Simple RTP Multiplexing Transfer Methods for VoIP
draft-tanigawa-rtp-multiplex-01
Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
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Authors | Tohru Hoshi , Koji Tsukada , Keiko Tanigawa | ||
Last updated | 1998-11-16 | ||
RFC stream | (None) | ||
Intended RFC status | (None) | ||
Formats | |||
Stream | Stream state | (No stream defined) | |
Consensus boilerplate | Unknown | ||
RFC Editor Note | (None) | ||
IESG | IESG state | Expired | |
Telechat date | (None) | ||
Responsible AD | (None) | ||
Send notices to | (None) |
This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:
Abstract
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.
Authors
Tohru Hoshi
Koji Tsukada
Keiko Tanigawa
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)