WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-03

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Document Type Active Internet-Draft (rtcweb WG)
Last updated 2013-10-15
Replaces draft-cbran-rtcweb-codec
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Network Working Group                                          JM. Valin
Internet-Draft                                                   Mozilla
Intended status: Standards Track                                 C. Bran
Expires: April 18, 2014                                      Plantronics
                                                        October 15, 2013

             WebRTC Audio Codec and Processing Requirements
                       draft-ietf-rtcweb-audio-03

Abstract

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

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
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   Internet-Drafts are draft documents valid for a maximum of six months
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   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 18, 2014.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   described in the Simplified BSD License.

Table of Contents

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   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   2
   3.  Codec Requirements  . . . . . . . . . . . . . . . . . . . . .   2
   4.  Audio Level . . . . . . . . . . . . . . . . . . . . . . . . .   3
   5.  Acoustic Echo Cancellation (AEC)  . . . . . . . . . . . . . .   4
   6.  Legacy VoIP Interoperability  . . . . . . . . . . . . . . . .   4
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   4
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .   5
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   5
   10. Normative References  . . . . . . . . . . . . . . . . . . . .   5
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   5

1.  Introduction

   An integral part of the success and adoption of the Web Real Time
   Communications (WebRTC) will be the voice and video interoperability
   between WebRTC applications.  This specification will outline the
   audio processing and codec requirements for WebRTC client
   implementations.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

3.  Codec Requirements

   To ensure a baseline level of interoperability between WebRTC
   clients, a minimum set of required codecs are specified below.  If
   other suitable audio codecs are available for the browser to use, it
   is RECOMMENDED that they are also be included in the offer in order
   to maximize the possibility to establish the session without the need
   for audio transcoding.

   WebRTC clients are REQUIRED to implement the following audio codecs.

   o  Opus [RFC6716], with the payload format specified in [Opus-RTP]
      and any ptime value up to 120 ms

   o  G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and a
      ptime of 20 - see section 4.5.14 of [RFC3551]

   o  Telephone Event - [RFC4733]

   For all cases where the client is able to process audio at a sampling
   rate higher than 8 kHz, it is RECOMMENDED that Opus be offered before

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   PCMA/PCMU.  For Opus, all modes MUST be supported on the decoder
   side.  The choice of encoder-side modes is left to the implementer.
   Clients MAY use the offer/answer mechanism to signal a preference for
   a particular mode or ptime.

4.  Audio Level

   It is desirable to standardize the "on the wire" audio level for
   speech transmission to avoid users having to manually adjust the
   playback and to facilitate mixing in conferencing applications.  It
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