WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-08

The information below is for an old version of the document
Document Type Expired Internet-Draft (rtcweb WG)
Authors Jean-Marc Valin  , Cary Bran 
Last updated 2015-11-02 (latest revision 2015-04-30)
Replaces draft-cbran-rtcweb-codec
Stream Internent Engineering Task Force (IETF)
Formats
Expired & archived
pdf htmlized (tools) htmlized bibtex
Reviews
Additional Resources
- Mailing list discussion
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)
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This Internet-Draft is no longer active.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-ietf-rtcweb-audio-08.txt

Abstract

This document outlines the audio codec and processing requirements for WebRTC endpoints.

Authors

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.)