Papers on RTP

A large number of papers, masters theses and reports related to RTP have been published. Below is a small selection.

Delivering Voice over IP Networks
Daniel Minoli and Emma Minoli
Wiley, 1998. ISBN 0-471254827

IPng and the TCP/IP Protocols
Stephen A. Thomas
John Wiley, 1996. ISBN 0-471-13088-5

This book includes coverage of Internet Protocol Version 6 (IPv6), its extension headers, flows and security features; ICMP for IPv6, including neighbor discovery, duplicate address detection, and address autoconfiguration; latest UDP and TCP enhancements such as timestamps, window scaling and header prediction; full coverage of the Open Shortest Path First protocol, including multicast routing; RIP and its support for IPv6; IDRP for routing between autonomous systems; real-time applications using RTP; reserving resources with RSVP; domain name service for IPv6, including dynamic updates; configuring hosts automatically using DHCP; the transition to IPng from the current IP; migrating Novel and OSI networks to IPng.
Timer Reconsideration for Enhanced RTP Scalability PDF
Jonathan Rosenberg and Henning Schulzrinne
Proc. of Infocom, (San Francisco, California), March/April 1998.

RTP, the Real Time Transport Protocol, has gained widespread acceptance as the transport protocol for voice and video on the Internet. Its companion control protocol, the Real Time Control Protocol (RTCP), is used for loose session control, QoS reporting, and media synchronization, among other func-tions. The RTP specification describes an algorithm for determin-ing the RTCP packet transmission rate at a host participating in a multicast RTP session. This algorithm was designed to allow RTP to be used in sessions with anywhere from one to a mil-lion members. However, we have discovered several problems with this algorithm when used with very large groups with rapidly changing group membership. One problem is the flood of RTCP packets which occurs when many users join a multicast RTP ses-sion at nearly the same time. To solve this problem, we present a novel adaptive timer algorithmcalled reconsideration. We present a mathematical analysis of this algorithm, and demonstrate that it performs extremely well, reducing the congestion problem by several orders of magnitude. We also back up these results with simulation.
Transmission of MPEG-2 Streams over Non-Guaranteed Quality of Service Networks
Andrea Basso, G. L. Cash and M. Reha Civanlar
Proc. of Picture Coding Symposium, (Berlin, Germany), Sept. 1997.

The explosive growth of the Internet and the intranets attracted a great deal of attention to the implementations and performances of networked multimedia services, which involve the transport of real-time multimedia streams over non-guaranteed quality of service (QoS) networks. In this paper, we discuss some issues related to the transport of MPEG-2 streams over such networks by means of the most recent transport protocols compliant with the Real-Time Transport Protocol (RTP) [1] defined by the Internet Engineering Task Force (IETF). MPEG-2 encoded audio and video transmission is important for several applications including high quality video-on-demand as a part of information-on-demand and high quality video conferencing using the existing network infrastructures.
Timer reconsideration for enhanced RTP scalability
Jonathan Rosenberg and Henning Schulzrinne
Internet Draft, Internet Engineering Task Force, July 1997. Work in progress.

RTP, the Real Time Transport Protocol, has gained widespread acceptance as the transport protocol for voice and video on the Internet. It provides services such as timestamping, sequence numbering, and payload identification. It also contains a control component, the Real Time Control Protocol (RTCP), which is used for loose session control, QoS reporting, and media synchronization, among other functions. The RTP specification describes an algorithm for determining the RTCP packet transmission rate at a host participating in a multicast RTP session. This algorithm was designed to allow RTP to be used in sessions with anywhere from one to a million members. However, we have discovered several problems with this algorithm when used with very large groups with rapidly changing group membership. One problem is the flood of RTCP packets which occurs when many users join a multicast RTP session at nearly the same time. To solve this problem, we present a novel adaptive timer algorithm called reconsideration. We present a mathematical analysis of this algorithm, and demonstrate that it performs extremely well, reducing the congestion problem by several orders of magnitude. We also back up these results with simulation.
Issues and options for an aggregation service within RTP
Jonathan Rosenberg and Henning Schulzrinne
November 1996.

This memorandum discusses the issues and options involved in the design of a new transport protocol for multiplexed voice within a single packet. The intended application is the interconnection of devices which provide 'trunking' or long distance telephone service over the Internet. Such devices have many voice connections simultaneously between them. Multiplexing them into the same connection improves on the efficiency, enables the use of low bitrate voice codecs, and improves scalability. Options and issues concerning timestamping, payload type identification, length indication, and channel identification are discussed. Several possible header formats are identified, and their efficiencies are compared.
Extensions to RTP to support Mobile Networking
Kevin Brown and Suresh Singh
3rd Intl. Workshop on Mobile Multimedia Communications, Sept. 25-27, 1996, Princeton, NJ

In this paper, we identify limitations of the real-time protocol (RTP) regarding mobile networking and low-speed links and propose solutions to these problems. In particular, we propose schemes to limit the bandwidth used on the wireless link by RTP data messages and RTCP control messages.
The Realtime Transport Protocol
R. Klein
Technical Report, Northwest Alliance for Computational Science and Engineering, Oct. 1996.
Audio and video over packet networks - issues, architecture and protocols
Henning Schulzrinne
Interop'94, (Paris, France), Oct. 1994.
Network support for dynamically scaled multimedia data streams
Don Hoffman, Michael Speer, and Gerard Fernando
Proceedings of the 4th International Workshop on Network and Operating System Support for Digital Audio and Video, (Lancaster, U.K.), pp. 251-262, Lancaster University, Nov. 1993; Lecture Notes in Computer Science 846.
As multimedia applications such as video-on-demand and video conferencing become more common, the classes of systems and networks participating in these applications are becoming more diverse. Where several endpoints need to access the same video stream simultaneously, multicast protocols are often employed to reduce the duplication of network traffic across common links. Previous literature has discussed the concept that hierarchical media encodings may be used to achieve some form of stream scalability within a multicast network. This paper discusses the networking issues associated with encoding hierarchical streams and mapping them to a multimedia transport service interface.
Piloting of multimedia integrated communications for European researchers (MICE)
P. T. Kirstein, M. J. Handley, and M. A. Sasse
Proceedings of the International Networking Conference (INET), (San Francisco, California), pp. DCA-1 - DCA-12, Internet Society, Aug. 1993
The paper describes multimedia conferencing and describes the facilities currently available. It discusses briefly the activities which require standardization and the progress in this direction to date. It gives an overview of the MICE project, which utilizes existing conferencing rooms, workstations, codecs and software, and existing network infrastructure, to offer researchers conferencing facilities within Europe, as well as a link to the US. The goals of the project, its achievements to date, and problems encountered are discussed in detail. Finally, we outline forthcoming activities.
RTP: The real-time transport protocol
Henning Schulzrinne
MCNC 2nd Packet Video Workshop, vol. 2, (Research Triangle Park, North Carolina), Dec. 1992.
Multicast congestion control in the distribution of variable bit rate video.
Ian Wakeman and Jon Crowcroft
Computer Science Department, University College London, Jan. 1994.
Internet services: from electronic mail to real-time multimedia
Henning Schulzrinne
in Proc. of KIVS (Kommunikation in Verteilten Systemen) (K. Franke, U. Hübner, and W. Kalfa, eds.), Informatik aktuell, (Chemnitz, Germany), pp. 21--34, Gesellschaft für Informatik, Springer Verlag, Feb. 1995.
QOS for real-time services: playout delay and application control
Henning Schulzrinne
in Proc. of 46th RACE Concertation Meeting (RCM), (Brussels, Belgium), Mar. 1995.
When can we unplug the phone and the radio?
H. Schulzrinne
in Proc. International Workshop on Network and Operating System Support for for Digital Audio and Video (NOSSDAV), Lecture Notes in Computer Science (LNCS), (Durham, New Hampshire), pp. 183--184, Springer, Apr. 1995.
Kommunikationsunterstützung für multimediale Anwendungen
P. Harrschar
Diplomarbeit, Institut für Telematik, Fakultät für Informatik, Universität Karlsruhe, Karlsruhe, Germany, July 1995.
Bewertung von adaptiven Ausspielalgorithmen für paketvermittelte Audiodaten (Evaluation of adaptive playout algorithms for packet audio)
C. Sieckmeyer
Studienarbeit, Dept. of Electrical Engineering, TU Berlin, Berlin, Germany, Oct. 1995.
vic: A flexible framework for packet video
S. McCanne and V. Jacobson
in Proc. of ACM Multimedia '95, Nov. 1995.
An application level video gateway
E. Amir, S. McCanne, and H. Zhang
in Proc. of ACM Multimedia, (San Francisco, California), Nov. 1995.
An application level video gateway
E. Amir
Master's thesis, University of California, Berkeley, Berkeley, California, Dec. 1995.
Dynamic QoS control of multimedia applications based on RTP
I. Busse, B. Deffner, and H. Schulzrinne
Computer Communications, Jan. 1996.
Adaptive speech compression for packet communication systems
D. T. Magill
in Conference record of the IEEE National Telecommunications Conference, (??), pp. 29D--1 -- 29D--5, ?? 1973.
Specifications for the network voice protocol NVP
D. Cohen
RFC 741, Internet Engineering Task Force, Nov. 1977.

Last updated by Henning Schulzrinne