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

Document type: Expired Internet-Draft (individual)
Document stream: No stream defined
Last updated: 2012-11-17 (latest revision 2012-05-16)
Intended RFC status: Unknown
Other versions: (expired, archived): plain text, pdf, html

Stream State:No stream defined
Document shepherd: No shepherd assigned

IESG State: Expired
Responsible AD: (None)
Send notices to: No addresses provided

This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found here:
http://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>
BoB <bo.burman@ericsson.com>

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