. Boutremans and Le Boudec, J. Y.",
title = "End-to-end Congestion Control for Flows with Variable Packet Size",
institution = "EPFL-DI-ICA",
number = "SSC/2002/82",
year = "2002",
month = "December",
submitter = {Johan Garcia},
bibdate = {Nov 10, 2004 at 21:28:57 (CEST)}
}
@TechReport{LeBoudec2002,
author = "J. Widmer, C. Boutremans and Le Boudec, J. Y.",
title = "End-to-end Congestion Control for Flows with Variable Packet Size",
institution = "EPFL-DI-ICA",
number = "SSC/2002/82",
year = "2002",
month = "December",
submitter = {Johan Garcia},
bibdate = {Nov 10, 2004 at 21:28:57 (CEST)}
}
@InProceedings{Grinnemo04c,
author = {K-J Grinnemo and A. Brunstrom},
title = {Some Observations on the Performance of SCTP-controlled Failovers in M3UA-based SIGTRAN Networks},
year = {2004},
location = {Second Swedish National Computer Networking Workshop (SNCNW) 2004},
month = {November},
submitter = {Karl-Johan Grinnemo},
bibdate = {November 12, 2004, 8:56 am}
}
@InProceedings{Grinnemo04d,
author = { S. Lindskog and K-J Grinnemo and A. Brunstrom},
title = {Physical Separation for Data Protection based on SCTP Multihoming},
year = {2004},
location = {Second Swedish National Computer Networking Workshop (SNCNW)},
month = {November},
submitter = {Karl-Johan Grinnemo},
bibdate = {November 12, 2004, 8:57 am}
}
@InProceedings{Zheng03,
author = {P. Zheng and L.M. Ni},
title = {EMPOWER: A network emulator for wireline and wireless networks},
year = {2003},
location = {San Francisco, California, USA},
organization = {IEEE Computer and Communications Societies},
booktitle = {Proceedings of IEEE InfoCom},
month = {March},
submitter = {Stefan Alfredsson},
URL = {http://ieeexplore.ieee.org/search/srchabstract.jsp?arnumber=1209215&isnumber=27207&punumber=8585&k2d},
bibdate = {November 15, 2004, 7:53 pm},
annote = {(authors abstract:) The increasing need of protocol development environments and network performance evaluation tools gives rise to the research of flexible, scalable, and accurate network emulators. The desired network emulator should be able to facilitate the emulation of either wireline or wireless networks. In the case when network topology is critical to the underlying network protocol, the emulator should provide specific mechanisms to emulate network topology. In this paper, we present a distributed network emulation system EMPOWER, which not only can fulfill those requirements, but also can generate user-defined network conditions and traffic dynamics at packet level. EMPOWER is highly scalable in that each emulator node could be configured to emulate multiple network nodes. Some significant research issues such as topology mapping scheme and scalability of the emulator are discussed and addressed. Preliminary emulation results show that EMPOWER is capable of assisting the study of both wireless and wireline network protocols and applications.}
}
@TechReport{Ronngren00,
author = {Jonas Rönngren and Kristofer Sandlund},
title = {Trace Driven Loss Patterns in Dummynet},
year = {2000},
institution = {CDT, Luleå University of Technology},
number = {smd080/dist2000},
adress = {Luleå, Sweden},
type = {Lab report},
month = {May},
submitter = {Stefan Alfredsson},
bibdate = {November 16, 2004, 12:41 pm},
annote = {The report is available at http://www.cdt.luth.se/net/courses/99-00/smd080/dist2000/dnet-rep.ps on 16 Nov 2004. The source code is locally available in papers/Ronngren00_dnet-dist.tar.gz, and was fetched from http://www.cdt.luth.se/net/courses/99-00/smd080/dist2000/dnet-dist.tar.gz}
}
@Article{Appenzeller04,
author = {Guido Appenzeller and Isaac Keslassy and Nick McKeown},
title = {Sizing Router Buffers },
journal = {Proceeding of ACM SIGCOMM},
address= {Portland, Oregon, USA},
month = {September},
year = {2004},
bibdate = {Wednesday, December 01, 2004 at 12:07:56 (CET)},
url = {papers/Appenzeller04_sizingRouterBuffers.pdf},
annote = {Discusses the rule of thumb of router buffer settings ie mean flow RTT*link capacity and suggests that it should de modified to also include /sqr(n) factor to decrease the buffer size when many flows are present. Interesting since it relates to the setting of buffers for our simulations/experiments.},
submitter = {Johan Garcia}
}
@InProceedings{test04,
author = {Test Testsson},
title = {test04},
year = {2004},
location = {location},
organization = {organization},
month = {January},
bibdate = {December 1, 2004, 1:04 pm},
annote = {test.}
}
@Article{Casetti02,
author = {Claudio Casetti and Mario Gerla and Saverio Mascolo and M.Y. Sansadidi and Ren Wang},
title = {{TCP W}estwood: End-to-End Congestion Control for Wired/Wireless Networks},
journal = {Wireless Networks},
volume = {8},
number = {5},
pages = {467-479},
month = {September},
year = {2002},
bibdate = {Wednesday, December 01, 2004 at 12:21:52 (CET)},
url = {papers/Casetti02__Westwood.pdf},
annote = {Journal version of their Mobicom01 paper...?},
submitter = {Johan Garcia}
}
@misc{rfc3042,
author="M. Allman and H. Balakrishnan and S. Floyd",
title={Enhancing {TCP}'s Loss Recovery Using Limited Transmit},
series="Request for Comments",
number="3042",
howpublished="RFC 3042 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2001,
month=jan,
url="http://www.ietf.org/rfc/rfc3042.txt",
annote = {Limited transmit is a TCP mechanism that extends the error recovery performed by TCP. The idea is to avoid timeout and instead increase the chance of fast retransmit. The limited transmit follows the conservation of packets principle: new data is transmitted upon arrival of a dupAck if it is allowed by the advertised window and if the number of outstanding segments $\leq cwnd + 2$. Limited transmit is beneficial when the cwnd is small or if a many segments are lost in the same window.},
}
@InProceedings{Grinnemo04b,
author = {Karl-Johan Grinnemo and Anna Brunstrom},
title = {Impact of Traffic Load on {SCTP} Failovers in {SIGTRAN}},
year = {2004},
booktitle = {International Conference on Networking 2005 ({ICN'05})},
address = {Grand Hôtel des Mascareignes, Reunion Island},
month = apr,
url = {papers/Grinnemo04b.pdf},
submitter = {Karl-Johan Grinnemo},
bibdate = {December 13, 2004, 11:30 pm}
}
@Article{Eddy04,
author = {Wesley M. Eddy and Shawn Ostermann and Mark Allman},
title = {New Techniques for Making Transport Protocols Robust to Corruption-Based Loss},
journal = {Computer Communication Review},
year = {2004},
number = {5},
volume = {34},
month = {October},
submitter = {Stefan Alfredsson},
bibdate = {January 5, 2005, 4:03 pm}
}
@Proceedings{Lindskog05a,
author = {Stefan Lindskog and Karl-Johan Grinnemo and Anna Brunstrom},
title = {Data Protection Based on Physical Separation: Concepts and Application Scenarios},
year = {2005},
location = {Singapore},
organization = {Third Internation Workshop on Internet Communications Security (WICS)},
month = {February},
submitter = {Karl-Johan Grinnemo},
bibdate = {February 18, 2005, 7:37 am},
annote = {In this paper using multiple physical network paths is proposed as a way to accomplish data protection.}
}
@Book{Burnett04,
author = {Robert Burnett and Anna Brunstrom and Anders Nilsson},
title = {Perspectives on Multimedia - Communication, Media and Information Technology},
year = {2004},
publisher = {Wiley},
note = {ISBN 0-470-86863-5},
ISBN = {0-470-86863-5},
submitter = {Stefan Alfredsson},
bibdate = {March 16, 2005, 3:19 pm}
}
@Article{Floyd05,
author = {Sally Floyd},
title = {Changes proposed to {TCP} (webpage) },
journal = {http://www.icir.org/floyd/tcp\_small.html},
year = {checked 2005-04-12},
bibdate = {Tuesday, April 12, 2005 at 12:23:26 (CEST) },
url = {http://www.icir.org/floyd/tcp_small.html },
submitter = {Johan Garcia}
}
@article{Wang03,
author = {S. Y. Wang et al.},
title = {The Design and Implementation of the {NCTUns} 1.0 Network Simulator},
journal = {Computer Networks},
year = {2003},
volume = {42},
number = {2},
pages = {175-197},
month = jun,
submitter = {Stefan Alfredsson}
}
@INPROCEEDINGS{Perennou04,
AUTHOR = {Tanguy Pérennou and Emmanuel Conchon and Laurent Dairaine and Michel Diaz},
TITLE = {Two-Stage Wireless Network Emulation},
BOOKTITLE = {Workshop on Challenges of Mobility (WCM 2004)},
MONTH = {August},
YEAR = {2004},
submitter = {Stefan Alfredsson}
}
@article{Carson03,
author = {M. Carson and D. Santay},
title = {{NIST Net}: A Linux-based Network Emulation Tool},
journal = {ACM SIGCOMM Computer Communication Review},
year = {2003},
number = {3},
volume = {33},
pages = {111-126},
submitter = {Stefan Alfredsson}
}
@Article{Rizzo98,
author = {Luigi Rizzo},
title = {Dummynet and Forward Error Correction},
journal = {Proc. Freenix 98, New Orleans},
month = jun,
year = {1998},
bibdate = {Friday, April 15, 2005 at 00:35:25 (CEST)},
submitter = {Johan Garcia}
}
@article{Floyd02,
author = {Sally Floyd, Mark Handley, Eddie Kohler},
title = {Problem Statement for {DCCP}},
journal = {draft-ietf-dccp-problem-00.txt, Work in progress},
year = {2002},
month = {October},
url = {http://www.icir.org/kohler/dcp/draft-ietf-dccp-problem-00.txt},
bibdate = {Apr 17, 2005},
submitter = {Stefan Alfredsson}
}
@inproceedings{Svensson03,
author = {A. Svensson and A. Ahlén and A. Brunstrom and T. Ottosson and M. Sternad},
title = {An {OFDM} Based System Proposal for {4G} Downlinks},
booktitle = {Proceedings of Multi-Carrier Spread Spectrum Workshop},
address = {Oberpfaffenhofen, Germany},
year = {2003},
url = {http://www.signal.uu.se/Publications/pdf/c0309.pdf},
submitter = {Stefan Alfredsson}
}
@inproceedings{Sternad03,
author = {M. Sternad and T. Ottoson and A. Ahl\'en and A. Svensson},
title = {Attaining both Coverage and High Spectral Efficiency with Adaptive {OFDM} Downlinks},
booktitle = {Proceedings of the IEEE Vehicular Technology Conference (VTC-Fall)},
address = {Orlando, Florida, USA},
month = oct,
year = {2003},
url = {http://www.signal.uu.se/Publications/pdf/c0309.pdf},
submitter = {Stefan Alfredsson}
}
@inproceedings{Wang03a,
author = {W. Wang and T. Ottosson and M. Sternad and A. Ahl\'en and A. Svensson},
title = {Impact of multiuser diversity and channel variability on adaptive {OFDM}},
booktitle = {Proceedings of the IEEE Vehicular Technology Conference (VTC-Fall)},
address = {Orlando, Florida, USA},
month = oct,
year = {2003},
submitter = {Stefan Alfredsson}
}
@inproceedings{Widenius03,
author = {A. Widenius},
title = {Simulation of the {TCP/IP} over the Wireless Link},
booktitle = {PCC/Ericsson/TeliaSonera Research Days},
address = {Stockholm},
year = {2003},
month = {Nov. 18-19},
submitter = {Stefan Alfredsson}
}
@inproceedings{Ekman01,
author = {T. Ekman and M. Sternad and A. Ahl\'en},
title = {Unbiased Power Prediction on Broadband Channels},
booktitle = {Proceedings of the IEEE Vehicular Technology Conference (VTC-fall)},
address = {Vancouver, Canada},
year = {2002},
month = {Sep},
submitter = {Stefan Alfredsson}
}
@inproceedings{Ekman02,
author = {T. Ekman and M. Sternad and A. Ahl\'en},
title = {Unbiased Power Prediction on Broadband Channels},
booktitle = {IEEE Vehicular Technology Conference (VTC-Fall)},
address = {Vancouver, Canada},
year = {2002},
month = sep,
submitter = {Stefan Alfredsson}
}
@inproceedings{Sternad04,
author = {M. Sternad and S. Falahati},
title = {Maximizing throughput with adaptive {M-QAM} based on imperfect channel predictions},
booktitle={Proceedings of the IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)},
address = {Barcelona, Spain},
year = {2004},
month = sep,
submitter = {Stefan Alfredsson}
}
@misc{IEEE80211,
author = {{The Working Group for WLAN Standards}},
title = {{IEEE} 802.11 Wireless local area networks},
note = {\url{http://grouper.ieee.org/groups/802/11/} Visited 2011-09-02},
submitter = {Stefan Alfredsson}
}
@misc{Bluetooth,
author = {{The Bluetooth Special Interest Group}},
title = {The official {Bluetooth} SIG member website},
note = {\url{http://www.bluetooth.org/} Visited 2012-02-10},
submitter = {Stefan Alfredsson}
}
@phdthesis{Johansson04,
author = {Mathias Johansson},
title = {Resource Allocation under Uncertainty},
school = {Signals and Systems, Uppsala University},
year = {2004},
submitter = {Stefan Alfredsson}
}
@phdthesis{Casimiro04,
author = {Nilo Casimiro Ericsson},
title = {Revenue Maximization in Resource Allocation},
school = {Signals and Systems, Uppsala University},
year = {2004},
submitter = {Stefan Alfredsson}
}
@InProceedings{Grinnemo05a,
author = {Karl-Johan Grinnemo and Torbjorn Andersson and Anna Brunstrom},
title = {Performance Benefits of Avoiding Head-of-Line Blocking in SCTP},
year = {2005},
location = {Tahiti},
organization = {IEEE and IARIA},
month = {October},
submitter = {Karl-Johan Grinnemo},
bibdate = {July 6, 2005, 12:09 pm}
}
@InProceedings{Natu03,
author = {Ambarish Natu and David Taubman},
title = {Unequal Protection of JPEG2000 Code-Streams in Wireless Channels},
year = {2002},
location = {IEEE Global Telecommunications Conference (GLOBECOM)},
month = {November},
submitter = {Hannes Persson},
bibdate = {July 18, 2005, 9:05 pm},
annote = {This paper addresses the problem of robust transmission of image data across wireless links. The underlying thought is to provide guidance concerning the selection of JPEG2000 coding parameters and appropriate combinations of Reed-Solomon (RS) codes to be able to obtain a robust transmission. There exists some degree of protection against errors in JPEG2000. Although these techniques help to some extent, they are unable to recover lost data. In this paper an error-correcting scheme is realised by applying RS codes, with different settings concerning their protection capabilities, to parts of the image data. Thus obtaining unequal protection. In JPEG2000 the arithmetic compressed data is divided into smaller parts called packets. A packet consists of a header that holds vital information about how to decompress the packet payload (i.e code-blocks). Encoding and decoding of code-blocks are independant processes, so bit error in the bit-stream of any code-block does not propogate through out the bit-stream. By applying different protection on packet headers and payloads, firstly higher robustness against bit-errors is obtained and secondly high image compression is preserved. Simulation results (using a binary symmetric channel with bit-error rate of 10-4 and 10-3) with the proposed technique show large improvements in image quality compared to the baseline protection in JPEG2000.}
}
@inproceedings{Sundaresan03,
author = {K. Sundaresan and V. Anantharaman and H. Hsieh and R. Sivakumar},
title = {{ATP}: a reliable transport protocol for ad-hoc networks},
booktitle = {MobiHoc '03: Proceedings of the 4th ACM international symposium on Mobile ad hoc networking \& computing},
year = {2003},
isbn = {1-58113-684-6},
pages = {64--75},
location = {Annapolis, Maryland, USA},
doi = {http://doi.acm.org/10.1145/778415.778424},
publisher = {ACM Press},
address = {New York, NY, USA},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:15 pm}
}
@InProceedings{Anastasi05,
author = {G. Anastasi and E. Ancillotti and M. Conti and P. Passarella},
title = {{TPA}: A Transport Protocol for Ad Hoc Networks},
year = {2005},
booktitle = {Proceedings of IEEE ISCC},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 11:49 am}
}
@InProceedings{Anastasi03,
author = {G. Anastasi and A. Passarella},
title = {Towards a Novel Transport Protocol for Ad Hoc Networks},
year = {2003},
booktitle = {Proceedings of IFIP PWC},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 11:50 am}
}
@Article{Liu01,
author = {J. Liu and S. Singh},
title = {{ATCP}: {TCP} for Mobile Ad Hoc Networks},
journal = {IEEE JSAC},
year = {2001},
number = {7},
volume = {19},
month = {July},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 11:54 am}
}
@InProceedings{Hsieh02,
author = {H. Hsieh and R. Sivakumar},
title = {{pTCP}: An end-to-end transport layer protocol for striped connections},
year = {2002},
booktitle = {Proceedings of IEEE ICNP},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 11:56 am}
}
@InProceedings{Chen04,
author = {K. Chen and K. Nahrstedt},
title = {Limitations of equation-based congestion control in mobile ad hoc networks},
year = {2004},
booktitle = {Proceedings of WWAN+ICDCS'04},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:01 pm}
}
@InProceedings{Chen04b,
author = {K. Chen and K. Nahrstedt and N. Vaidya},
title = {The utility of explicit rate-based flow control in mobile ad hoc networks},
year = {2004},
booktitle = {Proceedings of IEEE WCNC},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:02 pm}
}
@InProceedings{Tang03,
author = {K. Tang and K. Obraczka and S. Lee and M. Gerla},
title = {Reliable Adaptive Lightweight Multicast Protocol},
year = {2003},
booktitle = {Proceedings of IEEE ICC},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:03 pm}
}
@Article{Mao06c,
author = {S. Mao and D. Bushmitch and S. Narayanan and S.S. Panwar},
title = {{MRTP}: A multi-flow real-time transport protocol for ad hoc networks},
journal = {IEEE Transactions on Multimedia},
volume = {},
number = {},
month = {},
year = {},
pages = {},
note = {to appear},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:03 pm}
}
@InProceedings{Riga05,
author = {N. Riga and A. Medina and I. Matta and C. Partridge and J. Redi and I. Castineyra },
title = {Transport Services for Energy Constrained Environments},
year = {2005},
booktitle = {SIGCOMM'05 Work In Progress Session},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:09 pm}
}
@InProceedings{Eshak03,
author = {Eshak, N. and Baba, M.D.},
title = {Design a new transport protocol (wireless {TCP}) to support mobility for mobile ad hoc networks},
year = {2003},
booktitle = {Proceedings of IEEE NCTT},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:10 pm}
}
@InProceedings{FuZ03,
author = {Z.Fu, X. Meng},
title = {A Transport Protocol for Supporting Multimedia Streaming in Mobile Ad Hoc Networks},
year = {2003},
booktitle = {IEEE JSAC},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:12 pm}
}
@InProceedings{FuZ05,
author = {Z. Fu and H. Luo and P. Zerfors and S. Lu and M. Gerla},
title = {The Impact of Multihop Wireless Channel on {TCP} Performance},
year = {2005},
booktitle = {IEEE Transactions on Mobile Computing},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:13 pm}
}
@InProceedings{FuZ02,
author = {Z. Fu and B. Greenstein and X. Meng and S. Lu},
title = {Design and Implementation of a {TCP}-Friendly Transport Protocol for Ad Hoc Wireless Networks},
year = {2002},
booktitle = {Proceedings of IEEE ICNP},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:13 pm}
}
@InProceedings{Sun01,
author = {D. Sun and H. Man},
title = {{ENIC} - an improved reliable transport scheme for mobile ad hoc networks},
year = {2001},
booktitle = {Proceedings of IEEE GLOBECOM},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:14 pm}
}
@InProceedings{Xylomenos03,
author = {G. Xylomenos and G. C. Polyzos},
title = {Wireless link layer enhancements for {TCP} and {UDP} applications},
year = {2003},
booktitle = {Proceedings of IPDPS/WMAN},
submitter = {Stefan Alfredsson},
bibdate = {January 8, 2006, 12:15 pm}
}
@InProceedings{Zhai05,
author = {H. Zhai and X. Chen and Y. Fang},
title = {Rate-Based Transport Control for Mobile Ad Hoc Networks},
year = {2005},
booktitle = {Proceedings of IEEE WCNC},
submitter = {Stefan Alfredsson},
bibdate = {January 9, 2006, 11:30 am}
}
@MastersThesis{Heimlicher05,
author = {Simon Heimlicher},
title = {{SAFT} -- Store And Forward Transport in Mobile Ad-hoc Networks},
year = {2005},
school = {TIK/ETH Zurich},
url = {http://www.tik.ee.ethz.ch/~simonh/public/sh-20050430-saft.pdf},
submitter = {Stefan Alfredsson},
bibdate = {January 9, 2006, 11:39 am}
}
@Article{Velayutham06,
author = {A. Velayutham and K. Sundaresan and R. Sivakumar},
title = {{NCTP}: A Non-Conversational Transport Layer Protocol for Wireless Ad-hoc Networks},
journal = {under submission},
year = {2006},
submitter = {Stefan Alfredsson},
URL = {http://users.ece.gatech.edu/~vel/nctp.pdf},
bibdate = {February 14, 2006, 9:17 pm}
}
@article{Rhee05,
author = {Injong Rhee and Lisong Xu},
title = {Limitations of equation-based congestion control},
journal = {ACM SIGCOMM Computer Communication Review},
volume = {35},
number = {4},
year = {2005},
issn = {0146-4833},
pages = {49--60},
doi = {http://doi.acm.org/10.1145/1090191.1080099},
publisher = {ACM Press},
submitter = {Stefan Alfredsson},
address = {New York, NY, USA},
}
@misc{rfc2960,
author="R. Stewart and Q. Xie and K. Morneault and C. Sharp and H. Schwarzbauer and T. Taylor and I. Rytina and M. Kalla and L. Zhang and V. Paxson",
title={Stream Control Transmission Protocol},
series="Request for Comments",
number="2960",
howpublished="RFC 2960 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2000,
month=oct,
url="http://www.ietf.org/rfc/rfc2960.txt",
}
@Misc{Caro02,
author = {A. Caro JR and J.R Iyengar and P.D. Amer and G.J Heinz and R.R. Stewart},
title = {Using SCTP multihoming for fault tolerance and load balancing},
year = {2002},
submitter = {Karl-Johan Grinnemo},
bibdate = {March 3, 2006, 3:46 pm}
}
@PhDThesis{Caro05,
author = {A. Caro JR},
title = {End-to-end Fault Tolerance using Transport Layer Multihoming},
year = {2005},
school = {University of Delaware},
submitter = {Johan Eklund},
bibdate = {April 12, 2006, 12:04 pm}
}
@Manual{Claffy98,
author = {K. Claffy and G. Miller and K. Thompson},
title = {The Nature of the Beast Recent Traffic Measurements from an Internet Backbone},
year = {1998},
submitter = {Johan Eklund},
bibdate = {May 2, 2006, 5:16 pm},
annote = {Recent Traffic Measurements from an Internet Backbone. In Proceedings of the INET '98 Conference, Geneva, Switzerland , July 1998. Internet Society.}
}
@inproceedings{ ayesta02effect,
author = "U. Ayesta and K. Avrachenkov",
title = "The effect of the initial window size and limited transmit algorithm on
the transient behavior of {TCP} transfers",
booktitle = "15th ITC Specialist Seminar on Internet Traffic Engineering and Traffic
Management, W{\"u}rzburg, Germany",
month = jul,
year = "2002",
url = "citeseer.ist.psu.edu/ayesta02effect.html"
}
@InProceedings{Sternad06,
author = {M. Sternad and T. Svensson and G. Klang},
title = {The {WINNER} {B3G} System {MAC} Concept},
year = {2006},
booktitle = {To be published in proceedings of {VTC} 2006-Fall},
organization = {{IEEE}},
month = {September},
submitter = {Stefan Alfredsson},
bibdate = {July 4, 2006, 3:49 pm}
}
@techreport{Falahati05,
author = {S. Falahati ed. },
title = {Assessment of adaptive transmission technologies},
institution = {IST-2003-507581 WINNER},
year = "2005",
type = {WINNER Deliverable},
number = {D2.4},
note = {Can be downloaded from http://www.ist-winner.org},
url = "http://www.ist-winner.org/DeliverableDocuments/D2.4.pdf"
}
@TechReport{Eklund05,
author = {J. Eklund},
title = {Performance of Network Redundancy Mechanisms in SCTP},
year = {2005},
institution = {Division for Information Technology},
number = {205:48},
adress = {Karlstad University},
month = {December},
submitter = {Johan Eklund},
bibdate = {August 10, 2006, 1:42 pm}
}
@InProceedings{Alfredsson06,
author = {S. Alfredsson and A. Brunstrom and M. Sternad},
title = {Transport Protocol Performance over {4G} Links: Emulation Methodology and Results},
year = {2006},
booktitle={Proceedings of the International Symposium on Wireless Communication Systems (ISWCS)},
address={Valencia, Spain},
month = sep,
submitter = {Stefan Alfredsson},
}
@article{Xu05,
author = {Kai Xu and Ye Tian and Nirwan Ansari},
title = {Improving {TCP} performance in integrated wireless communications networks},
journal = {Computer Networks},
year = {2005},
volume = {47},
number = {2},
pages = {219-237},
month = {Feb},
submitter = {Stefan Alfredsson}
}
@misc{ Kim04,
author = "K. kim and F. Baccelli",
title = "TCP Throughput Analysis under Transmission Error and Congestion Losses",
text = "{TCP} Throughput Analysis under Transmission Error and Congestion Losses. In Proceedings of IEEE INFOCOM, Hongkong, March 2004.", year = "2004",
url = "citeseer.ist.psu.edu/baccelli04tcp.html" }
@InProceedings{Bhandarkar05,
author = {Sumitha Bhandarkar and Nauzad Erach Sadry and A.L. Narasimha Reddy and Nitin H. Vaidya},
title = {{TCP-DCR}: A Novel Protocol for Tolerating Wireless Channel Errors},
year = {2005},
month = sep,
pages = {517-529},
volume = 4,
booktitle = {IEEE Transactions on Mobile Computing},
submitter = {Stefan Alfredsson},
}
@inproceedings{Budzisz07,
author = {L. Budzisz and R. Ferrus and K-J Grinnemo and A. Brunstrom and F. Casadevall},
title = {Analytical Estimation of the Failover Time in SCTP Multihoming Scenarios},
booktitle = {proceedings of IEEE Wireless Communications and Networking Conference (WCNC). Hong Kong. },
year = {2007},
location = {Hong Kong},
month = {March},
submitter = {Johan Eklund},
bibdate = {February 28, 2007, 5:28 pm},
}
@InProceedings{Iyengar04,
author = {Iyengar J and Amer P and Stewart R},
title = {Current multipath transfer using transport layer multihoming - performance under varying bandwidth proportions},
booktitle = {Proceedings IEEE Military Communications Conference (MILCOM)'},
year = {2004},
month = {October},
submitter = {Johan Eklund},
bibdate = {February 28, 2007, 5:50 pm}
}
@InProceedings{Sternad03b,
author = {M. Sternad and D. Aronsson},
title = {Channel estimation and prediction for adaptive {OFDM} downlinks},
booktitle = {Proceedings of the IEEE Vehicular Technology Conference (VTC-Fall)},
address = {Orlando, Florida, USA},
month=oct,
year = {2003}
}
@PhDThesis{Grinnemo06,
author = {K-J. Grinnemo},
title = {Transport Services for Soft Real-Time Applications in {IP} Networks},
year = {2006},
school = {Karlstad University},
bibdate = {March 22, 2007, 11:29 am}
}
@InProceedings{Eklund06,
author = {Eklund. J and Brunstrom. A},
title = {Impact of SACK delay and link delay on Failover performance in SCTP},
booktitle = {The Third IASTED International Conference on communications and computer networks},
year = {2006},
address = {Lima, Peru},
month = {October},
submitter = {Johan Eklund},
bibdate = {March 22, 2007, 12:05 pm}
}
@InProceedings{Eklund08,
author = {Eklund. J and Brunstrom. A and Grinnemo, K-J},
title = {On the Relation Between SACK Delay and SCTP Failover Performance for Different Traffic Distributions},
booktitle = {The Fifth International Conference on Broadband Communications, Networks, and Systems},
year = {2008},
address = {London, UK},
month = {September},
submitter = {Johan Eklund},
bibdate = {May 29, 2008, 15:05 pm}
}
@InProceedings{White02,
author = {B. White and J. Lepreau and L. Stoller and R. Ricci and S. Guruprasad and M. Newbold and M. Hibler and C. Barb and A.Joglekar},
title = {An integrated experimental environment for distributed systems and networks},
booktitle = {Proc. of the Fifth Symposium on Operating Systems Design and Implementation, pages 255-270},
organization = {Boston, MA, Dec. 2002},
year = {2003}
}
@InProceedings{Casetti04,
author = { Casetti, C. and Gaiotto, W. },
title = { Westwood {SCTP}: load balancing over multipaths using bandwidth-aware source scheduling },
booktitle = { {60th IEEE Vehicular Technology Conference (VTC 2004-Fall)} },
PAGES = { 3025-3029 },
year = { 2004 },
submitter = {Johan Eklund},
bibdate = {April 13, 2007, 17:05 pm}
}
@InProceedings{Rembarz05,
author = { R. Rembarz S. Baucke and P. Mahonen },
title = { Enhancing resilience for high availability IP-based signaling transport },
booktitle={Proceedings of the IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)},
PAGES = { 2301-2305 },
year = { 2005 },
MONTH = { September },
submitter = {Johan Eklund},
bibdate = {April 13, 2007, 17:05 pm}
}
@InProceedings{Fracchia05,
author = { Fracchia, R. and Casetti, C. and Chiasserini, C. and Meo, M },
title = { {A WiSE extension of {SCTP} for wireless networks} },
booktitle = { IEEE International Conference on Communications, (ICC 2005) },
PAGES = { 1448-1453 },
year = { 2005 },
MONTH = { May },
submitter = {Johan Eklund},
bibdate = {April 13, 2007, 17:05 pm}
}
@misc{RFC3469,
author = {Sharma, V. and Hellstrand. F},
title = {{RFC 3469: MPLS} Framework for MPLS-based Recovery},
journal = {http://www.ietf.org/},
year = {2003},
month = {February},
submitter = {Johan Eklund },
bibdate = {April 18, 2007 at 16:35:17 (CET)}
}
@misc{RFC3768,
author = {R. Hinden, Ed},
title = {{RFC 3768: VRRP} Virtual Router Redundancy Protocol},
journal = {http://www.ietf.org/},
year = {2004},
month = {April},
submitter = {Johan Eklund },
bibdate = {April 18, 2007 at 16:35:17 (CET)}
}
@article{Carter94,
author = { Carter, R. and Crovella, M.},
title = {Measuring Bottleneck Link Speed in Packet-Switched Networks},
year = {1996},
month = {October},
submitter = {Johan Eklund},
bibdate = { April 18, 2007 at 16:35:17 (CET)}
}
@InProceedings {Scharf06,
author = {Scharf, M. and Kiesel, S.},
title = { {Head-of-line Blocking in TCP and SCTP: Analysis and Measurements} },
booktitle = {IEEE Global Telecommunications Conference, 2006, (Globecom 2006) },
PAGES = { 1-5 },
year = { 2006 },
MONTH = { Nov },
submitter = {Johan Eklund},
bibdate = {May 2, 2007, 15:05 pm}
}
@misc{RFC4166,
title = {{RFC 4166}: Telephony Signalling Transport over
Stream Control Transmission Protocol (SCTP) Applicability Statement},
author = {Coene, L. and Pastor-Balbas, J.},
month = Feb,
year = {2006},
note = {},
bibdate = {Fri June 01 11:34:09 MDT 2007},
}
@misc {Baucke07,
author = {Baucke, S. and Grinnemo, K-J. and Ludwig, R. and Wolisz, A.},
title = { {Using Relaxed Timer Backoff to Reduce SCTP Failover Times} },
booktitle = {Work in progress},
PAGES = {},
year = {2007},
MONTH = {},
submitter = {Johan Eklund},
bibdate = {June 2, 2007, 11:05 am}
}
@misc{Seth99,
title = "{Performance Requirements for TCAP Signaling in Internet Telephony, Work in Progress.}",
author = {Seth, T. and Broscius, A. and Huitema, C. and Lin, H-A P.},
month = Feb,
year = {1999},
note = {Work in progress},
bibdate = {Thu Aug 30 1:11:09 MDT 2007},
}
@article{JBIG07,
author = {3GPP},
title = {3GPP TS 23.228 v.8.1.0 (2007-06) 3rd Generation Partnership Project; Technical Specification Group Services and System Aspects; IP Multimedia Subsystem (IMS); Stage 2 (Release 8)
},
journal = {Geneva, Switzerland},
year = {2007},
month = {June},
submitter = {Johan Eklund},
bibdate = {Monday, September 3, 2007 at 17:33:10 (CET)}
}
@book{Camarillo06,
author = {Gonzalo Camarillo and Miguel-Angel García-Martín },
title = {The 3G IP Multimedia Subsystem (IMS): Merging the Internet and the Cellular Worlds, Second Edition },
publisher = {Johan Wiley and Sons},
year = {2006},
address = {},
annote = {}
}
@misc{RFC2719,
author = {L. Ong and I. Rytina and M. Garcia and H. Schwarzbauer and L. Coene and H. Lin and I. Juhasz and M. Holdreg and C. Sharp},
title = {{RFC 2719}: Framework Architecture for Signaling Transport},
year = {1999},
month = {October},
annote = {},
url = {},
bibdate = {Tuesday, Sept 07, 2007 at 10:49:33 (CET)},
submitter = {Johan Eklund}
}
@article{Allman98B,
author = {Allman M},
title = {On the generation and use of TCP acknowledgements},
journal = {Computer Communication Review. vol.28, no.5 pp 4-21, Oct 1998},
year = {1998},
pages = {4-21},
annote = {},
url = {}
}
@article{Xu01,
author = {S. Xu and T. Saadawi},
title = {Evaluation for TCP with Delayed ACK Option in Wireless multi-hop Networks},
journal = {IEEE VTS 54th Vehicular Technology Conference, vol. 1, pp. 267-271, 2001},
year = {2001},
pages = {267-271},
annote = {},
url = {}
}
@misc{RFC3286,
author = {L. Ong and J. Yoakum},
title = {{RFC 3286}: An Introduction to the Stream Control Transmission Protocol (SCTP)},
year = {2002},
month = {May},
annote = {},
url = {},
bibdate = {Thursday, Sept 13, 2007 at 16:49:33 (CET)},
submitter = {Johan Eklund}
}
@misc{RFC3257,
author = {L. Coene},
title = {{RFC 3257}: Stream Control Transmission Protocol Applicability Statement},
year = {2002},
month = {April},
annote = {},
url = {},
bibdate = {Tuesday, Sept 07, 2007 at 10:52:33 (CET)},
submitter = {Johan Eklund}
}
@misc{RFC4460,
author = {R. Stewart and I. Arias-Rodriguez and K. Poon and A. Caro and M. Tuexen},
title = {{RFC 4460}: Stream Control Transmission Protocol (SCTP) Specification Errata and Issues},
year = {2006},
month = {April},
annote = {},
url = {},
bibdate = {Tuesday, Sept 14, 2007 at 15:52:33 (CET)},
submitter = {Johan Eklund}
}
@misc{RFC4168,
author = {J. Rosenberg and H. Schulzrinne and G. Camarillo},
title = {{RFC 4168}: The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP)},
year = {2005},
month = {October},
annote = {},
url = {},
bibdate = {Tuesday, Sept 25, 2007 at 11:52:33 (CET)},
submitter = {Johan Eklund}
}
@misc{RFC4083,
author = {M. Garcia-Martin},
title = {{RFC 4083}: The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP)},
year = {2005},
month = {October},
annote = {},
url = {},
bibdate = {Tuesday, Sept 25, 2007 at 12:12:33 (CET)},
submitter = {Johan Eklund}
}
@article{29229,
author = {3GPP},
title = {TS 29.229: V7.5.0 (2007-03). Technical Specification Group Core Network and Terminals; Cx and Dx interfaces based on the Diameter protocol; Protocol details (Release 7)},
journal = { },
year = {2007},
month = {March},
url = {},
bibdate = {Tuesday, September 25, 2003 at 16:36:56 (CEST)},
submitter = {Johan Eklund}
}
@article{Caro06,
author = {A. Caro et al.},
title = {Rethinking end-to-end failover with transport layer multihoming},
journal = {Annales des télécoms},
year = {2006},
volume = {61},
number = {1-2},
submitter = {Johan Eklund},
bibdate = {Monday, October 15, 2007 at 16:18:20 (CEST)}
}
@article{Janardhan06,
author = {R. Janardhan et al.},
title = {Concurrent multipath transfer using SCTP multihoming over independent end-to-end paths},
journal = {IEEE/ACM Transactions on Networking (TON)},
year = {2006},
volume = {14, pages 951-964},
number = {},
submitter = {Johan Eklund},
bibdate = {Monday, October 15, 2007 at 16:22:20 (CEST)}
}
@inproceedings{Ribeiro05,
author = {E.Ribeiro and V.Leung},
title = {Asymmetric path delay optimization in mobile multi-homed SCTP multimedia transport},
booktitle = {Proc. {IEEE} Int. Conf. Image},
year = {2005},
address = {Quebec, Canada},
month = {Oct},
annote = {},
submitter = {Johan Eklund}
}
@inproceedings{Scholtz02,
author = {F.J. Scholtz},
title = {Statistical analysis of common channel signaling system no. 7 traffic},
booktitle = {In proceedings of 15th Internet Traffic Engineering and Traffic Management (ITC) Specialist Seminar},
year = {2002},
address = {Wurzburg, Germany},
month = {},
annote = {},
submitter = {Johan Eklund}
}
@article{Caro06b,
author = {A. Caro and P. Amer and R. Stewart},
title = {Retransmission Policies for Multihomed Transport Protocols},
journal = {Computer Communications},
year = {2006},
volume = {29},
pages = {1798-1810},
month = {June},
annote = {},
bibdate = {Monday, October 22, 2007 at 16:55:19 (CEST)},
submitter = {Johan Eklund}
}
@misc{RFC4960,
author = {R. Stewart},
title = {{RFC} 4960: {S}tream {C}ontrol {T}ransmission {P}rotocol},
year = {2007},
month = {September},
annote = {obsoletes RFC 2960 , RFC 3309},
url = {},
bibdate = {Wednesday, October 31, 2007 at 17:09:09 (CET)},
submitter = {Johan Eklund}
}
@article{Buga02,
author = {W. Buga},
title = {The Evolution of oftswitch architecture},
journal = {Annual Review of Communications},
year = {2002},
volume = {55},
pages = {73-76},
month = {},
annote = {},
bibdate = {Tuesday, November 06, 2007 at 13:55:19 (CEST)},
submitter = {Johan Eklund}
}
@misc{RFC5061,
author = {R. Stewart and Q.Xie and M. Tuexen and S. Maruyama and M. Kozuka},
title = {{RFC} 4960: {S}tream {C}ontrol {T}ransmission {P}rotocol {D}ynamic {A}ddress {R}econfiguration}},
year = {2007},
month = {September},
annote = {}
url = {},
bibdate = {Friday, November 09, 2007 at 12:09:09 (CET)},
submitter = {Johan Eklund}
}
@misc{RFC3758,
author="R. Stewart and M. Ramalho and Q. Xie and M. Tuexen and P. Conrad",
title={Stream Control Transmission Protocol {(SCTP)} Partial Reliability Extension},
series="Request for Comments",
number="3758",
howpublished="RFC 3758 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2004,
month=may,
url="http://www.ietf.org/rfc/rfc3758.txt",
}
@misc{RFC3261,
title = {{RFC} 3261:{SIP} Session Initiation Protocol},
author = {J. Rosenberg and H. Schulzrinne and G. Camarillo and A. Johanston and J.Peterson and R. Sparks and M. Handley and E.Schooler},
month = {June},
year = {2002},
annote = { },
bibdate = {Tues Feb 6 16:34:09 MDT 2007},
submitter = {Johan Eklund}
}
@article{Raghunathan07,
author = {V. Raghunathan and P.R. Kumar},
title = {A counterexample in congestion control of wireless networks},
journal = {Performance Evaluation},
publisher = {Elsevier},
year = {2007},
volume = {64, Pages 399-418},
number = {5},
doi = {http://dx.doi.org/10.1016/j.peva.2006.08.005},
submitter = {Stefan Alfredsson},
bibdate = {feb 22 16:01:15 CET 2008}
}
@ARTICLE{Fu07,
title={Modeling TCP Veno Throughput over Wired/Wireless Networks},
author={Cheng Peng Fu and Bin Zhou and Jian Ling Zhang},
journal={Communications Letters, IEEE},
year={September 2007},
volume={11},
number={9},
pages={723-725},
keywords={Internet, local area networks, radio links, transport protocolsInternet, LAN, TCP Veno throughput modeling, WAN, wired links, wireless links},
doi={10.1109/LCOMM.2007.070221},
ISSN={1089-7798},
submitter = {Stefan Alfredsson}
}
@ARTICLE{Scalia07,
title={PiggyCode: A MAC Layer Network Coding Scheme to Improve TCP Performance Over Wireless Networks},
author={Scalia, Luca and Soldo, Fabio and Gerla, Mario},
journal={Global Telecommunications Conference, 2007. GLOBECOM '07. IEEE},
year={26-30 Nov. 2007},
volume={},
number={},
pages={3672-3677},
doi={10.1109/GLOCOM.2007.697},
ISSN={},
submitter = {Stefan Alfredsson}
}
@ARTICLE{Jayananthan07,
title={Analytical Model of TCP with Enhanced Recovery Mechanism for Wireless Environments},
author={Jayananthan, A. and Sirisena, H. and Garg, V.},
journal={Communications, 2007. ICC '07. IEEE International Conference on},
year={24-28 June 2007},
volume={},
number={},
pages={4506-4511},
doi={10.1109/ICC.2007.744},
ISSN={},
submitter = {Stefan Alfredsson}
}
@inproceedings{Wang07a,
author = {Neng-Chung Wang and Chi-Lun Chiou and Yung-Fa Huang},
title = {{TCP} enhancement for transmission in a variable bandwidth wireless environment},
booktitle = {Proceedings of the ACM International Conference on Wireless Communications and Mobile Computing (IWCMC)},
year = {2007},
isbn = {978-1-59593-695-0},
pages = {37--42},
address = {Honolulu, Hawaii, USA},
doi = {http://doi.acm.org/10.1145/1280940.1280949},
}
@ARTICLE{Wang07b,
title={Improving {TCP} performance by {TCP-Acknowledge} agent based on {AP ARQ} over wireless links},
author={Ling Wang, and Guangxi Zhu, and Gang Su, and Weimin Wu,},
journal={Intelligent Signal Processing and Communication Systems, 2007. ISPACS 2007. International Symposium on},
year={Nov. 28 2007-Dec. 1 2007},
volume={},
number={},
pages={710-713},
doi={10.1109/ISPACS.2007.4445986},
ISSN={},
}
@ARTICLE{Ming07,
title={Cross-layer Resource Control to Improve TCP Performance over Wireless Network},
author={Li, Ming},
journal={Computer and Information Science, 2007. ICIS 2007. 6th IEEE/ACIS International Conference on},
year={11-13 July 2007},
volume={},
number={},
pages={706-711},
doi={10.1109/ICIS.2007.86},
ISSN={},
}
@inproceedings{Koo07,
title = {Sender-Based {TCP} Scheme for Improving Performance in Wireless Environment.},
author = {Jahwan Koo and Sung-Gon Mun and Hyunseung Choo},
booktitle = {Proceedings of the International Conference on Computational Science (ICCS)},
pages = {538--541},
publisher = {Springer},
series = {Lecture Notes in Computer Science},
url = {http://dblp.uni-trier.de/db/conf/iccS/iccS2007-4.html#KooMC07},
volume = {4490},
year = {2007},
description = {dblp},
ee = {http://dx.doi.org/10.1007/978-3-540-72590-9_73}, isbn = {978-3-540-72589-3}, date = {2007-07-23},
keywords = {dblp }
}
@inproceedings{Cui07,
title={Adaptive Increase and Adaptive Decrease Algorithm for Wireless {TCP}},
author={Cui, Lin and Koh, Seok J. and Cui, Xin and Kim, Yong Jin},
booktitle={International Conference on Natural Computation (ICNC)},
address={Hainan, China},
year={2007},
month=aug,
volume={2},
number={},
pages={392-398},
doi={10.1109/ICNC.2007.160},
ISSN={}, }
@INPROCEEDINGS{Malkowski07,
AUTHOR = {Malkowski, M. and Heier, S.},
TITLE = {Interaction between UMTS MAC Scheduling and TCP Flow Control Mechanisms},
YEAR = {2003},
MONTH = {Apr},
PAGES = {1373-1376},
ADDRESS = {Beijing, China},
BOOKTITLE = {Proceedings of 2003 International Conference on Communication Technology},
AFFILIATION = {Chair of Communication Networks (ComNets), Faculty 6, RWTH Aachen University},
ISBN = {7-563-50686-1},
KEYWORDS = {UMTS, Radio Interface Protocols, MAC Scheduler, QoS, WWW Traffic Model, UMTS Simulator},
URL = {http://www.comnets.rwth-aachen.de}
}
@InProceedings{Vacirca03,
author = {F. Vacirca and A. De Vendictis and A. Todini and A. Baiocchi},
title = {On the Effects of {ARQ} Mechanisms on {TCP} Performance in Wireless Environments},
booktitle={Proceedings of the IEEE Global Telecommunications Conference (GLOBECOM)},
year = {2003},
MONTH = dec,
address={San Francisco, USA}
}
@ARTICLE{Alcaraz06,
title={Improving {TCP} Performance over {3G} Links with an {ACK} Rate Control Algorithm},
author={Alcaraz, J.J. and Cerdan, F.},
journal={Proceedings of the International Symposium of Wireless Communication Systems (ISWCS)},
year={6-8 Sept. 2006},
volume={},
number={},
pages={480--484},
keywords={3G mobile communication, radio links, transport protocols, wireless channels3G cellular links, ACK rate control algorithm, RLC layer, TCP performance, automatic reconfiguration, downlink buffer, error recovery mechanisms, parameter setting, radio bearer bandwidth, wireless channel},
doi={10.1109/ISWCS.2006.4362344},
ISSN={}
}
@inproceedings{Mahmoodi07,
author = {Toktam Mahmoodi and Vasilis Friderikos and Oliver Holland and A. Hamid Aghvami},
title = {Cross-Layer Design to Improve Wireless {TCP} Performance with Link-Layer Adaptation},
booktitle = {Proceedings of the IEEE Vehicular Technology Conference (VTC-fall)},
year = {2007},
month = sep,
pages = {1504-1508},
ee = {http://dx.doi.org/10.1109/VETECF.2007.320},
bibsource = {DBLP, http://dblp.uni-trier.de}
}
@inproceedings{Klein04,
title={Avoiding spurious {TCP} timeouts in wireless networks by delay injection},
author={Klein, T.E. and Leung, K.K. and Parkinson, R. and Samuel, L.G.},
booktitle={Proceedings of the IEEE Global Telecommunications Conference (GLOBECOM)},
address={Dallas, Texas, USA},
year={2004},
volume={5},
number={},
pages={2754--2759},
keywords={ Internet, delays, packet radio networks, transport protocols Internet, Transmission Control Protocol, delay injection, network congestion level, spurious TCP timeouts, throughput, wireless networks},
doi={10.1109/GLOCOM.2004.1378856},
ISSN={ },
}
@article{Chan05,
author = {Mun Choon Chan and Ramachandran Ramjee},
title = {{TCP/IP} Performance over {3G} Wireless Links with Rate and Delay Variation},
journal = {Wireless Networks},
volume = {11},
number = {1-2},
year = {2005},
pages = {81-97},
ee = {http://www.springerlink.com/index/10.1007/s11276-004-4748-7},
bibsource = {DBLP, http://dblp.uni-trier.de}
}
@article{Chan08,
author = {Mun Choon Chan and Ram Ramjee},
title = {Improving {TCP/IP} Performance over Third-Generation Wireless Networks},
journal = {IEEE Transactions on Mobile Computing},
volume = {7},
number = {4},
year = {2008},
issn = {1536-1233},
pages = {430--443},
doi = {http://dx.doi.org/10.1109/TMC.2007.70737},
publisher = {IEEE Educational Activities Department},
address = {Piscataway, NJ, USA},
}
@ARTICLE{Wang07,
title={Improving {TCP} performance by {TCP-Acknowledge} agent based on {AP ARQ} over wireless links},
author={Ling Wang and Guangxi Zhu and Gang Su and Weimin Wu},
journal={International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)},
year=2007,
volume={},
number={},
pages={710-713},
doi={10.1109/ISPACS.2007.4445986}
}
@ARTICLE{Lim07,
title={An Adaptive End-to-End Loss Differentiation Scheme for {TCP} over Wired/Wireless Networks},
author={C. Lim and J. Jang},
journal={IJCSNS International Journal of Computer Science and Network Security},
year={2007},
volume={7},
number={3},
month={March},
pages={72-83},
}
@techreport{WINNER-D5.2,
author = {J. Meinilä and T. Jämsä and P. Kyösti and D. Laselva and H. El-Sallabi and J. Salo and C. Schneider and D. Baum},
title = {Determination of Propagation Scenarios},
institution = {IST-2003-507581 WINNER},
year = "2004",
type = {WINNER Deliverable},
number = {D5.2},
note = {Available at \url{http://www.ist-winner.org/DeliverableDocuments/D5.2_v1.1.pdf}},
url = "http://www.ist-winner.org/DeliverableDocuments/D5.2_v1.1.pdf"
}
@book{Proakis00,
author = {John G. Proakis},
title = {Digital communications},
publisher = {McGraw-Hill},
year = {2000},
address = {Boston}
}
@book{Goldsmith05,
author = {Andrea Goldsmith},
title = {Wireless Communications},
publisher = {Cambridge University Press},
year = {2005},
address = {Cambridge, New York},
note = {ISBN 978-0-521-83716-3}
}
@article{Chase85,
title={{Code combining--a maximum-likelihood decoding approach for combining an arbitrary number of noisy packets}},
author={David Chase},
journal={{IEEE} Transactions on Communications},
volume={33},
number={5},
pages={385--393},
year={1985}
}
@InProceedings{Sternad07,
author = {M. Sternad and T. Svensson and T.Ottosson and A. Ahlen and A. Svensson and A. Brunstrom},
title = {Towards Systems Beyond {3G} Based on Adaptive {OFDMA} Transmission},
booktitle = {Proceedings of the IEEE, Special Issue on Adaptive Transmission},
volume ={95:12},
pages ={2432--2455},
year = {2007}
}
@InProceedings{Garcia06,
author = {J. Garcia and S. Alfredsson and A. Brunstrom},
title = {The Impact of Loss Generation on Emulation-Based Protocol Evaluation},
booktitle = {Proceedings International Conference on Parallel and Distributed Computing and Networks (PDCN)},
address={Innsbruck, Austria},
year = 2006,
month = feb
}
@inproceedings{Alfredsson03,
author = {S. Alfredsson and A. Brunstrom},
title = {{TCP-L}: {A}llowing Bit Errors in Wireless {TCP}},
booktitle = {Proceedings of IST Mobile and Wireless Communications Summit 2003},
year = {2003},
month = jun,
address = {Aveiro, Portugal},
submitter = {Stefan Alfredsson},
bibdate = {Sunday, March 23, 2003 at 16:39:22 (CET)}
}
@article{Alfredsson03b,
author = {Stefan Alfredsson},
title = {Study of a Commercial {VoIP} Service Offering},
journal = {Project Report for 2G5564/KTH, Karlstad University, Sweden},
url = {http://www.cs.kau.se/~alfs/voip/},
year = {2003},
month = {Feb},
submitter = {Stefan Alfredsson}
}
@techreport{Wennstrom04,
author = {Annika Wennström and Stefan Alfredsson and Anna Brunstrom},
title = {{TCP over Wireless Networks}},
institution = {Karlstad University Press},
year = 2004,
type = {Karlstad University Studies},
number = {2004:21}
}
@InProceedings{Alfredsson04,
author = {Stefan Alfredsson and Anna Brunstrom and Mikael Sternad},
title = {A {4G} Link Level Emulator for Transport Protocol Evaluation},
year = {2004},
month=nov,
address={Karlstad, Sweden},
booktitle = {Proceedings of the Swedish National Computer Networking Workshop (SNCNW)},
}
@article{Alfredsson05,
author = {Stefan Alfredsson},
title = {{TCP} in Wireless Networks: Challenges, Optimizations and Evaluations},
journal = {Licentiate thesis, Karlstad University, Sweden},
year = {2005},
submitter = {Stefan Alfredsson},
}
@inproceedings{Alfredsson05b,
author = {S. Alfredsson and A. Brunstrom and M. Sternad},
title = {Emulation and validation of a {4G} system proposal},
year = {2005},
address = {Linkoping, Sweden},
booktitle = {Proceedings of {Radiovetenskap och Kommunikation}},
month =jun,
submitter = {Stefan Alfredsson}
}
% edit 2013-09-25 - year of publication was wrong (2007, should be 2008).
@InProceedings{Alfredsson08a,
author = {S. Alfredsson and A. Brunstrom and M. Sternad},
title = {Cross-layer analysis of {TCP} performance in a {4G} system},
month = oct,
year = {2008},
address={Reykjavik, Iceland},
booktitle={Proceedings of the International Symposium on Wireless Communication Systems (ISWCS)},
}
@InProceedings{Alfredsson07a,
author = {S. Alfredsson and A. Brunstrom and M. Sternad},
title = {Impact of {4G} Wireless Link Configurations on {VoIP} Network Performance},
month = sep,
year = {2007},
address = {Dubrovnik, Croatia},
booktitle={Proceedings of the International Conference on Software, Telecommunications and Computer Networks (SoftCOM)},
}
@InProceedings{Falahati03,
author = {S. Falahati and A. Svensson and M. Sternad and H. Mei},
title = {Adaptive Trellis-Coded Modulation over Predicted Flat Fading Channels},
booktitle = {Proceedings of the IEEE Vehicular Technology Conference (VTC-Fall)},
address={Orlando, Florida, USA},
year = {2003},
month = oct
}
@misc{adobeflash,
author = {{Adobe Corporation}},
title = {Flash Player},
note = {\url{http://www.adobe.com/products/flashplayer/} Visited 2011-09-27},
}
@INPROCEEDINGS{Henna09,
author={Henna, S.},
booktitle={Next Generation Mobile Applications, Services and Technologies, 2009. NGMAST '09. Third International Conference on},
title={A Throughput Analysis of {TCP} Variants in Mobile Wireless Networks},
year={2009},
month={sept.},
volume={},
number={},
pages={279 -284},
keywords={MANET routing protocol;TCP throughput analysis;bit
error;mobile wireless network;random packet loss;wireless
link;ad hoc networks;error statistics;mobile radio;routing
protocols;transport protocols;},
doi={10.1109/NGMAST.2009.71},
ISSN={},}
@INPROCEEDINGS{Bae02,
author={Sang Bae and Kaixin Xu and Sungwook Lee and Gerla, M.},
booktitle={Global Telecommunications Conference, 2002. GLOBECOM '02. IEEE},
title={Measured analysis of {TCP} behavior across multihop wireless and wired networks},
year={2002},
month={nov.},
volume={1},
number={},
pages={ 153 - 157 vol.1},
keywords={ 802.11 wireless LAN stations; Internet; TCP
behavior; TCP connections; TCP performance analysis;
bandwidth sharing; congestion control; database updating;
file transfer; mobile wireless networks; multihop 802.11
networks; multihop wired networks; multihop wireless
networks; reliability; testbed topology; throughput;
unfairness; wired infrastructure; wireless ad-hoc networks;
wireless channel; wireless transport protocols; IEEE
standards; Internet; ad hoc networks; network topology;
performance evaluation; telecommunication congestion
control; telecommunication standards; telecommunication
traffic; transport protocols; wireless LAN;},
doi={10.1109/GLOCOM.2002.1188060},
ISSN={ },}
@INPROCEEDINGS{Kulkarni09,
author={Kulkarni, P. and Sooriyabandara, M. and Lu Li},
booktitle={Proceedings of the IEEE Wireless Communications and Networking Conference (WCNC)},
title={Improving {TCP} Performance in Wireless Networks by Classifying Causes of Packet Losses},
year={2009},
month=apr,
volume={},
number={},
keywords={TCP E2E;TCP Reno;TCP performance;congestion
loss;packet loss;transport protocol;wireless loss;wireless
network;radio networks;telecommunication congestion
control;transport protocols;},
doi={10.1109/WCNC.2009.4917752},
ISSN={1525-3511},}
@INPROCEEDINGS{Lai10,
author={Chengdi Lai and Ka-Cheong Leung and Li, V.O.K.},
booktitle={Proceedings of the IEEE International Conference on Computer Communications (INFOCOM)},
title={Enhancing Wireless {TCP}: {A} Serialized-Timer Approach},
year={2010},
month=mar,
volume={},
number={},
keywords={congestion control;congestion response;congestive
losses;network overload;network resources;packet
reordering;packet retransmission;random losses;sequencing
control;serialized-timer approach;wireless TCP;wireless
networks;packet radio networks;telecommunication congestion
control;transport protocols;wireless channels;},
doi={10.1109/INFCOM.2010.5462205},
ISSN={0743-166X},}
@INPROCEEDINGS{Jubari10,
author={Al-Jubari, A.M. and Othman, M.},
booktitle={Information Technology (ITSim), 2010 International Symposium in},
title={A new delayed {ACK} strategy for {TCP} in multi-hop wireless networks},
year={2010},
month={june},
volume={2},
number={},
pages={946 -951},
keywords={IETF RFC 1122;TCP throughout;acknowledgment
generation;adaptive delay ACK algorithm;data packet;delayed
ACK strategy;multihop wireless networks;computer
networks;radio networks;transport protocols;},
doi={10.1109/ITSIM.2010.5561577},
ISSN={2155-897},}
@INPROCEEDINGS{Nehme03,
author={Nehme, A. and Phillips, W. and Robertson, W.},
booktitle={Electrical and Computer Engineering, 2003. IEEE CCECE 2003. Canadian Conference on},
title={The effect of reordering and dropping packets on {TCP} over a slow wireless link},
year={2003},
month={may},
volume={3},
number={},
pages={ 1555 - 1558 vol.3},
keywords={ TCP enhancement; buffer exhaustion; data
application; duplicate selective acknowledgment; network
simulator; packet dropping; packet reordering; receiver
advertised window limiting; transmission control protocol;
wireless link; wireless network; packet radio networks;
radio links; transport protocols;},
doi={10.1109/CCECE.2003.1226202},
ISSN={0840-7789 },}
@INPROCEEDINGS{Seddik09,
author={Seddik-Ghaleb, A. and Ghamri-Doudane, Y. and Senouci, S.-M.},
booktitle={Communication Systems and Networks and Workshops,
2009. COMSNETS 2009. First International},
title={{TCP WELCOME TCP} variant for Wireless Environment, Link losses, and COngestion packet loss ModEls},
year={2009},
month={jan.},
volume={},
number={},
pages={1 -8},
keywords={MANET;TCP WELCOME;TCP variant;congestion packet
loss models;data packet loss;link losses;loss
differentiation;mobile ad hoc networks;multihop wireless
networks;multiple loss differentiation algorithms;packet
loss cause;packet losses;wireless channel errors;wireless
environment;ad hoc networks;mobile radio;radio
links;transport protocols;},
doi={10.1109/COMSNETS.2009.4808863},
ISSN={},}
@INPROCEEDINGS{Hoque07,
author={Hoque, K. and Haque, R.R. and Hossain, M.A. and Farazi, M.S.F. and Hossain, G.},
booktitle={Proceedings of the 10th International Conference on Computer and Information Technology (ICCIT)},
title={Modeling and performance of {TCP} in a {MC-CDMA} system for {4G} communications},
year={2007},
month=dec,
volume={},
number={},
keywords={4G cellular network;4G wireless communication
system;MC-CDMA system;TCP based multicarrier access
technique;broadband communication;mobile commerce;transport
layer protocol;4G mobile communication;broadband
networks;cellular radio;code division multiple access;radio
networks;transport protocols;},
doi={10.1109/ICCITECHN.2007.4579450},
ISSN={},}
@INPROCEEDINGS{Hossain02,
author={Hossain, E. and Dong In Kim and Bhargava, V.K.},
booktitle={Proceedings of the IEEE International Conference on Communications (ICC)},
title={{TCP} performance under dynamic link adaptation in cellular multi-rate {WCDMA} networks},
year={2002},
month={},
volume={3},
number={},
pages={1834--1839},
abstract={ This paper models and analyzes the performance of
TCP (transmission control protocol) under joint rate and
power adaptation with constrained BER requirements for
downlink data transmission in a multi-cell VSF (variable
spreading factor) WCDMA system. The performance of TCP in a
wide-area Internet environment is evaluated by using
computer simulations considering user mobility, short-term
fading (i.e., multipath fading) and long-term fading (i.e.,
shadowing). The motivation is to explore the inter-layer
protocol interactions and to identify suitable transport and
radio link layer mechanisms to improve wireless TCP
performance in a cellular WCDMA environment.},
keywords={ TCP; WCDMA system; cellular environment; computer
simulations; constrained BER requirements; downlink data
transmission; dynamic link adaptation; inter-layer protocol
interactions; joint rate power adaptation; long-term fading;
multi-cell VSF; multi-cell systems; multi-rate systems;
multipath fading; performance; radio link layer mechanisms;
shadowing; short-term fading; transmission control protocol;
transport layer mechanisms; user mobility; variable
spreading factor; wide-area Internet environment; Internet;
adaptive control; cellular radio; code division multiple
access; data communication; error statistics; fading
channels; mobile computing; multipath channels; packet radio
networks; performance evaluation; power control;
telecommunication control; transport protocols;},
doi={10.1109/ICC.2002.997165},
ISSN={},}
@INPROCEEDINGS{Jurvansuu07,
author={Jurvansuu, M. and Prokkola, J. and Hanski, M. and Perala, P.},
booktitle={Proceedings of the IEEE International Conference Communications (ICC)},
title={{HSDPA} Performance in Live Networks},
year={2007},
month=jun,
volume={},
number={},
pages={467--471},
keywords={HSDPA;TCP;WCDMA;Web browsing;end-user
perspective;high-speed downlink packet access;link level
retransmissions;transmission control protocol;user datagram
protocol;voice over IP;wideband code division multiple
access;3G mobile communication;broadband networks;code
division multiple access;packet radio networks;radio access
networks;radio links;transport protocols;},
doi={10.1109/ICC.2007.83},
ISSN={},}
@INPROCEEDINGS{Prokkola09,
author={Prokkola, J. and Perala, P.H.J. and Hanski, M. and Piri, E.},
booktitle={Proceedings of the IEEE International Conference on Communications (ICC)},
title={{3G/HSPA} Performance in Live Networks from the End User Perspective},
year={2009},
month=jun,
volume={},
number={},
keywords={3G/HSPA network;3G/UMTS network;HARQ;TCP;channel
allocation;end user perspective;high speed packet
access;hybrid automatic repeat request;jitter
measurement;live network;mobile broadband Internet
access;one-way delay;uplink enhancement;3G mobile
communication;Internet;automatic repeat request;broadband
networks;channel allocation;radio access networks;transport
protocols;wireless channels;},
doi={10.1109/ICC.2009.5198575},
ISSN={1938-1883},}
@INPROCEEDINGS{Vulkan11,
author={Vulkan, C. and Heder, B.},
booktitle={Proceedings of the IEEE Vehicular Technology Conference (VTC Spring)},
title={Congestion Control in Evolved {HSPA} Systems},
year={2011},
month=may,
volume={},
number={},
keywords={3G architecture;3GPP technology;Iub
traffic;LTE;TCP;WCDMA;cell throughput;congestion control
mechanism;evolved HSPA systems;heterogeneous radio
systems;packet based flat architectural solution;radio layer
two protocol aspects;shared packet based transport
network;user data rate;3G mobile communication;Long Term
Evolution;code division multiple access;packet radio
networks;telecommunication congestion
control;telecommunication traffic;transport protocols;},
doi={10.1109/VETECS.2011.5956633},
ISSN={1550-2252},}
@INPROCEEDINGS{Lo04,
author={Lo, A. and Heijenk, G. and Niemegeers, I.},
booktitle={Computer Communications and Networks, 2004. ICCCN 2004. Proceedings. 13th International Conference on},
title={The impact of {UMTS/WCDMA} channel round-trip time variations on {TCP Vegas} throughput},
year={2004},
month={oct.},
volume={},
number={},
pages={67 -72},
keywords={TCP Vegas throughput;UMTS;Universal Mobile
Telecommunications System;WCDMA;cellular network;channel
round-trip time variation;congestion avoidance;data
loss;mobile Internet access;round-trip time
variation;transmission control protocol;3G mobile
communication;Internet;cellular radio;code division multiple
access;telecommunication congestion control;time-varying
channels;transport protocols;},
doi={10.1109/ICCCN.2004.1401590},
ISSN={1095-2055},}
@INPROCEEDINGS{Singh02,
author={Singh, A.K. and Iyer, S.},
booktitle={Proceedings of the International Workshop on Mobile and Wireless Communications Network (MWCN)},
title={{ATCP}: Improving {TCP} performance over mobile wireless environments},
address={Stockholm, Sweden},
year={2002},
month={},
volume={},
number={},
pages={239--243},
keywords={ 3-dupacks; ATCP; Freeze TCP; TCP Reno; TCP
performance improvement; WLAN environments; WWAN
environments; bit error rates; congestion control
mechanisms; fixed host; host mobility; mobile host; mobile
wireless environments; mobility induced disconnections;
network layer feedback; packet losses; performance
degradation; transmission control protocol; two-way data
transfers; wireless local area networks; wireless wide-area
networks; mobile radio; packet radio networks;
telecommunication congestion control; transport protocols;
wide area networks; wireless LAN;},
doi={10.1109/MWCN.2002.1045729},
ISSN={ },}
@INPROCEEDINGS{Li03,
author={Li, V.H. and Zhi-Qiang Liu},
booktitle={Proceedings of the IEEE Vehicular Technology Conference (VTC-Fall)},
title={{PET}: enhancing {TCP} performance over {3G} beyond networks},
year={2003},
month=oct,
volume={4},
number={},
pages={2302--2306},
keywords={ TCP flow control algorithms; active queue
management; beyond 3G networks; burst errors; congestion
control; distributed mode; enhancing TCP performance;
explicit congestion notification; integrated mode; links;
packet loss; wireless networks; wireline network; 3G mobile
communication; Internet; packet radio networks; queueing
theory; radio links; telecommunication congestion control;
transport protocols;},
doi={10.1109/VETECF.2003.1285940},
ISSN={1090-3038},}
@INPROCEEDINGS{Kumar08,
author={Kumar, D. and Barman, D. and Altman, E. and Kelif, J.-M.},
booktitle={Wireless Telecommunications Symposium, 2008. WTS 2008},
title={New cross-layer channel switching policy for {TCP} transmission on {3G UMTS} downlink},
year={2008},
month={april},
volume={},
number={},
pages={169 -176},
keywords={3G UMTS downlink;MAC;TCP transmission;cross-layer
channel switching policy;downlink data transmission;queue
size;terrestrial radio access network;transport
channels;universal mobile telecommunication system;3G mobile
communication;access protocols;data communication;queueing
theory;radio networks;subscriber loops;telecommunication
channels;telecommunication switching;transport protocols;},
doi={10.1109/WTS.2008.4547562},
ISSN={1934-5070},}
@INPROCEEDINGS{Yamamoto06,
author={Yamamoto, K. and Suzuki, H. and Ishikawa, N. and Miyake, M. and Inamura, H.},
booktitle={Proccedings of the International Conference on Computer Communications and Networks (ICCCN)},
title={A {TCP} Flow Control Scheme for {3G} Mobile Communication Networks},
year={2006},
month=oct,
volume={},
number={},
pages={229--236},
keywords={3G mobile communication networks;TCP flow control
scheme;data-rate variation;signal quality;wireless link;3G
mobile communication;radio links;telecommunication
congestion control;transport protocols;},
doi={10.1109/ICCCN.2006.286278},
ISSN={1095-2055},}
@INPROCEEDINGS{Leung04,
author={Leung, K.K. and Klein, T.E. and Mooney, C.F. and Haner, M.},
booktitle={Proceedings of the IEEE Vehicular Technology Conference (VTC-fall)},
title={Methods to improve {TCP} throughput in wireless networks with high delay variability},
year={2004},
month=sep,
volume={4},
number={},
pages={3015-3019},
keywords={ 3G network; RTO threshold; RTT; TCP throughput
gain; end-to-end performance; go-back-N retransmission
policies; high delay variability wireless networks;
out-of-order packet reception; packet timeout;
retransmission timeout threshold; selective repeat
retransmission policies; split TCP; spurious timeouts;
transmission control protocol; variable round-trip times; 3G
mobile communication; packet radio networks; transport
protocols;},
doi={10.1109/VETECF.2004.1400614},
ISSN={1090-3038 },}
@ARTICLE{Ren08,
author={Fengyuan Ren and Xiaomeng Huang and Feng Liu and Chuang Lin},
journal={IEEE Transactions on Wireless Communications},
title={Improving {TCP} Throughput over {HSDPA} Networks},
year={2008},
month=jun,
volume={7},
number={6},
pages={1993--1998},
keywords={HSDPA networks;High Speed Downlink Packet Access
network;Transmission Control Protocol;bandwidth
oscillation;cellular link resources;discrete-time stochastic
state space model;link adaptation techniques;queue
length;split connection Window Adaptation TCP Proxy;third
generation networks;3G mobile communication;cellular
radio;discrete time systems;queueing theory;state-space
methods;stochastic processes;transport protocols;},
doi={10.1109/TWC.2008.061007},
ISSN={1536-1276},}
@INPROCEEDINGS{Wang11,
author={Jingyuan Wang and Jiangtao Wen and Jun Zhang and
Yuxing Han},
booktitle={Proceedings of the IEEE International Conference on Computer Communications (INFOCOM)},
title="{TCP-FIT: An improved TCP congestion control algorithm and its performance}",
year={2011},
month=apr,
volume={},
number={},
pages={2894-2902},
keywords={3G networks;TCP congestion control;TCP-FIT control
algorithm;Wi-Fi;bandwidth-delay product;network
simulators;transport control protocol;wired Internet
application;wired line;wireless Internet
application;wireless fidelity;3G mobile
communication;Internet;telecommunication congestion
control;transport protocols;wireless LAN;},
doi={10.1109/INFCOM.2011.5935128},
ISSN={0743-166X},}
@ARTICLE{Alcaraz06b,
author={Alcaraz, J.J. and Cerdan, F. and Garcia-Haro, J.},
journal={Network, IEEE},
title="{Optimizing TCP and RLC interaction in the UMTS radio access network}",
year={2006},
month={march-april },
volume={20},
number={2},
pages={56 -64},
keywords={3GPP;ARQ algorithm;TCP;UMTS radio access
network;buffer management strategy;packet switched
services;reliable link layer protocol;transport protocol;3G
mobile communication;automatic repeat request;mobility
management (mobile radio);packet switching;radio access
networks;radio links;transport protocols;},
doi={10.1109/MNET.2006.1607897},
ISSN={0890-8044},}
@INPROCEEDINGS{Abed11,
author={Abed, G.A. and Ismail, M. and Jumari, K.},
booktitle={Modelling Symposium (AMS), 2011 Fifth Asia},
title={Appraisal of {Long Term Evolution} System with Diversified {TCP}'s},
year={2011},
month={may},
volume={},
number={},
pages={236 -239},
keywords={3GPP;4G wireless broadband technology;LTE;NS-2
Simulator;UMTS;diversified TCP;long term evolution
system;third generation partnership project;universal mobile
telecommunication system;3G mobile communication;4G mobile
communication;Long Term Evolution;transport protocols;},
doi={10.1109/AMS.2011.51},
ISSN={},}
@INPROCEEDINGS{Sandrasegaran10,
author={Sandrasegaran, K. and Reeves, S. and Ramli, H.A.M. and Basukala, R.},
booktitle={Proceedings of the Asia-Pacific Conference on Communications (APCC)},
title={Analysis of Hybrid {ARQ} in {3GPP} {LTE} systems},
year=2010,
month=nov,
volume={},
number={},
pages={418--423},
keywords={3GPP LTE systems;hybrid ARQ;packet
scheduler;wireless networks;3G mobile communication;Long
Term Evolution;automatic repeat request;},
doi={10.1109/APCC.2010.5679692},
ISSN={},}
@INPROCEEDINGS{Susitaival10,
author={Susitaival, R. and Wiemann, H. and \"Ostergaard, J. and Larmo, A.},
booktitle={Proceedings of IEEE Vehicular Technology Conference (VTC-Spring)},
title={Internet Access Performance in {LTE TDD}},
year={2010},
month=may,
volume={},
number={},
keywords={3GPP long term evolution;Internet access
performance;LTE TDD;TCP;downlink-heavy resource distribution
ratio;time division duplex uplink-downlink
configuration;transmission control protocol;uplink-heavy
ratio;3G mobile communication;Internet;telecommunication
traffic;time division multiplexing;transport protocols;},
doi={10.1109/VETECS.2010.5493986},
ISSN={1550-2252},}
@PhDThesis{LandstromPhd08,
author = {Sara Landström},
title = {{TCP/IP} Technology for Modern Network Environments},
year = {2008},
school = {Luleå University of Technology, Dept. of Computer Science and Engineering},
note = {Publication 2008:30, ISSN 1402-1544}
}
@article{Liu08a,
author = {Liu, Shao and Ba\c{s}ar, Tamer and Srikant, R.},
title = {{TCP-Illinois}: {A} loss- and delay-based congestion control algorithm for high-speed networks},
journal = {Performance Evaluation},
volume = {65},
issue = {6-7},
month = jun,
year = {2008},
issn = {0166-5316},
pages = {417--440},
numpages = {24},
url = {http://dl.acm.org/citation.cfm?id=1363365.1363461},
doi = {10.1016/j.peva.2007.12.007},
acmid = {1363461},
publisher = {Elsevier Science Publishers B. V.},
address = {Amsterdam, The Netherlands, The Netherlands},
keywords = {Congestion control, Fairness, Stability, Synchronization, TCP}
}
@INPROCEEDINGS{Shaikh10,
author={Shaikh, J. and Minhas, T.N. and Arlos, P. and Fiedler, M.},
booktitle={Proceedings of the International Workshop on Security and Communication Networks (IWSCN)},
address={Karlstad, Sweden},
title={Evaluation of delay performance of traffic shapers},
year={2010},
month=may,
volume={},
number={},
keywords={Internet;advance micro devices;coefficient of throughput variation;delay evaluation;delay performance evaluation;delay shaping;network emulation;protocol data unit;traffic shapers;Internet;delays;performance evaluation;protocols;telecommunication traffic;},
doi={10.1109/IWSCN.2010.5497994},
ISSN={},}
@book{mosP800,
author={{ITU-T}},
title={Recommendation {P.800}: Methods for objective and subjective assessment of quality},
year=1996,
publisher={ITU-T},
}
@book{pesqP862,
author={{ITU-T}},
title={Recommendation {P.862}: Perceptual evaluation of speech quality ({PESQ}), an objective method for end-to-end speech quality assessment of narrowband telephone networks and speech codecs},
year=2001,
publisher={ITU-T},
}
@book{emodelg107,
author={{ITU-T}},
title={Recommendation {G.107}: The {E-Model}, a computational model for use in transmission planning},
year=2000,
publisher={ITU-T},
}
@book{TIATSB116A,
author={{Telecommunications Industry Association}},
title={{TIA/TSB-116-A}: Voice quality recommendations for {IP} telephony.},
year=2006,
publisher={TIA},
note={Available at {http://www.tiaonline.org/standards/technology/voip/documents/TSB116-Afinalforglobal.pdf}},
}
@misc{Aronsson07,
author = {Daniel Aronsson},
title = "{Channel Estimation and Prediction from a Bayesian Perspective}",
howpublished = {Licentiate thesis, Signals and Systems, Uppsala University},
year = {2007},
month = jun,
submitter = {Stefan Alfredsson},
}
@book{Stallings07,
author = {William Stallings},
title = {Data and computer communications},
year = {2007},
isbn = {0-13-243310-9},
publisher = {Prentice Hall},
}
@INPROCEEDINGS{Ladha04,
author={Ladha, S. and Amer, P.D. and Caro, A., Jr. and Iyengar, J.R.},
booktitle={Global Telecommunications Conference ({GLOBECOM})},
title={On the prevalence and evaluation of recent {TCP} enhancements},
year={2004},
month=nov,
volume={3},
number={},
pages={1301--1307},
keywords={ Internet standards; SACK-based loss recovery algorithm; TCP enhancement deployment status; appropriate byte counting; bulk data transfers; congestion control; early retransmit method; initial congestion window increase; limited transmit method; loss recovery; selective acknowledgements; web servers; Internet; file servers; telecommunication congestion control; transport protocols;},
doi={10.1109/GLOCOM.2004.1378197},
ISSN={},}
@article{Dukkipati10,
author = {Dukkipati, Nandita and Refice, Tiziana and Cheng, Yuchung and Chu, Jerry and Herbert, Tom and Agarwal, Amit and Jain, Arvind and Sutin, Natalia},
title = {An argument for increasing {TCP}'s initial congestion window},
journal = {ACM SIGCOMM Computer Communication Review},
volume = {40},
issue = {3},
month = jun,
year = {2010},
issn = {0146-4833},
pages = {26--33},
numpages = {8},
url = {http://doi.acm.org/10.1145/1823844.1823848},
doi = {http://doi.acm.org/10.1145/1823844.1823848},
acmid = {1823848},
publisher = {ACM},
address = {New York, NY, USA},
keywords = {congestion control, internet measurements, tcp, web latency},
}
@misc{Ramachandran10,
author={S. Ramachandran and A. Jain},
year=2010,
title ={Web page stats: size and number of resources},
publisher={Google},
note={Available at \url{http://code.google.com/speed/articles/web-metrics.html} Visited 2011-12-28.},
}
@INPROCEEDINGS{Mondal07,
author={Mondal, A. and Kuzmanovic, A.},
booktitle={Proceedings of the 26th {IEEE International Conference on Computer Communications} ({INFOCOM})},
title={When {TCP} Friendliness Becomes Harmful},
year={2007},
month=may,
volume={},
number={},
pages={152--160},
keywords={Internet;TCP flows;TCP friendliness;network congestion;statistical multiplexing;Internet;statistical multiplexing;telecommunication congestion control;transport protocols;},
doi={10.1109/INFCOM.2007.26},
ISSN={0743-166X},}
@ARTICLE{Dettmer02,
author={Dettmer, R.},
journal={IEE Review},
title={The convergent phone},
year={2002},
month=jan,
volume={48},
number={1},
pages={23--27},
keywords={Internet telephony;PSTN;QoS levels;VoIP;WAN/LAN;network convergence;network management;network operators;reliability;telecoms business;Internet telephony;business communication;local area networks;wide area networks;},
doi={10.1049/ir:20020103},
ISSN={0953-5683},}
@INPROCEEDINGS{Parasad08,
author={Prasad, R.V. and Vijay, S. and Shankar, N. and Pawelczak, P. and Muralishankar, R. and Niemegeers, I.},
booktitle={Consumer Communications and Networking Conference, 2008. CCNC 2008. 5th IEEE}, title={A Holistic Study of {VoIP} Session Quality - The Knobs that Control},
year={2008},
month=jan.,
volume={},
number={},
pages={818--823},
keywords={G711 coded packets;Internet;VoIP packets;VoIP session quality;delay jitter;packet generation intervals;variable delay;Internet telephony;delays;jitter;packet switching;quality of service;},
doi={10.1109/ccnc08.2007.191},
ISSN={0197-2618},}
@misc{rfc3551,
author="H. Schulzrinne and S. Casner",
title={{RTP} Profile for Audio and Video Conferences with Minimal Control},
series="Request for Comments",
number="3551",
howpublished="RFC 3551 (Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2003,
month=jul,
url="http://www.ietf.org/rfc/rfc3551.txt",
}
@ARTICLE{Stivers09,
author={Tanya Stivers and N. J. Enfield and Penelope Brown and Christina Englert and Makoto Hayashi and Trine Heinemann and Gertie Hoymann and Federico Rossano and Jan Peter de Ruiter and Kyung-Eun Yoon and Stephen C. Levinson},
journal={Proceedings of the National Academy of Science (PNAS)},
title={Universals and cultural variation in turn-taking in conversation},
year={2009},
month=jun,
volume={106},
number={26},
doi={10.1073/pnas.0903616106},
}
@misc{mgen,
author={{U.S. Naval Research Laboratory}},
title={Multi-generator {MGEN}, version 4},
year =2008,
note={\url{http://cs.itd.nrl.navy.mil/work/mgen/} Visited 2012-03-14.},
}
@phdthesis{Aronsson11,
author={Daniel Aronsson},
title={{Channel Estimation and Prediction for {MIMO OFDM} Systems}},
school={Dept. Engineering Sciences, Signals and Systems, Uppsala University, Sweden},
year=2011,
pages=245,
isbn={987-91-506-2194-5},
}
@book{Kurose08,
author = {James F. Kurose and Keith W. Ross},
title = "{Computer Networking: A Top-Down Approach Featuring the Internet, 4th ed.}",
publisher = {Addison Wesley},
year = {2008},
note = {ISBN 978-0-321-49770-3}
}
@INPROCEEDINGS{Appenzeller04sizingrouter,
author = {Guido Appenzeller and Isaac Keslassy and Nick McKeown},
title = {Sizing Router Buffers},
booktitle = {Proceedings of {ACM SIGCOMM}},
year = {2004},
pages = {281--292},
publisher = {}
}
@article{Ganjali06,
author = {Yashar Ganjali and Nick McKeown},
title = {Update on Buffer Sizing in {Internet} Routers},
journal = {{ACM SIGCOMM} Computer Communications Review},
year = {2006},
pages = {67--70},
volume = {36:5},
month = oct,
}
@article{Andrew05,
author = {Lachlan Andrew and Tony Cui and Jinsheng Sun and Moshe Zukermanber and King-Tim Ko and Sammy Chan},
title = {Buffer Sizing for Nonhomogeneous {TCP} Sources},
journal = {{IEEE} Communication letters},
year = {2005},
pages = {567--569},
volume = {9:6},
month = jun,
}
@article{Dukkipati06,
author = {Dukkipati, Nandita and McKeown, Nick},
title = {Why flow-completion time is the right metric for congestion control},
journal = {SIGCOMM Computer Communication Review},
volume = {36},
issue = {1},
month = jan,
year = {2006},
issn = {0146-4833},
pages = {59--62},
numpages = {4},
url = {http://doi.acm.org/10.1145/1111322.1111336},
doi = {http://doi.acm.org/10.1145/1111322.1111336},
acmid = {1111336},
publisher = {ACM},
address = {New York, NY, USA},
}
@article{ETSITS123107,
author = "{3GPP}",
title = {Digital cellular telecommunications system (Phase 2+); {UMTS}; {LTE}; {Quality} of Service ({QoS}) concept and architecture ({3GPP TS} 23.107 version 10.0.0 Release 10)},
journal = {ETSI Technical Specification},
year = {2011},
month = mar,
}
@phdthesis{Ganjali07,
author={Yashar Ganjali},
title={{Buffer sizing in Internet routers}},
school={{Electrical engineering, Stanford University}},
year=2007,
month=mar,
pages=117,
}
@misc{rfc3439,
author="R. Bush and D. Meyer",
title={Some {Internet} Architectural Guidelines and Philosophy},
series="Request for Comments",
number="3439",
howpublished="RFC 3439 (Informational)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2002,
month=dec,
url="http://www.ietf.org/rfc/rfc3439.txt",
}
@book{Lin04,
author = {Lin, Shu and Costello, Daniel J.},
title = {Error Control Coding, Second Edition},
year = {2004},
isbn = {0130426725},
publisher = {Prentice-Hall, Inc.},
address = {Upper Saddle River, NJ, USA},
}
@INPROCEEDINGS{Xu11,
author={Wei Xu and Yinlong Xu and Xiaohu Wu and Kaiqian Ou},
booktitle={International Conference on Information Networking ({ICOIN})},
title={Modeling {TCP SACK} steady state performance in lossy networks},
year={2011},
month={jan.},
volume={},
number={},
pages={278--283},
keywords={TCP sack steady state performance modeling;TCP selective acknowledgment;lossy networks;packet loss;round trip time;transmission control protocol;packet switching;transport protocols;},
doi={10.1109/ICOIN.2011.5723193},
ISSN={1976-7684},}
@INPROCEEDINGS{Cardwell00,
author={Cardwell, N. and Savage, S. and Anderson, T.},
booktitle={INFOCOM 2000. Nineteenth Annual Joint Conference of the IEEE Computer and Communications Societies. Proceedings. IEEE}, title={Modeling {TCP} latency},
year={2000},
month={mar},
volume={3},
number={},
pages={1742--1751},
keywords={Internet;TCP latency;Web measurements;connection establishment;data transfer latency;modeling;packet loss rate;performance;round trip time;slow start;startup effects;steady-state throughput;transfer size;Internet;delay estimation;performance evaluation;telecommunication traffic;transport protocols;},
doi={10.1109/INFCOM.2000.832574},
ISSN={},}
@article{Svensson07,
author = {Arne Svensson},
title = {An Introduction to Adaptive {QAM} Modulation Schemes for Known and Predicted Channels},
journal = {Proceedings of the {IEEE}},
year = 2007,
volume = 95,
number = 12,
month = {December}
}
@book{Jakes74,
author={William C. Jakes},
title={Microwave mobile communications},
year=1994,
publisher={IEEE Press, Piscataway, N.J., USA},
isbn = {0-7803-1069-1}
}
@ARTICLE{Falahati04,
AUTHOR = "Sorour Falahati and Arne Svensson and Torbjörn Ekman and Mikael Sternad",
TITLE = "Adaptive Modulation Systems for Predicted Wireless Channels.",
JOURNAL = {IEEE Transactions on Communications},
PAGES = {307-316},
YEAR = {2004} }
@INPROCEEDINGS{Moller04,
author={Moller, N. and Johansson, K.H. and Hjalmarsson, H.},
booktitle={Proceedings of the IEEE Conference on Decision and Control (CDC)},
title={Making retransmission delays in wireless links friendlier to {TCP}},
address={Bahamas},
year={2004},
month=dec,
volume={5},
number={},
pages={5134--5139},
keywords={ TCP control actions; end-to-end congestion control; heterogeneous communication networks; link-layer retransmissions; packet delay; radio links; retransmission delays; transport protocol; uncertain time-varying traffic load; wireless links; data communication; telecommunication congestion control; transport protocols;},
doi={10.1109/CDC.2004.1429622},
ISSN={0191-2216},}
@INPROCEEDINGS{Inamura04,
author={Inamura, H. and Takahashi, O. and Nakano, H. and Ishikawa, T. and Shigeno, H.},
booktitle={Distributed Computing Systems, 2004. Proceedings. 24th International Conference on},
title={Impact of layer two {ARQ} on {TCP} performance in {W-CDMA} networks},
year={2004},
month={},
volume={},
number={},
pages={284--291},
keywords={3rd generation cellular wireless network technology; ARQ parameter; RLC; TCP; TCP parameter; TCP throughput; W-CDMA; delay-jitter; link layer retransmission protocol; radio link control; 3G mobile communication; cellular radio; code division multiple access; jitter; transport protocols;},
doi={10.1109/ICDCS.2004.1281593},
ISSN={1063-6927 },}
@misc{rfc1035,
author="P.V. Mockapetris",
title="{Domain names - implementation and specification}",
series="Request for Comments",
number="1035",
howpublished="RFC 1035 (Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1987,
month=nov,
alt_note="Updated by RFCs 1101, 1183, 1348, 1876, 1982, 1995, 1996, 2065, 2136, 2181, 2137, 2308, 2535, 2845, 3425, 3658, 4033, 4034, 4035, 4343, 5936, 5966",
url="http://www.ietf.org/rfc/rfc1035.txt",
}
@misc{rfc1157,
author="J.D. Case and M. Fedor and M.L. Schoffstall and J. Davin",
title={Simple Network Management Protocol ({SNMP})},
series="Request for Comments",
number="1157",
howpublished="RFC 1157 (Historic)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1990,
month=may,
url="http://www.ietf.org/rfc/rfc1157.txt",
}
@misc{rfc5681,
author="M. Allman and V. Paxson and E. Blanton",
title="{TCP Congestion Control}",
series="Request for Comments",
number="5681",
howpublished="RFC 5681 (Draft Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2009,
month=sep,
url="http://www.ietf.org/rfc/rfc5681.txt",
}
@misc{rfc3649,
author="S. Floyd",
title={HighSpeed {TCP} for Large Congestion Windows},
series="Request for Comments",
number="3649",
howpublished="RFC 3649 (Experimental)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2003,
month=dec,
url="http://www.ietf.org/rfc/rfc3649.txt",
}
@book{Tanenbaum03,
author = {Andrew S. Tanenbaum},
title = {Computer Networks, 4th ed.},
publisher = {Prentice-Hall International, Inc.},
year = {2003},
note = {ISBN: 0-13-066102-3},
}
@article{FIXME,
author={FIXME},
title={FIXME},
journal={FIXME},
year={FIXME}
}
@Book{ISO7498,
author = "{ISO}",
title = "{ISO/IEC 7498:1984}: Information processing systems -- Open Systems Interconnection -- Basic Reference Model",
publisher = {International Organization for Standardization},
year = "1984",
URL = "http://www.iso.org/iso/iso_catalogue/catalogue_ics/catalogue_detail_ics.htm?csnumber=14252",
}
@misc{TR25853,
author = {3GPP},
title = "{TS 25.853: V4.0.0 (2001-03). Technical Specification Group (TSG) RAN; Delay Budget within the Access Stratum}",
year = {2001},
url = {http://www.3gpp.org/ftp/Specs/html-info/25853.htm},
note={Available at \url{http://www.3gpp.org/ftp/Specs/html-info/25853.htm}},
}
@misc{netperf,
author = {Rick A. Jones},
title = {Netperf},
note = {Available at \url{http://www.netperf.org} Visited 2011-12-28.}
}
@misc{DN20091214,
author={Karoline Hoppe},
year=2009,
title ={{Telia Sonera} lanserar supernät},
howpublished={Dagens Nyheter 2009-12-14},
note={\url{http://www.dn.se/ekonomi/telia-sonera-lanserar-supernat} Visited 2012-03-14.},
}
@misc{TS20091214,
author={{Telia Sonera}},
year=2009,
title ={{TeliaSonera} first in the world with {4G} services},
howpublished={Telia Sonera press release 2009-12-14},
note={\url{http://www.teliasonera.com/en/newsroom/press-releases/2009/12/teliasonera-first-in-the-world-with-4g-services/} Visited 2012-03-14.},
}
@book{ITUR:2134,
author={{ITU-R}},
title="{Report ITU-R M.2134: Requirements related to technical performance for IMT-Advanced radio interface(s)}",
year=2008,
publisher={ITU-R},
note={Available at \url{http://www.itu.int/pub/R-REP-M.2134-2008/en} Visited 2012-03-13.}
}
@book{ITUR:2135,
author={{ITU-R}},
title="{Report ITU-R M.2135-1: Guidelines for evaluation of radio interface technologies for IMT-Advanced}",
year=2009,
publisher={ITU-R},
note={Available at \url{http://www.itu.int/pub/R-REP-M.2135-1-2009} Visited 2012-03-14.}
}
@misc{freebsd,
author={Justin T. Gibbs},
year=2011,
title ="{The FreeBSD Project}",
publisher={The FreeBSD Foundation},
note={\url{http://www.freebsd.org} Visited 2012-03-14.},
}
@book{Dahlman11,
author = {Erik Dahlman and Stefan Parkvall and Johan Sköld},
title = {{4G LTE/LTE-Advanced} for Mobile Broadband},
year=2011,
publisher={Academic Press},
note = {ISBN 978-0-12-385489-6}
}
@misc{Sternad11p,
author={Mikael Sternad},
year=2011,
month=aug,
title ="Personal communication"
}
@article{GSMEFR,
author = {{ETSI}},
title = "{Digital cellular telecommunications system (Phase 2+) (GSM); Enhanced Full Rate (EFR) speech transcoding (GSM 06.60 version 8.0.1 Release 1999)}",
journal = {ETSI Technical Specification},
year = {2010},
month = nov,
}
@article{Faezah09,
author = {Faezah, Jasman and Sabira, Khatun},
title = {Adaptive Modulation for {OFDM} Systems},
journal = {International Journal of Communication Networks and Information Security ({IJCNIS})},
volume = {1},
number = {2},
year = {2009},
ee = {http://ijcnis.kust.edu.pk/article/view/4773},
bibsource = {DBLP, http://dblp.uni-trier.de}
}
@misc{ITUICT,
author = "{ITU-D}",
title = "{ITU} Statistics; {I}nternet users per 100 inhabitants 2010",
note = {Available at \url{http://www.itu.int/ict/statistics} Visited 2011-09-27.}
}
@misc{HTML5,
author = "{World Wide Web Consortium}",
editor = {Hickson, Ian and Hyatt, David},
key = {Hickson},
howpublished = {W3C Working Draft},
note = {\url{http://www.w3.org/TR/html5/} Visited 2011-12-28.},
title = {{HTML 5}: A Vocabulary and Associated {APIs} for {HTML and XHTML}},
url = {http://www.w3.org/TR/html5/},
year = 2011,
abstract = {This specification defines the 5th major revision of the core language of the World Wide Web: the Hypertext Markup Language (HTML). In this version, new features are introduced to help Web application authors, new elements are introduced based on research into prevailing authoring practices, and special attention has been given to defining clear conformance criteria for user agents in an effort to improve interoperability.}
}
@misc{WIPFinalreport09,
author = "Mikael Sternad",
title = "{The SSF Wireless IP Project: Final Report}",
year = 2009,
month = oct,
note = {Submitted to the SSF. Available online at \url{http://www.signal.uu.se/Research/PCCWIP/WirelessIP_Finalreport.pdf} . Visited 2011-10-10.}
}
@misc{WINNERFinalreport08,
author = "Uwe Herzog",
title = "{WINNER II Final Report (IST-4-027756 D7.1.5 v1.0)}",
year = 2008,
month = feb,
note = {Available online at \url{http://www.ist-winner.org/deliverables.html} . Visited 2011-10-10.}
}
@misc{WINNER,
author = "{WWRF}",
title = "{Wireless World Initiative New Radio}",
note = {Available online at \url{http://www.ist-winner.org}. Visited 2011-10-10.}
}
@misc{WWRF,
author = "{WWRF}",
title = "{The Wireless World Research Forum}",
note = {Available online at \url{http://www.wireless-world-research.org}. Visited 2011-10-10.}
}
@misc{WWI,
author = "{WWI}",
title = "{The Wireless World Initiative}",
note = {Available online at \url{http://www.wireless-world-initiative.org/}. Visited 2011-10-10.}
}
@book{ITUMIS11,
author = "{ITU ICT Data and Statistics Division}",
title = "Measuring the Information Society",
publisher = "International Telecommunication Union",
year = 2011,
month = sep,
isbn = {92-61-13801-2},
note = {Available at \url{http://www.itu.int/pub/D-IND-ICTOI-2011/en} or \url{http://www.itu.int/ITU-D/ict/publications/idi/2011/Material/MIS\_2011\_without\_annex\_5.pdf} Visited 2011-10-11.}
}
@inproceedings{Xu04,
author = {Xu, Lisong and Harfoush, Khaled and Rhee, Injong},
booktitle={Proceedings of the IEEE International Conference on Computer Communications (INFOCOM)},
address={Hong Kong},
title = {Binary Increase Congestion Control ({BIC}) for Fast Long-Distance Networks.},
url = {http://dblp.uni-trier.de/db/conf/infocom/infocom2004.html#XuHR04},
year = 2004,
ee = {http://www.ieee-infocom.org/2004/Papers/52_4.PDF},
}
@article{Ha08,
Author = {Ha, Sangtae and Rhee, Injong and Xu, Lisong},
ISSN = {01635980},
Journal = {ACM SIGOPS Operating Systems Review},
Number = {5},
Pages = {64--74},
Title = {{CUBIC}: a new {TCP}-friendly high-speed {TCP} variant},
Volume = {42},
Year = {2008},
}
@INPROCEEDINGS{Leith04,
author = {Douglas Leith and Robert Shorten},
title = {{H-TCP: TCP} for high-speed and long-distance networks},
booktitle = {Proceedings of the 2nd Workshop on Protocols for Fast
Long Distance Networks (PFLDnet)},
address={Argonne, USA},
month=feb,
year = {2004}
}
@article{Kuzmanovic06,
author = {Kuzmanovic, Aleksandar and Knightly, Edward W.},
title = {{TCP-LP}: low-priority service via end-point
congestion control},
journal = {IEEE/ACM Transactions on Networking},
volume = {14},
issue = {4},
month = aug,
year = {2006},
issn = {1063-6692},
pages = {739--752},
numpages = {14},
url = {http://dx.doi.org/10.1109/TNET.2006.879702},
doi = {http://dx.doi.org/10.1109/TNET.2006.879702},
acmid = {1217652},
publisher = {IEEE Press},
address = {Piscataway, NJ, USA},
keywords = {TCP, TCP-LP, TCP-transparency, available
bandwidth, service prioritization},
}
@ARTICLE{Kelly03,
title = {{Scalable TCP: Improving Performance in
Highspeed Wide Area Networks}},
author = {Tom Kelly},
year = {2003},
month = apr,
journal = {ACM Computer Communication Review},
volume = {32},
number = {2},
url = {http://www.cl.cam.ac.uk/research/dtg/www/files/publications/public/arb33/scalable_improve_hswan.pdf},
owner = {Alastair Beresford (arb33)},
}
@ARTICLE{Grieco04,
author = {Luigi A. Grieco and Saverio Mascolo},
title = {Performance evaluation and comparison of
{Westwood+}, {New Reno}, and {Vegas} {TCP} congestion control},
journal = {ACM Computer Communication Review},
year = {2004},
pages = {25--38}
}
@inproceedings{Baiocchi07,
author = {A. Baiocchi and A. P. Castellani and F. Vacirca},
title = "{YeAH-TCP: Yet Another Highspeed TCP}",
address={Los Angeles, USA},
booktitle = {Proceedings of Fifth International Workshop on Fast Long Distance Networks (PFLDnet)},
year = 2007,
}
@ARTICLE{Caini04,
author = {Carlo Caini and Rosario Firrincieli},
title = {{TCP Hybla}: a {TCP} enhancement for heterogeneous networks},
journal = {International Journal of Satellite Communications and Networking},
year = {2004},
volume = {22}
}
@article{Carbone10,
author = {Carbone, Marta and Rizzo, Luigi},
title = {Dummynet revisited},
journal = {SIGCOMM Compututer Communcation Review},
volume = {40},
issue = {2},
month = apr,
year = {2010},
issn = {0146-4833},
pages = {12--20},
numpages = {9},
url = {http://doi.acm.org/10.1145/1764873.1764876},
doi = {http://doi.acm.org/10.1145/1764873.1764876},
acmid = {1764876},
publisher = {ACM},
address = {New York, NY, USA},
keywords = {emulation, network protocols, performance
evaluation, wireless emulation},
}
@article{Chesterfield04,
title = "Transport Level Optimisations for Streaming Media Over Wide-area Wireless Networks",
author = "Julian Chesterfield and Rajiv Chakravorty and Suman Banerjee and Pablo Rodriguez and Ian Pratt and Jon
Crowcroft",
year=2004,
month=mar,
journal="Proceedings of Wireless Optimization Workshop (WiOpt'04), Cambridge, UK",
}
@INPROCEEDINGS{Cheng03,
author={Jung-Fu Cheng},
booktitle={IEEE Global Telecommunications Conference (GLOBECOM'03)},
title={On the coding gain of incremental redundancy over chase combining},
year={2003},
month={dec.},
volume={1},
number={},
pages={107--112},
doi={10.1109/GLOCOM.2003.1258212},
ISSN={},}
@INPROCEEDINGS{Meriaux10,
author={Mériaux, F. and Kieffer, M.},
booktitle={Acoustics Speech and Signal Processing (ICASSP), 2010 IEEE International Conference on},
title="{Robust IP and UDP-lite header recovery for packetized multimedia transmission}",
year={2010},
month={march},
volume={},
number={},
pages={2358--2361},
doi={10.1109/ICASSP.2010.5496115},
ISSN={1520-6149},}
@INPROCEEDINGS{Ravichandran08,
author={Ravichandran, A. and Tacca, M. and Welzl, M. and Fumagalli, A.},
booktitle={Proceedings of the IEEE Global Telecommunications Conference (GLOBECOM)},
title={{LN-MAC}: a Cross-layer Explicit Loss Notification Solution for {TCP} over {IEEE 802.11}},
year={2008},
month=dec,
volume={},
number={},
pages={1--5},
keywords={IEEE 802.11;Internet
access;LN-MAC;TCP;WiFi;cross-layer explicit loss
notification solution;packet loss;IEEE standards;transport
protocols;wireless LAN;},
doi={10.1109/GLOCOM.2008.ECP.1015},
ISSN={1930-529X},}
@misc{RFC4653,
author="S. Bhandarkar and A. L. N. Reddy and M. Allman and E. Blanton",
title={Improving the Robustness of {TCP} to Non-Congestion Events},
series="Request for Comments",
number="4653",
howpublished="RFC 4653 (Experimental)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2006,
month=aug,
url="http://www.ietf.org/rfc/rfc4653.txt",
}
@INPROCEEDINGS{Gurtov03,
author={Gurtov, A. and Ludwig, R.},
booktitle={Proceedings of the IEEE International Conference on Computer Communications (INFOCOM)},
title={Responding to spurious timeouts in {TCP}},
year={2003},
volume={3},
number={},
pages={2312--2322},
abstract={ Delays on Internet paths, especially including
wireless links, can be highly variable. On the other hand, a
current trend for modern TCPs is to deploy a fine-grain
retransmission timer with a lower minimum timeout value than
1 s suggested by RFC2988. Spurious TCP timeouts cause
unnecessary retransmissions and congestion control back-off.
The Eifel algorithm detects spurious TCP timeouts and
recovers by restoring the connection state saved before the
timeout. This paper presents an enhanced version of the
Eifel response to spurious timeouts and illustrates its
performance benefits on paths with a high delay-bandwidth
product. The refinements concern the following issues (1) an
efficient operation in presence of packet losses (2)
appropriate restoration of congestion control, and (3)
adapting the retransmit timer to avoid further spurious
timeouts. In our simulations the Eifel algorithm on paths
with a high delay-bandwidth product can increase throughput
by up to 250% and at the same decrease the load on the
network by 3%. The proposed response also shows adequate
performance on heavily congested paths.},
keywords={ Eifel algorithm; Eifel response; Internet path
delay; congested path performance; congestion control
restoration; connection state restoration; high
delay-bandwidth product; minimum timeout value; network
throughput; packet losses; retransmission timer; spurious
TCP timeout response; wireless link; Internet; delays; radio
links; telecommunication congestion control; transport
protocols;},
doi={10.1109/INFCOM.2003.1209251},
ISSN={0743-166X},}
@INPROCEEDINGS{Chebrolu11,
author={Chebrolu, K. and Mishra, A.},
booktitle={Communication Systems and Networks (COMSNETS),
2011 Third International Conference on},
title={Loss behavior analysis and its application in design of link quality metrics},
year={2011},
month={jan.},
volume={},
number={},
pages={1--10},
keywords={ETX link quality metric;ROMA metric;application
layer;higher-layer protocols;link behaviour;loss behavior
analysis;loss rate measurements;packet count function;packet
loss;routing layer;routing protocols;stability based link
quality metric;wireless 802.11a testbed;wireless
losses;radio links;routing protocols;wireless LAN;},
doi={10.1109/COMSNETS.2011.5716415},
ISSN={},}
@INPROCEEDINGS{Lefevre01,
author={Lefevre, F. and Vivier, G.},
booktitle={Proceedings of IEEE Vehicular Technology Conference (VTC-spring)},
title={Optimizing {UMTS} link layer parameters for a {TCP} connection},
year={2001},
month=may,
volume={4},
number={},
pages={2318--2322},
keywords={ARQ;Internet;RLQ protocol;TCP
connection;UMTS;acknowledged mode;link layer
parameters;mobile wireless systems;radio link
control;retransmission mechanism;transmission control
protocol;automatic repeat request;mobile
radio;optimisation;transport protocols;},
doi={10.1109/VETECS.2001.944014},
ISSN={},}
@INPROCEEDINGS{Lott07,
author={Lott, C. and Milenkovic, O. and Soljanin, E.},
booktitle={Proceedings of the IEEE Workshop on Information Theory for Wireless Networks},
title={Hybrid {ARQ}: Theory, State of the Art and Future Directions},
year={2007},
month=jul,
volume={},
number={},
pages={1--5},
keywords={forward error correction;hybrid ARQ;mobile phone
networks;throughput performance;wireless channels;automatic
repeat request;forward error correction;mobile
radio;telecommunication network reliability;wireless
channels;},
doi={10.1109/ITWITWN.2007.4318035},
ISSN={},}
@ARTICLE{Ratnam03,
author = {Karunaharan Ratnam and Ibrahim Matta},
title = {{WTCP}: An Efficient Mechanism for Improving Wireless Access to {TCP} Services},
journal = {International Journal of Communication Systems},
year = {2003},
volume = {16},
pages = {47--62}
}
@article{Singh07,
author = {Jatinder Pal Singh and
Yan Li and
Nicholas Bambos and
Ahmad Bahai and
Baowen Xu and
Gerd Zimmermann},
title = {{TCP} Performance Dynamics and Link-Layer Adaptation Based
Optimization Methods for Wireless Networks},
journal = {IEEE Transactions on Wireless
Communications},
volume = {6},
number = {5},
year = {2007},
pages = {1864-1879},
ee = {http://dx.doi.org/10.1109/TWC.2007.360388},
bibsource = {DBLP, http://dblp.uni-trier.de}
}
@article{Chebrolu05,
author = {Kameswari Chebrolu and
Bhaskaran Raman and
Ramesh R. Rao},
title = {A Network Layer Approach to Enable {TCP} over Multiple Interfaces},
journal = {Wireless Networks},
volume = {11},
number = {5},
year = {2005},
pages = {637-650},
ee = {http://dx.doi.org/10.1007/s11276-005-3518-5},
bibsource = {DBLP, http://dblp.uni-trier.de}
}
@inproceedings{Oliveira09,
author = {de Oliveira, Danillo Franscys Borges and Munaretto, Anelise and Ziviani, Artur and Fonseca, Mauro},
title = {A proxy-based architecture for {TCP} to mitigate packet loss on wireless networks},
booktitle = {Proceedings of the 2nd IFIP conference on Wireless Days},
year = {2009},
isbn = {978-1-4244-5660-4},
location = {Paris, France},
pages = {332--337},
numpages = {6},
url = {http://dl.acm.org/citation.cfm?id=1834116.1834177},
acmid = {1834177},
address = {Piscataway, NJ, USA},
}
@INPROCEEDINGS{Jung06,
author={Hakyung Jung and Nakjung Choi and Yongho Seok and Taekyoung Kwon and Yanghee Choi},
booktitle={IEEE International Conference on Communications (ICC)},
title={Augmented {Split-TCP} over Wireless {LANs}},
year={2006},
month=jun,
volume={12},
number={},
pages={5420--5425},
keywords={},
doi={10.1109/ICC.2006.255523},
ISSN={8164-9547},}
@misc{rfc3135,
author="J. Border and M. Kojo and J. Griner and G. Montenegro and Z. Shelby",
title={Performance Enhancing Proxies Intended to Mitigate Link-Related Degradations},
series="Request for Comments",
number="3135",
howpublished="RFC 3135 (Informational)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2001,
month=jun,
url="http://www.ietf.org/rfc/rfc3135.txt",
}
@inproceedings{Wang11b,
author = {Wang, Zhaoguang and Qian, Zhiyun and Xu, Qiang and Mao, Zhuoqing and Zhang, Ming},
title = {An untold story of middleboxes in cellular networks},
booktitle = {Proceedings of SIGCOMM},
year = {2011},
isbn = {978-1-4503-0797-0},
address = {Toronto, Ontario, Canada},
pages = {374--385},
numpages = {12},
url = {http://doi.acm.org/10.1145/2018436.2018479},
doi = {http://doi.acm.org/10.1145/2018436.2018479},
acmid = {2018479},
keywords = {cellular data network, firewall, middlebox, nat, tcp performance},
}
@misc{Touch11,
author="Joe Touch",
title={Automating the Initial Window in {TCP}},
howpublished="Internet Draft (work in progress; expires 2012-07-17)",
publisher="IETF tcpm working group",
year=2011,
month=jul,
note={Available at \url{http://tools.ietf.org/html/draft-touch-tcpm-automatic-iw-02}. Accessed 2012-03-14}}
}
@misc{rfc3580,
author="P. Congdon and B. Aboba and A. Smith and G. Zorn and J. Roese",
title={{IEEE} {802.1X} Remote Authentication Dial In User Service ({RADIUS}) Usage Guidelines},
series="Request for Comments",
number="3580",
howpublished="RFC 3580 (Informational)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2003,
month=sep,
url="http://www.ietf.org/rfc/rfc3580.txt",
}
@INPROCEEDINGS{Tan06,
author={Tan, K. and Song, J. and Zhang, Q. and Sridharan, M.},
booktitle={Proceedings of the IEEE International Conference on Computer Communications (INFOCOM)},
title={A Compound {TCP} Approach for High-Speed and Long Distance Networks},
year={2006},
month=apr,
volume={},
number={},
pages={1--12},
keywords={},
doi={10.1109/INFOCOM.2006.188},
ISSN={0743-166X},}
@misc{rfc4340,
author="E. Kohler and M. Handley and S. Floyd",
title={Datagram Congestion Control Protocol ({DCCP})},
series="Request for Comments",
number="4340",
howpublished="RFC 4340 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2006,
month=mar,
url="http://www.ietf.org/rfc/rfc4340.txt",
}
@misc{rfc4341,
author="S. Floyd and E. Kohler",
title={Profile for Datagram Congestion Control Protocol ({DCCP}) Congestion Control ID 2: {TCP}-like Congestion Control},
series="Request for Comments",
number="4341",
howpublished="RFC 4341 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2006,
month=mar,
url="http://www.ietf.org/rfc/rfc4341.txt",
}
@misc{rfc4342,
author="S. Floyd and E. Kohler and J. Padhye",
title={Profile for Datagram Congestion Control Protocol ({DCCP}) Congestion Control ID 3: {TCP}-Friendly Rate Control ({TFRC})},
series="Request for Comments",
number="4342",
howpublished="RFC 4342 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2006,
month=mar,
note="Updated by RFC 5348",
url="http://www.ietf.org/rfc/rfc4342.txt",
}
@misc{rfc5348,
author="S. Floyd and M. Handley and J. Padhye and J. Widmer",
title={{TCP} Friendly Rate Control ({TFRC}): Protocol Specification},
series="Request for Comments",
number="5348",
howpublished="RFC 5348 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2008,
month=sep,
url="http://www.ietf.org/rfc/rfc5348.txt",
}
@inproceedings{Wang05a,
author = {Li-Chun Wang and Ching-Hao Lee},
title = {A {TCP}-Physical Cross-Layer Congestion Control Mechanism for the Multirate {WCDMA} System Using Explicit Rate Change Notification},
booktitle = {Proceedings of the 19th International Conference on Advanced Information Networking and Applications (AINA)},
year = {2005},
pages = {449-452},
doi={10.1109/AINA.2005.69},
}
@misc{Welzl04,
author="Michael Welzl",
title={{TCP} Corruption Notification Options},
howpublished="Internet Draft (work in progress; expired dec 2004)",
publisher="IETF",
year=2004,
month=jun,
note={Available at \url{http://tools.ietf.org/html/draft-welzl-tcp-corruption-00}. Accessed 2011-11-24}}
}
@article{Welzl05,
author = {Welzl, Michael},
title = {Passing corrupt data across network layers: an overview of recent developments and issues},
journal = {EURASIP Journal of Applied Signal Processing},
volume = {2005},
month = jan,
year = {2005},
issn = {1110-8657},
pages = {242--247},
numpages = {6},
url = {http://dx.doi.org/10.1155/ASP.2005.242},
doi = {http://dx.doi.org/10.1155/ASP.2005.242},
acmid = {1287109},
publisher = {Hindawi Publishing Corp.},
address = {New York, NY, United States},
keywords = {DCCP, TCP, UDP Lite, checksum, link layer ARQ, wireless links},
}
@inproceedings{Welzl08,
title = {{TCP/IP} over {IEEE 802.11b WLAN}: the Challenge of Harnessing Known-Corrupt Data},
author = {Michael Welzl and Mattia Rossi and Andrea Fumagalli and Marco Tacca},
year = {2008},
month = may,
doi = {http://dx.doi.org/10.1109/ICC.2008.59},
researchr = {http://researchr.org/publication/WelzlRFT08},
pages = {280--284},
booktitle = {Proceedings of the IEEE International Conference on Communications (ICC)},
address = {Beijing, China},
}
@inproceedings{Weingartner09,
author = {Weingartner, Elias and vom Lehn, Hendrik and Wehrle, Klaus},
title = {A performance comparison of recent network simulators},
booktitle = {Proceedings of the IEEE International Conference on Communications (ICC)},
address = {Dresden, Germany},
year = {2009},
abstract = {A widespread methodology for performance analysis
in the field of communication systems engineering is network
simulation. While ns-2 has established itself as virtually the
standard network simulation tool, other network simulators have
gained more and more attention during the last years. In this
paper, we briefly survey new developments in the field of network
simulation and conduct a performance comparison study by
implementing an identical simulation set-up in five simulators,
namely ns-2, OMNet++, ns-3, SimPy and JiST/SWANS. Our results
reveal large differences according to both run-time performance
and memory usage.},
url =
{http://ds.informatik.rwth-aachen.de/members/weingaertner/publications/2008-06-Weingaertner-ICC-NetworkSimulatorComparison}
}
@article{Elocla10,
author = {El-Ocla, Hosam},
title = {{TCP CERL}: congestion control enhancement over wireless networks},
journal = {Wireless Networks},
volume = {16},
issue = {1},
month = {January},
year = {2010},
issn = {1022-0038},
pages = {183--198},
numpages = {16},
url = {http://dx.doi.org/10.1007/s11276-008-0123-4},
doi = {http://dx.doi.org/10.1007/s11276-008-0123-4},
acmid = {1741438},
publisher = {Kluwer Academic Publishers},
address = {Hingham, MA, USA},
keywords = {Congestive loss, Lossy networks, Random loss, TCP
NewReno, TCP Reno, TCP Vegas, TCP Veno, TCP Westwood,
Transmission Control Protocol (TCP) congestion control, Wireless
networks},
}
@inproceedings{Gao08,
author = {Dawei Gao and Yantai Shu and Li Yu and M. Y. Sanadidi and Mario Gerla},
title = {{TCP SPC}: Statistic Process Control for Enhanced Transport over Wireless Links},
booktitle={Proceedings of the IEEE Global Telecommunications Conference (GLOBECOM)},
address={New Orleans, USA},
year = {2008},
pages = {5453-5457},
ee = {http://dx.doi.org/10.1109/GLOCOM.2008.ECP.1041},
bibsource = {DBLP, http://dblp.uni-trier.de}
}
@article{Biaz05,
author = {Saad Biaz and
Nitin H. Vaidya},
title = {{'{D}e-randomizing'} congestion losses to improve {TCP} performance over wired-wireless networks},
journal = {IEEE/ACM Transactions on Networking},
volume = {13},
number = {3},
year = {2005},
pages = {596--608},
ee = {http://dx.doi.org/10.1109/TNET.2005.850205},
bibsource = {DBLP, http://dblp.uni-trier.de}
}
@inproceedings{Garcia08,
author = {Johan Garcia and Per Hurtig and Anna Brunstrom},
title = {{KauNet}: A Versatile and Flexible Emulation System},
booktitle = {Proceedings of the 5th Swedish National Computer Networking Workshop (SNCNW)},
year = 2008,
month = apr
}
@article{Mondal08,
author = {Mondal, Amit and Kuzmanovic, Aleksandar},
title = {Removing exponential backoff from TCP},
journal = {SIGCOMM Computer Communication Review},
volume = {38},
issue = {5},
month = {September},
year = {2008},
issn = {0146-4833},
pages = {17--28},
numpages = {12},
url = {http://doi.acm.org/10.1145/1452335.1452338},
doi = {http://doi.acm.org/10.1145/1452335.1452338},
acmid = {1452338},
publisher = {ACM},
address = {New York, NY, USA},
keywords = {TCP, congestion collapse, exponential backoff algorithm, implicit packet conservation principle},
}
@inproceedings{Stone00,
author = {Stone, Jonathan and Partridge, Craig},
title = {When the CRC and TCP checksum disagree},
booktitle = {Proceedings of ACM SIGCOMM},
year = {2000},
isbn = {1-58113-223-9},
location = {Stockholm, Sweden},
pages = {309--319},
numpages = {11},
url = {http://doi.acm.org/10.1145/347059.347561},
doi = {http://doi.acm.org/10.1145/347059.347561},
acmid = {347561},
publisher = {ACM},
address = {New York, NY, USA},
}
@inproceedings{Deak09,
author = {De\'{a}k, Csaba and Farkas, L\'{o}r\'{a}nt and
Hom\'{a}nyi, Gergely and Vulk\'{a}n, Csaba},
title = {Downlink streaming performance over evolved {HSPA}},
booktitle = {Proceedings of the 2009 International Conference on
Wireless Communications and Mobile Computing (IWCMC)},
year = {2009},
isbn = {978-1-60558-569-7},
address = {Leipzig, Germany},
pages = {1035--1039},
numpages = {5},
url = {http://doi.acm.org/10.1145/1582379.1582607},
doi = {http://doi.acm.org/10.1145/1582379.1582607},
acmid = {1582607},
keywords = {evolved HSPA, mean opinion score, simulation,
streaming},
}
@inproceedings{Korhonen07,
author = {Korhonen, Jari and Frossard, Pascal},
title = {Bit-error Resilient Packetization for Streaming {H.264/AVC} Video},
booktitle = {Proceedings of the International workshop on mobile video},
year = {2007},
month=sep,
isbn = {978-1-59593-779-7},
address = {Augsburg, Bavaria, Germany},
pages = {25--30},
numpages = {6},
url = {http://doi.acm.org/10.1145/1290050.1290057},
doi = {http://doi.acm.org/10.1145/1290050.1290057},
acmid = {1290057},
keywords = {H.264/AVC, UDP-Lite, bit error resilience, video streaming},
}
@conference{Sperschneider02,
title={Error Resilient Source Coding with Differential Variable Length Codes and its Application to {MPEG Advanced Audio Coding}},
author={Sperschneider, Ralph and Homm, Daniel and Chambat, Louis-Henry},
booktitle={Proceedings of the Audio Engineering Society Convention 112},
month={4},
year={2002},
url={http://www.aes.org/e-lib/browse.cfm?elib=11319}
}
@article{Persson07,
author={Hannes Persson and Anna Brunstrom and Tony Ottosson},
title={Utilizing Cross-Layer Information to Improve Performance in {JPEG2000} Decoding},
journal={Advances in Multimedia},
year=2007,
volume=2007,
publisher={Hindawi},
}
@book{Foley97,
author={James D. Foley},
title={Computer graphics: principles and practice},
publisher={Addison-Wesley Systems Programming Series},
year=1997,
isbn={0-201-84840-6}
}
@misc{RFC1958,
author="B. Carpenter",
title={Architectural Principles of the {Internet}},
series="Request for Comments",
number="1958",
howpublished="RFC 1958 (Informational)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=1996,
month=jun,
url="http://www.ietf.org/rfc/rfc1958.txt",
}
@inproceedings{Honda11,
author = {Honda, Michio and Nishida, Yoshifumi and Raiciu, Costin and Greenhalgh, Adam and Handley, Mark and Tokuda, Hideyuki},
title = {Is it still possible to extend {TCP}?},
booktitle = {Proceedings of the 2011 ACM SIGCOMM conference on Internet measurement},
series = {IMC '11},
year = {2011},
isbn = {978-1-4503-1013-0},
location = {Berlin, Germany},
pages = {181--194},
numpages = {14},
url = {http://doi.acm.org/10.1145/2068816.2068834},
doi = {http://doi.acm.org/10.1145/2068816.2068834},
acmid = {2068834},
publisher = {ACM},
address = {New York, NY, USA},
keywords = {TCP, measurements, middleboxes, protocol design},
}
@inproceedings{Mcreary05,
author = {McCreary, Doug and Li, Kang and Watterson, Scott A. and Lowenthal, David K.},
doi = {10.1117/12.592252},
booktitle = {Proceedings of Multimedia Computing and Networking},
address={San Jose, CA, USA},
pages = {126--130},
publisher = {SPIE},
title = {{TCP-RC: a receiver-centered TCP protocol for delay-sensitive applications}},
url = {http://dx.doi.org/10.1117/12.592252},
year = {2005}
}
@INPROCEEDINGS{Malkowski03,
AUTHOR = {Malkowski, M. and Heier, S.},
TITLE = {Interaction between {UMTS MAC} Scheduling and {TCP} Flow Control Mechanisms},
YEAR = {2003},
MONTH = {Apr},
PAGES = {1373-1376},
ADDRESS = {Beijing, China},
BOOKTITLE = {Proceedings of 2003 International Conference on Communication Technology},
AFFILIATION = {Department of Communication Networks
(ComNets), Faculty 6, RWTH Aachen University},
ISBN = {7-563-50686-1},
KEYWORDS = {UMTS, Radio Interface Protocols, MAC
Scheduler, QoS, WWW Traffic Model,
UMTS Simulator},
URL = {http://www.comnets.rwth-aachen.de}
}
@INPROCEEDINGS{Chahed03,
author={Chahed, T. and Canton, A.-F. and Elayoubi, S.-E.},
booktitle={Proceeings of the IEEE International Conference on Communications (ICC)},
title={End-to-end {TCP} performance in {W-CDMA / UMTS}},
year={2003},
month=may,
volume={1},
number={},
pages={71-75},
keywords={ BER; QoS; SIR; UMTS; W-CDMA; bit error ratio; despread
bit energy; end-to-end TCP performance; interference density
ratio; power control algorithm; quality of service;
signal-to-interference ratio; transport control protocol;
universal mobile telecommunication system; wideband code division
multiple access; wired mobile sections; wireless mobile sections;
3G mobile communication; code division multiple access; error
statistics; interference (signal); power control; quality of
service; transport protocols;},
doi={10.1109/ICC.2003.1204145},
ISSN={},}
@INPROCEEDINGS{Rossi03,
author={Rossi, M. and Scaranari, L. and Zorzi, M.},
booktitle={Proceedings of the IEEE Vehicular Technology Conference (VTC Fall)},
title={On the {UMTS RLC} parameters setting and their impact on higher layers performance},
year={2003},
address={Orlando, Florida, USA},
month=oct,
volume={3},
number={},
pages={1827--1832},
keywords={ RLC parameters setting; TCP flow; UMTS; Universal
Mobile Telecommunications System; channel metrics; higher layers
performance; link layer parameters; radio link control; 3G mobile
communication; radio links; transport protocols;},
doi={10.1109/VETECF.2003.1285341},
ISSN={1090-3038},}
@ARTICLE{Assaad06,
author={Assaad, M. and Zeghlache, D.},
journal={IEEE Journal on Selected Areas in Communications},
title={Cross-Layer design in {HSDPA} system to reduce the {TCP} effect},
year={2006},
month=mar,
volume={24},
number={3},
pages={614--625},
doi={10.1109/JSAC.2005.862414},
ISSN={0733-8716},}
@INPROCEEDINGS{Bucknell07,
author={Bucknell, P. and Baker, M.P.J.},
booktitle={Proceedings of the International Symposium on Wireless Communication Systems (ISWCS)},
title={Cross-Layer Interactions in {UMTS} Evolved {UTRAN} ({E-UTRAN})},
year={2007},
month=oct,
volume={},
number={},
pages={262--266},
keywords={HARQ error recovery;UMTS Layer 1 protocols;UMTS Layer 2
protocol;UMTS evolved UTRAN;cross-layer interactions;hybrid ARQ
control signalling;medium access control;radio link layer
control;throughput performance improvements;throughput
reductions;transmission errors;3G mobile communication;access
protocols;telecommunication network topology;},
doi={10.1109/ISWCS.2007.4392342},
ISSN={},}
@INPROCEEDINGS{Svoboda08,
author={Svoboda, P. and Ricciato, F.},
booktitle={Modeling and Optimization in Mobile, Ad Hoc, and
Wireless Networks and Workshops, 2008. WiOPT 2008. 6th
International Symposium on},
title={Analysis and detection of bottlenecks via {TCP} footprints
in live {3G} networks},
year={2008},
month={april},
volume={},
number={},
pages={37--42},
keywords={TCP footprints;bottleneck analysis;bottleneck
detection;capacity bottleneck;live 3G networks;mobile network;3G
mobile communication;telecommunication traffic;transport
protocols;},
doi={10.1109/WIOPT.2008.4586033},
ISSN={},}
@inproceedings{Liu08b,
author = {Liu, Xin and Sridharan, Ashwin and Machiraju, Sridhar and Seshadri, Mukund and Zang, Hui},
title = {Experiences in a {3G} network: interplay between the wireless channel and applications},
booktitle = {Proceedings of the 14th ACM International Conference on Mobile Computing and Networking(MOBICOM)},
year = {2008},
isbn = {978-1-60558-096-8},
address = {San Francisco, California, USA},
pages = {211--222},
numpages = {12},
url = {http://doi.acm.org/10.1145/1409944.1409969},
doi = {http://doi.acm.org/10.1145/1409944.1409969},
acmid = {1409969},
keywords = {1xEV-DO, 3G, CDMA, DRC, SINR, TCP, cellular, cross-layer, measurement, mobility, proportional fair (PF)},
}
@INPROCEEDINGS{Lin03b,
author={Haitao Lin and Das, S.K.},
booktitle={Proceedings of the IEEE Personal, Indoor and Mobile Radio Communications
conference (PIMRC)},
title={Performance study of {TCP/RLP/MAC} in next generation {CDMA} systems},
year={2003},
month=sep,
volume={1},
number={},
pages={648--652},
keywords={ 3G networks; 4G networks; Internet; MAC; RLP; TCP;
cdma2000 system; code division multiple access; correlated fading
channel model; link layer retransmissions; media access control;
next generation CDMA systems; radio link protocol; transport
control protocol; wireless data network; wireless link errors;
wireless links; 3G mobile communication; 4G mobile communication;
Internet; code division multiple access; data communication;
fading channels; radio links; transport protocols;},
doi={10.1109/PIMRC.2003.1264352},
ISSN={},}
@ARTICLE{Tan08,
author={Wee Lum Tan and Fung Lam and Wing Cheong Lau},
journal={IEEE Transactions on Mobile Computing},
title={An Empirical Study on the Capacity and Performance of {3G} Networks},
year={2008},
month=jun,
volume={7},
number={6},
pages={737 -750},
keywords={3G network;call admission control;multiple commercial
third-generation network;network resources allocation;3G mobile
communication;telecommunication congestion
control;telecommunication traffic;},
doi={10.1109/TMC.2007.70788},
ISSN={1536-1233},}
@inproceedings{Garcia07,
title = {{KauNet}: improving reproducibility for wireless and mobile research},
booktitle = {Proceedings of the 1st international workshop on System evaluation for mobile platforms},
author = {J. Garcia and E. Conchon and T. Perennou and A. Brunstrom},
address = {San Juan, Puerto Rico},
year = {2007},
pages = {21--26}
}
@book{Murty04,
author ={C. S. R. Murthy and B. S. Manoj},
title ={Ad Hoc Wireless Networks},
publisher = {Prentice Hall},
year = 2004,
note = {ISBN 0-13-147023-X}
}
@INPROCEEDINGS{Holma06,
author={Holma, H. and Kuusela, M. and Malkamaki, E. and Ranta-aho, K. and Chen Tao},
booktitle={Proceedings of the IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC)},
address={Helsinki, Finland},
title={{VOIP} over {HSPA} with {3GPP} Release 7},
year={2006},
month=sep,
volume={},
number={},
keywords={3GPP Release 5;3GPP Release 6;3GPP Release 7;HSPA;VoIP;circuit switched voice calls;high speed packet access;mobile receiver algorithms;packet optimization;radio network algorithms;voice-over-IP;3G mobile communication;Internet telephony;circuit switching;code division multiple access;},
doi={10.1109/PIMRC.2006.254111},
ISSN={},}
@ARTICLE{Floyd01c,
author={Floyd, S. and Paxson, V.},
journal={IEEE/ACM Transactions on Networking},
title={Difficulties in simulating the {I}nternet},
year={2001},
month=aug,
volume={9},
number={4},
pages={392--403},
keywords={Internet simulation;collaborative research;heterogeneous network;network congestion level;network simulator;network traffic;protocols;simulation parameter space;Internet;digital simulation;protocols;research initiatives;reviews;telecommunication congestion control;telecommunication traffic;},
doi={10.1109/90.944338},
ISSN={1063-6692},}
@INPROCEEDINGS{Bateman10,
author={Bateman, M. and Bhatti, S.},
booktitle={Proceedings of the 24th IEEE International Conference on Advanced Information Networking and Applications (AINA)},
title={{TCP} Testing: {H}ow Well Does ns2 Match Reality?},
address={Perth, Australia},
year={2010},
month=apr,
volume={},
number={},
pages={276--284},
keywords={TCP testing;high bandwidth-delay product;linux kernel;ns2 simulator;transmission control protocol;transport protocol;Linux;digital simulation;program testing;transport protocols;},
doi={10.1109/AINA.2010.133},
ISSN={1550-445X},}
@article{Floyd03,
author = {Floyd, Sally and Kohler, Eddie},
title = {Internet research needs better models},
journal = {SIGCOMM Computer Communication Review},
volume = {33},
issue = {1},
month = {January},
year = {2003},
issn = {0146-4833},
pages = {29--34},
numpages = {6},
url = {http://doi.acm.org/10.1145/774763.774767},
doi = {http://doi.acm.org/10.1145/774763.774767},
acmid = {774767},
publisher = {ACM},
address = {New York, NY, USA},
}
@phdthesis{Hurtig11,
author={Per Hurtig},
title={Transport-Layer Performance for Applications and Technologies of the Future Internet},
school={Dept. of Computer Science, Karlstad University, Sweden},
year=2012,
month=feb,
pages={248},
note = {Karlstad University Studies 2011:65, ISBN 978-91-7063-404-8}
}
@misc{rfc6182,
author="A. Ford and C. Raiciu and M. Handley and S. Barre and J. Iyengar",
title={Architectural Guidelines for Multipath {TCP} Development},
series="Request for Comments",
number="6182",
howpublished="RFC 6182 (Informational)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2011,
month=mar,
url="http://www.ietf.org/rfc/rfc6182.txt",
}
@INPROCEEDINGS{Yang11,
author={Peng Yang and Wen Luo and Lisong Xu and Deogun, J. and Ying Lu},
booktitle={Proceedings of the 31st International Conference on Distributed Computing Systems (ICDCS)},
title={{TCP} Congestion Avoidance Algorithm Identification},
year={2011},
month=jun,
volume={},
number={},
pages={310--321},
keywords={AIMD algorithm;BIC algorithm;CTCP algorithm;CUBIC algorithm;Internet traffic;TCP congestion avoidance algorithm identification;Web servers;heterogeneous congestion control;homogeneous congestion control;Internet;telecommunication congestion control;transport protocols;},
doi={10.1109/ICDCS.2011.27},
ISSN={1063-6927},}
@article{Schervish87,
title={A Review of Multivariate Analysis},
author={Mark J. Schervish},
journal={Statistical Science},
publisher={The Institute of Mathematical Statistics},
volume=2,
number=4,
year=1987,
pages={396--413},
doi={10.1214/ss/1177013111},
note={Available at \url{http://projecteuclid.org/euclid.ss/1177013111}. Visited 2012-01-04.}
}
@article{Shannon48,
author = {C. Shannon},
title = {A Mathematical Theory of Communication},
journal = {The Bell System Technical Journal},
year = {1948},
volume = {27},
month = oct
}
@misc{wipweb,
author={Mikael Sternad},
year=2011,
title ={Publications and Workshops within the {Wireless IP} Project},
howpublished={Project web page},
note={\url{http://www.signal.uu.se/Publications/wip.html}. Visited 2012-03-14.}
}
@article{Lochin11,
author = {Lochin, Emmanuel and Pérennou, Tanguy and Dairaine, Laurent},
affiliation = {CNRS; LAAS, 7 avenue du colonel Roche, 31077 Toulouse, France},
title = {When should {I} use network emulation?},
journal = {Annals of Telecommunications},
publisher = {Springer Paris},
issn = {0003-4347},
year=2011,
month=jul,
url = {http://dx.doi.org/10.1007/s12243-011-0268-5},
doi = {10.1007/s12243-011-0268-5},
note = {(online pre-print, doi:10.1007/s12243-011-0268-5)}
}
@article{Benitez10,
author = {L\'{o}pez-Ben\'{\i}tez, Miguel and Bernardo, Francisco and Vu\v{c}evi\'{c}, Nemanja and Umbert, Anna},
title = {Real-Time emulation of heterogeneous wireless networks with end-to-edge quality of service guarantees: the {AROMA} testbed},
journal = {EURASIP Journal on Wireless Communication and Networking},
volume = {2010},
month = apr,
year = {2010},
issn = {1687-1472},
pages = {22:1--22:12},
articleno = {22},
numpages = {12},
url = {http://dx.doi.org/10.1155/2010/961302},
doi = {http://dx.doi.org/10.1155/2010/961302},
acmid = {1864698},
publisher = {Hindawi Publishing Corp.},
address = {New York, NY, United States},
}
@article{Marin10,
author = {Marin, C\'{e}dric and Leprovost, Yann and Kieffer, Michel and Duhamel, Pierre},
title = {Robust {MAC}-lite and soft header recovery for packetized multimedia transmission},
journal = {IEEE Transactions on Communication},
volume = {58},
issue = {3},
month = {March},
year = {2010},
issn = {0090-6778},
pages = {775--784},
numpages = {10},
url = {http://dx.doi.org/10.1109/TCOMM.2010.03.080303},
doi = {http://dx.doi.org/10.1109/TCOMM.2010.03.080303},
acmid = {1821092},
publisher = {IEEE Press},
address = {Piscataway, NJ, USA},
keywords = {MAP estimation, codes, communication systems, decoding, protocols},
}
@ARTICLE{IEEE80211-std,
author={{IEEE LAN/MAN Standards Committee}},
journal={IEEE Std 802.11-2007},
title="{IEEE Standard for Information Technology- Telecommunications and Information Exchange Between Systems- Local and Metropolitan Area Networks- Specific Requirements Part 11: Wireless LAN Medium Access Control (MAC) and Physical Layer (PHY) Specifications}",
year={2007},
month=jun,
volume={},
number={},
doi={10.1109/IEEESTD.2007.373646},
ISSN={},
}
@ARTICLE{IEEE80211n-std,
author={{IEEE LAN/MAN Standards Committee}},
journal={IEEE Std 802.11n-2009},
title="{IEEE Standard for Information technology--Telecommunications and information exchange between systems--Local and metropolitan area networks--Specific requirements Part 11: Wireless LAN Medium Access Control (MAC) and Physical Layer (PHY) Specifications Amendment 5: Enhancements for Higher Throughput}",
year={2009},
volume={},
number={},
doi={10.1109/IEEESTD.2009.5307322},
ISSN={},
}
@book{Schwartz05,
title={Mobile wireless communications},
author={Schwartz, Mischa},
isbn={9780521843478},
lccn={2004049268},
googleurl={http://books.google.se/books?id=rAiyyKM9ppoC},
year={2005},
publisher={Cambridge University Press}
}
@misc{ITUabout,
author="{International Telecommunication Union}",
title="{About ITU}",
note={\url{http://www.itu.int/en/about/Pages/default.aspx}. Visited 2012-03-12.},
year=2012,
}
@misc{RFC3416,
author="R. Presuhn",
title="{Version 2 of the Protocol Operations for the Simple Network Management Protocol (SNMP)}",
series="Request for Comments",
number="3416",
howpublished="RFC 3416 (Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2002,
month=dec,
url="http://www.ietf.org/rfc/rfc3416.txt",
}
@misc{RFC3782,
author="S. Floyd and T. Henderson and A. Gurtov",
title="{The NewReno Modification to TCP's Fast Recovery Algorithm}",
series="Request for Comments",
number="3782",
howpublished="RFC 3782 (Proposed Standard)",
publisher="IETF",
organization="Internet Engineering Task Force",
year=2004,
month=apr,
url="http://www.ietf.org/rfc/rfc3782.txt",
}
@inproceedings{Alfredsson13wowmom,
author = {Stefan Alfredsson and Giacomo Del Giudice and Johan Garcia and Anna Brunstrom and Luca De Cicco and Saverio Mascolo},
title = {{Impact of TCP Congestion Control on Bufferbloat in Cellular Networks}},
booktitle = {Proceedings of the 14th IEEE International Symposium on a World of Wireless, Mobile and Multimedia Networks (WoWMoM)},
year = {2013},
_mon = jun,
address={Madrid, Spain}
url={http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-27779}
}
@inproceedings{Alfredsson13sncnw,
author = {Stefan Alfredsson and Giacomo Del Giudice and Johan Garcia and Anna Brunstrom and Luca De Cicco and Saverio Mascolo},
title = {{Observations of Bufferbloat in Swedish Cellular Networks}},
booktitle = {Proceedings of the 9th Swedish National Computer Networking Workshop (SNCNW)},
year = {2013},
address={Lund, Sweden}
}
@inproceedings{Alfredsson13softcom,
author = {Stefan Alfredsson},
title = {{A Case Study of Recent Mifare Classic Field Deployments From an~(In-)Security Perspective}},
booktitle = {Proceedings of the 21st International Conference on Software, Telecommunications and Computer Networks (SoftCOM)},
year = {2013},
address={Primo\v{s}ten, Croatia}
}
@phdthesis{Alfredsson12diss,
author={Stefan Alfredsson},
title={{A Cross-Layer Perspective on Transport Protocol Performance in Wireless Networks}},
school={Dept. of Computer Science, Karlstad University, Sweden},
year=2012,
month=may,
pages={201},
note = {Karlstad University Studies 2012:7, ISBN 978-91-7063-411-6}
}