RTP Multiplexing

Only the first draft is being actively pursued.
Tunneling multiplexed Compressed RTP ('TCRTP')
B. Thompson, T. Koren, D. Wing
November 2000.
This document describes a method to improve the end-to-end bandwidth utilization of RTP streams over an IP network using compression and multiplexing. This document describes the application of existing protocols for compression, multiplexing, and end to end tunneling.

Earlier Work

Multiplexing Scheme for RTP Flows between Access Routers
Gregor Bochmann, Gang Luo, Khalil El-khatib
Internet Draft, Internet Engineering Task Force, July 1999.
This draft proposes a light-weight data driven approach for multiplexing low bit rate RTP streams at the edge router of the Internet in order to reduce the RTP/UDP/IP header overhead associated with each RTP stream. Audio packets from different sources in a local access network destined to different users in the same remote access network are multiplexed into one packet, with the original RTP/UDP/IP header of each packet replaced with a mini-header (2 bytes), resulting in a reduction of the overhead. The de-multiplexing capability of the peer routers is determined in a very simple way.

Multiplexing Compressed RTP/UDP Packets in a PPP Frame
Rajesh Pazhyannur, Irfan Ali
Internet Draft, Internet Engineering Task Force, July 1999.
This draft proposes a scheme to increase the capacity of low-speed transmission links, T-1/E-1 or smaller, to transport low-bit rate voice applications over PPP. Given the relatively small payload size of the voice packets (average of 8 bytes for certain cellular networks) protocol overheads due to PPP, IP, UDP, and RTP are significant in determining the link capacity. The scheme proposed here is to multiplex compressed RTP/UDP packets into a single PPP frame. As a result, the PPP overhead (5-7 bytes) is distributed over multiple compressed RTP/UDP packets.

Realtime Traffic over Cellular Access Networks
Lars Westberg, Morgan Lindquist
Internet Draft, Internet Engineering Task Force, June 1999.
The draft discusses problems with transport of realtime traffic over cellular access channels and their implications for protocol enhancements.

Simple RTP Multiplexing Transfer Methods for VoIP
Tohru Hoshi, Koji Tsukada, Keiko Tanigawa
Internet Draft, Internet Engineering Task Force, November 1998.
This document proposes a simple voice stream multiplexing method which is designed to reduce the IP-UDP header overhead of RTP (real-time transport protocol) streams and to decrease the number of packets in the end-to-end transport functions. The proposed multiplexing method is to concatenate RTP packets destined for the same Internet Telephony Gateway (IP-GW) into a single UDP packet. The benefits of this method are that no new additional headers are required and the current well-defined H.323 and RTP standards can be used. Furthermore, this method is a general RTP packet multiplexing method that is applicable not only to an IP-GW but also to other multiplexing applications, as well as trunking VoIP streams application with insertion and deletion of RTP streams on the way.

Tunneled multiplexed Compressed RTP ("TCRTP")
T. Koren, P. Ruddy, B. Tompson, A. Tweedly, D. Wing
Internet Draft, Internet Engineering Task Force, June 1999.
This document describes a mechanism which improves the end-to-end bandwidth utilization of RTP streams over an IP network by compressing the UDP and RTP headers and allowing the packets of multiple RTP streams to be multiplexed into one IP packet.

An RTP Payload Format for User Multiplexing
Jonathan Rosenberg and Henning Schulzrinne
Internet Draft, Internet Engineering Task Force, May 1998.
This memo describes an RTP payload format for multiplexing data from multiple users into a single RTP packet. Such multiplexing is especially useful for transporting voice data between Internet telephony gateways. It causes significant reductions in header overheads and improves scalability.

GeRM: Generic RTP Multiplexing
Mark Handley
Internet Draft, Internet Engineering Task Force, Nov. 1998
This document describes GeRM, an RTP payload format for generic multiplexing of multiple RTP streams.

User Multiplexing in RTP payload between IP Telephony Gateways
B. Subbiah and S. Sengodan
Internet Draft, Internet Engineering Task Force, August 1998.
This draft proposes a method to multiplex a number of low bit rate audio streams (upto 256) into a single RTP/UDP/IP connection between IP telephony gateways. Audio samples from different users are assem- bled into an RTP payload thus reducing the overhead of RTP/UDP/IP headers. To identify users sharing a single RTP/UDP/IP connection, a 2 byte MINI-Header is proposed. A channel negotiation procedure to assign and release channels on a single UDP connection between gateways is described.

Issues and options for an aggregation service within RTP
Jonathan Rosenberg and Henning Schulzrinne
(expired) Internet Draft, Internet Engineering Task Force, Nov. 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.

Last updated by Henning Schulzrinne