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

Expired Internet-Draft (None)
Document Stream: No stream defined
Last updated: 2012-05-16
Intended RFC status: (None)
Other versions: (expired, archived): plain text, pdf, html

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Shepherd writeup

IESG State: Expired
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This Internet-Draft is no longer active. Unofficial copies of old Internet-Drafts can be found here:
http://tools.ietf.org/id/draft-westerlund-rtcweb-codec-control.

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)