Codec Control for WebRTC
draft-westerlund-rtcweb-codec-control-00

 
Document Type Expired Internet-Draft (individual)
Last updated 2012-11-17 (latest revision 2012-05-16)
Stream (None)
Intended RFC status (None)
Formats
Expired & archived
plain text pdf html
Stream Stream state (No stream defined)
Document shepherd No shepherd assigned
IESG IESG state Expired
Telechat date
Responsible AD (None)
Send notices to (None)

Email authors IPR References Referenced by Nits Search lists

This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found at
https://www.ietf.org/archive/id/draft-westerlund-rtcweb-codec-control-00.txt

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

Magnus Westerlund (magnus.westerlund@ericsson.com)
Bo Burman (bo.burman@ericsson.com)

(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)