WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-06

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Document Type Active Internet-Draft (rtcweb WG)
Last updated 2014-09-05
Replaces draft-cbran-rtcweb-codec
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Network Working Group                                          JM. Valin
Internet-Draft                                                   Mozilla
Intended status: Standards Track                                 C. Bran
Expires: March 9, 2015                                       Plantronics
                                                       September 5, 2014

             WebRTC Audio Codec and Processing Requirements
                       draft-ietf-rtcweb-audio-06

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-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on March 9, 2015.

Copyright Notice

   Copyright (c) 2014 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
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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Table of Contents

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

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].

   o  G.711 PCMA and PCMU with the payload format specified in section
      4.5.14 of [RFC3551].

   o  [RFC3389] comfort noise (CN).  Receivers MUST support RFC3389 CN
      for streams encoded with G.711 or any other supported codec that
      does not provide its own CN.  Since Opus provides its own CN

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      mechanism, the use of RFC3389 CN with Opus is NOT RECOMMENDED.
      Use of DTX/CN by senders is OPTIONAL.

   o  The audio/telephone-event media format as specified in [RFC4733].
      WebRTC clients are REQUIRED to be able to generate and consume the
      following events:

         +------------+--------------------------------+-----------+
         |Event Code  | Event Name                     | Reference |
         +------------+--------------------------------+-----------+
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