WebRTC Audio Codec and Processing Requirements

The information below is for an old version of the document
Document Type Expired Internet-Draft (rtcweb WG)
Last updated 2015-04-27 (latest revision 2014-10-24)
Replaces draft-cbran-rtcweb-codec
Stream IETF
Intended RFC status Proposed Standard
Expired & archived
pdf htmlized bibtex
Stream WG state WG Document
Document shepherd Cullen Jennings
Shepherd write-up Show (last changed 2015-04-08)
IESG IESG state Expired
Consensus Boilerplate Unknown
Telechat date
Responsible AD (None)
Send notices to (None)

This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found at


This document outlines the audio codec and processing requirements for WebRTC client application and endpoint devices.


Jean-Marc Valin (jmvalin@jmvalin.ca)
Cary Bran (cary.bran@plantronics.com)

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