WebRTC Audio Codec and Processing Requirements
RFC 7874

Document Type RFC - Proposed Standard (May 2016; No errata)
Last updated 2016-05-25
Replaces draft-cbran-rtcweb-codec
Stream IETF
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Stream WG state Submitted to IESG for Publication (wg milestone: Apr 2014 - Audio Processing and... )
Document shepherd Cullen Jennings
Shepherd write-up Show (last changed 2016-01-27)
IESG IESG state RFC 7874 (Proposed Standard)
Consensus Boilerplate Yes
Telechat date
Responsible AD Alissa Cooper
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IANA IANA review state Version Changed - Review Needed
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Internet Engineering Task Force (IETF)                         JM. Valin
Request for Comments: 7874                                       Mozilla
Category: Standards Track                                        C. Bran
ISSN: 2070-1721                                              Plantronics
                                                                May 2016

             WebRTC Audio Codec and Processing Requirements

Abstract

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

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc7874.

Copyright Notice

   Copyright (c) 2016 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
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Valin & Bran                 Standards Track                    [Page 1]
RFC 7874                      WebRTC Audio                      May 2016

Table of Contents

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

1.  Introduction

   An integral part of the success and adoption of 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 endpoints.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "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
   endpoints, a minimum set of required codecs are specified below.  If
   other suitable audio codecs are available for the WebRTC endpoint to
   use, it is RECOMMENDED that they also be included in the offer in
   order to maximize the possibility of establishing the session without
   the need for audio transcoding.

   WebRTC endpoints are REQUIRED to implement the following audio
   codecs:

   o  Opus [RFC6716] with the payload format specified in [RFC7587].

   o  PCMA and PCMU (as specified in ITU-T Recommendation G.711 [G.711])
      with the payload format specified in Section 4.5.14 of [RFC3551].

Valin & Bran                 Standards Track                    [Page 2]
RFC 7874                      WebRTC Audio                      May 2016

   o  [RFC3389] comfort noise (CN).  WebRTC endpoints 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 mechanism, the use of [RFC3389] CN with Opus is NOT
      RECOMMENDED.  Use of Discontinuous Transmission (DTX) / CN by
      senders is OPTIONAL.

   o  the 'audio/telephone-event' media type as specified in [RFC4733].
      The endpoints MAY send DTMF events at any time and SHOULD suppress
      in-band dual-tone multi-frequency (DTMF) tones, if any.  DTMF
      events generated by a WebRTC endpoint MUST have a duration of no
      more than 8000 ms and no less than 40 ms.  The recommended default
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