Skip to main content

WebRTC Codec and Media Processing Requirements
draft-cbran-rtcweb-codec-01

The information below is for an old version of the document.
Document Type
This is an older version of an Internet-Draft whose latest revision state is "Replaced".
Authors Cary Bran , Cullen Fluffy Jennings
Last updated 2011-10-29 (Latest revision 2011-07-01)
Replaced by draft-ietf-rtcweb-audio, draft-ietf-rtcweb-video, draft-ietf-rtcweb-video, RFC 7874, RFC 7742
RFC stream (None)
Formats
Additional resources
Stream Stream state (No stream defined)
Consensus boilerplate Unknown
RFC Editor Note (None)
IESG IESG state I-D Exists
Telechat date (None)
Responsible AD (None)
Send notices to (None)
draft-cbran-rtcweb-codec-01
Network Working Group                                            C. Bran
Internet-Draft                                               Plantronics
Intended status: Standards Track                             C. Jennings
Expires: May 1, 2012                                               Cisco
                                                        October 29, 2011

             WebRTC Codec and Media Processing Requirements
                      draft-cbran-rtcweb-codec-01

Abstract

   This document outlines the codec and media 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.  This document may not be modified,
   and derivative works of it may not be created, and it may not be
   published except as an Internet-Draft.

   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 May 1, 2012.

Copyright Notice

   Copyright (c) 2011 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.

Bran & Jennings            Expires May 1, 2012                  [Page 1]
Internet-Draft              Abbreviated-Title               October 2011

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . . . 3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . 3
   3.  Codec Requirements  . . . . . . . . . . . . . . . . . . . . . . 3
     3.1.  Audio Codec Requirements  . . . . . . . . . . . . . . . . . 3
     3.2.  Video Codec Requirements  . . . . . . . . . . . . . . . . . 3
   4.  WebRTC Client Requirements  . . . . . . . . . . . . . . . . . . 4
   5.  Legacy VoIP Interoperability  . . . . . . . . . . . . . . . . . 5
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 5
   7.  Security Considerations . . . . . . . . . . . . . . . . . . . . 5
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . 5
   9.  Normative References  . . . . . . . . . . . . . . . . . . . . . 5
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . . . 6

Bran & Jennings            Expires May 1, 2012                  [Page 2]
Internet-Draft              Abbreviated-Title               October 2011

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
   media 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

   This section covers the audio and video codec requirements for WebRTC
   client applications.  To ensure a baseline level of interoperability
   between WebRTC clients, a minimum set of required codecs are
   specified below.  While this section specifies the codecs that will
   be mandated for all WebRTC client implementations, it leaves the
   question of supporting additional codecs to the will of the
   implementer.

3.1.  Audio Codec Requirements

   WebRTC clients are REQUIRED to implement the following audio codecs.

   o  PCMA/PCMU - 1 channel with a rate of 8000 Hz and a ptime of 20 -
      see section 4.5.14 of [RFC3551]

   o  Telephone Event - [RFC4734]

   o  Opus [draft-ietf-codec-opus]

3.2.  Video Codec Requirements

   If the MPEG-LA issues an intent to offer H.264 baseline profile on a
   royalty free basis for use in browsers before March 15, 2012, then
   the REQUIRED video codecs will be H.264 baseline.  If this does not
   happen by that the date, then the REQUIRED video codec will be VP8
   [I-D.webm].

   The following feature list applies to all required video codecs.

   Required video codecs:

Bran & Jennings            Expires May 1, 2012                  [Page 3]
Internet-Draft              Abbreviated-Title               October 2011

   o  MUST support at least 10 frames per second (fps) and SHOULD
      support 30 fps

   o  If VP8, then MUST support a the bilinear and none reconstruction
      filters

   o  OPTIONALLY offer support for additional color spaces

   o  MUST support a minimum resolution of 320X240

   o  SHOULD support resolutions of 1280x720, 720x480, 1024x768,
      800x600, 640x480, 640 x 360 , 320x240

4.  WebRTC Client Requirements

   It is plausible that the dominant near to mid-term WebRTC usage model
   will be people using the interactive audio and video capabilities to
   communicate with each other via web browsers running on a notebook
   computer that has built-in microphone and speakers.  The notebook-as-
   communication-device paradigm presents challenging echo cancellation
   and audio gain problems, the specific remedy of which will not be
   mandated here.  However, while no specific algorithm or standard will
   be required by WebRTC compatible clients, functionality such as
   automatic gain control, echo cancellation, headset detection and
   passing call control events to connected devices will improve the
   user experience and should be implemented by the endpoint device.

   To address the problems outlined above, suitable implementations of
   the functionality listed below SHOULD be available within an RTC-Web
   endpoint device.

   o  Automatic gain control

   o  Ability to override automatic gain control to manually set gain

   o  Auto-adjustments to gain control and echo cancellation algorithms
      based on if a headset or internal speakers/microphone is being
      used

   o  Echo cancellation, including acoustic echo cancellation

   o  Headset detection

   o  Call control event notification to connected devices such as
      headsets

Bran & Jennings            Expires May 1, 2012                  [Page 4]
Internet-Draft              Abbreviated-Title               October 2011

5.  Legacy VoIP Interoperability

   The codec requirements above will ensure, at a minimum, voice
   interoperability capabilities between WebRTC client applications and
   legacy phone systems.

   Video interoperability will be dependent upon the MPEG-LA decision
   regarding H.264 baseline.

6.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.

7.  Security Considerations

   The codec requirements have no additional security considerations
   other than those captured in
   [I-D.ekr-security-considerations-for-rtc-web].

8.  Acknowledgements

   This draft incorporates ideas and text from various other drafts.  In
   particularly we would like to acknowledge, and say thanks for, work
   we incorporated from Harald Alvestrand.

9.  Normative References

   [I-D.ekr-security-considerations-for-rtc-web]
              Rescorla, E., "Security Considerations for RTC-Web",
              May 2011.

   [I-D.webm]
              Google, Inc., "VP8 Data Format and Decoding Guide",
              July 2010.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

Bran & Jennings            Expires May 1, 2012                  [Page 5]
Internet-Draft              Abbreviated-Title               October 2011

   [RFC4734]  Schulzrinne, H. and T. Taylor, "Definition of Events for
              Modem, Fax, and Text Telephony Signals", RFC 4734,
              December 2006.

Authors' Addresses

   Cary Bran
   Plantronics
   345 Encinial Street
   Santa Cruz, CA  95060
   USA

   Phone: +1 206 661-2398
   Email: cary.bran@plantronics.com

   Cullen Jennings
   Cisco
   170 West Tasman Drive
   San Jose, CA  95134
   USA

   Phone: +1 408 421-9990
   Email: fluffy@cisco.com

Bran & Jennings            Expires May 1, 2012                  [Page 6]