Codec Control for WebRTC
draft-westerlund-rtcweb-codec-control-00
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Authors | Magnus Westerlund , Bo Burman | ||
| Last updated | 2012-11-17 (Latest revision 2012-05-16) | ||
| 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This Internet-Draft is no longer active.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 how WebRTC should handle media codec control between peers. With media codec control we mean such parameters as video resolution and frame-rate. This includes both initial establishment of capabilities using the SDP based JSEP signalling and during ongoing real-time interactive sessions in response to user and application events. The solution uses SDP for initial boundary establishment that are rarely, if ever changed. During the session the RTCP based Codec Operations Point (COP) signaling solution is used for dynamic control of parameters enabling timely and responsive controls.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)