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Additional WebRTC audio codecs for interoperability with legacy networks.
draft-proust-rtcweb-audio-codecs-for-interop-01

Document Type Replaced Internet-Draft (individual)
Expired & archived
Authors Stephane Proust , Espen Berger , Bernhard Feiten , Bo Burman , Kalyani Bogineni , Miao Lei , Enrico Marocco
Last updated 2015-02-15 (Latest revision 2014-08-14)
Replaced by draft-ietf-rtcweb-audio-codecs-for-interop
RFC stream (None)
Intended RFC status (None)
Formats
Stream Stream state (No stream defined)
Consensus boilerplate Unknown
RFC Editor Note (None)
IESG IESG state Replaced by draft-ietf-rtcweb-audio-codecs-for-interop
Telechat date (None)
Responsible AD (None)
Send notices to (None)

This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:

Abstract

To ensure a baseline level of interoperability between WebRTC clients, [I-D.ietf-rtcweb-audio] requires a minimum set of codecs. However, to maximize the possibility to establish the session without the need for audio transcoding, it is also recommended to include in the offer other suitable audio codecs that are available to the browser. This document provides some guidelines on the suitable codecs to be considered for WebRTC clients to address the most relevant interoperability use cases.

Authors

Stephane Proust
Espen Berger
Bernhard Feiten
Bo Burman
Kalyani Bogineni
Miao Lei
Enrico Marocco

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