Multiple Synchronization sources (SSRC) in RTP Session Signaling
draft-westerlund-avtcore-max-ssrc-00
| Document | Type |
This is an older version of an Internet-Draft whose latest revision state is "Expired".
Expired & archived
|
|
|---|---|---|---|
| Authors | Magnus Westerlund , Bo Burman , Fredrik Jansson | ||
| Last updated | 2012-04-26 (Latest revision 2011-10-24) | ||
| RFC stream | (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This Internet-Draft is no longer active.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
RTP has always been a protocol that supports multiple participants each sending their own media streams in an RTP session. Unfortunately many implementations are designed only for point to point voice over IP with a single source in each end-point. Even client implementations aimed at video conferences have often been built with the assumption around central mixers that only deliver a single media stream per media type. Thus any application that wants to allow for more advance usage where multiple media streams are sent and received by an end-point has an issue with legacy implementations. This document describes the problem and proposes a solution for how to use multiple SSRCs within one RTP session and at the same time handle the legacy issues.
Authors
Magnus Westerlund
Bo Burman
Fredrik Jansson
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)