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

Document Type Expired Internet-Draft (individual)
Last updated 2013-08-29 (latest revision 2013-02-25)
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This Internet-Draft is no longer active.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-marjou-rtcweb-audio-codecs-for-interop-01.txt

Abstract

This document presents use-cases underlining why WebRTC needs AMR-WB, AMR and G.722 as additional relevant voice codecs to satisfactorily ensure interoperability with existing systems. It also presents a way forward that takes into consideration the concerns expressed against the addition of codecs besides Opus and G.711. It is especially recognized that unjustified additional costs on browsers must be avoided. Therefore, the proposed solution intends to fully rely on the codecs already supported on the devices implementing the browsers and for which license and implementation costs have been already paid. It is expected that this way forward will significantly limit the costs and technical impacts on browsers while greatly improving interoperability with legacy systems and overall quality. It intents to be considered as a good compromise beneficial to all parties and to the whole industry: the user quality experience will be optimized as a whole at limited additional costs without incurring high costs for both networks to support transcoding and browsers to support additional codecs.

Authors

Xavier Marjou (xavier.marjou@orange.com)
Stephane Proust (stephane.proust@orange.com)
Kalyani Bogineni (kalyani.bogineni@verizonwireless.com)
Roland Jesske (r.jesske@telekom.de)
Miao Lei (lei.miao@huawei.com)
Enrico Marocco (enrico.marocco@telecomitalia.it)
Espen Berger (espeberg@cisco.com)

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