draft-ietf-rtcweb-audio-00.txt   draft-ietf-rtcweb-audio-01.txt 
Network Working Group JM. Valin Network Working Group JM. Valin
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Standards Track C. Bran Intended status: Standards Track C. Bran
Expires: March 11, 2013 Plantronics Expires: May 27, 2013 Plantronics
September 7, 2012 November 23, 2012
WebRTC Audio Codec and Processing Requirements WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-00 draft-ietf-rtcweb-audio-01
Abstract Abstract
This document outlines the audio codec and processing requirements This document outlines the audio codec and processing requirements
for WebRTC client application and endpoint devices. for WebRTC client application and endpoint devices.
Status of this Memo Status of this Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on March 11, 2013. This Internet-Draft will expire on May 27, 2013.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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WebRTC client implementations, it leaves the question of supporting WebRTC client implementations, it leaves the question of supporting
additional codecs to the will of the implementer. additional codecs to the will of the implementer.
WebRTC clients are REQUIRED to implement the following audio codecs. WebRTC clients are REQUIRED to implement the following audio codecs.
o Opus [RFC6716], with any ptime value up to 120 ms o Opus [RFC6716], with any ptime value up to 120 ms
o G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and a 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] ptime of 20 - see section 4.5.14 of [RFC3551]
o Telephone Event - [RFC4734] o Telephone Event - [RFC4733]
For all cases where the client is able to process audio at a sampling 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 rate higher than 8 kHz, it is RECOMMENDED that Opus be offered before
PCMA/PCMU. For Opus, all modes MUST be supported on the decoder PCMA/PCMU. For Opus, all modes MUST be supported on the decoder
side. The choice of encoder-side modes is left to the implementer. side. The choice of encoder-side modes is left to the implementer.
Clients MAY use the offer/answer mechanism to signal a preference for Clients MAY use the offer/answer mechanism to signal a preference for
a particular mode or ptime. a particular mode or ptime.
4. Audio Level 4. Audio Level
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It is plausible that the dominant near to mid-term WebRTC usage model It is plausible that the dominant near to mid-term WebRTC usage model
will be people using the interactive audio and video capabilities to will be people using the interactive audio and video capabilities to
communicate with each other via web browsers running on a notebook communicate with each other via web browsers running on a notebook
computer that has built-in microphone and speakers. The notebook-as- computer that has built-in microphone and speakers. The notebook-as-
communication-device paradigm presents challenging echo cancellation communication-device paradigm presents challenging echo cancellation
problems, the specific remedy of which will not be mandated here. problems, the specific remedy of which will not be mandated here.
However, while no specific algorithm or standard will be required by However, while no specific algorithm or standard will be required by
WebRTC compatible clients, echo cancellation will improve the user WebRTC compatible clients, echo cancellation will improve the user
experience and should be implemented by the endpoint device. experience and should be implemented by the endpoint device.
SHOULD include an AEC and if not, SHOULD ensure that the speaker-to- WebRTC clients SHOULD include an AEC and if that is not possible, the
microphone gain is below unity at all frequencies to avoid clients SHOULD ensure that the speaker-to-microphone gain is below
instability when none of the client has echo cancellation. For unity at all frequencies to avoid instability when none of the client
clients that do not control the audio capture and playback devices has echo cancellation. For clients that do not control the audio
directly, it is RECOMMENDED to support echo cancellation between capture and playback devices directly, it is RECOMMENDED to support
devices running at slight different sampling rates, such as when a echo cancellation between devices running at slight different
webcam is used for microphone. sampling rates, such as when a webcam is used for microphone.
The client SHOULD allow either the entire AEC or the non-linear The client SHOULD allow either the entire AEC or the non-linear
processing (NLP) to be turned off for applications, such as music, processing (NLP) to be turned off for applications, such as music,
that do not behave well with the spectral attenuation methods that do not behave well with the spectral attenuation methods
typically used in NLPs. It SHOULD have the ability to detect the typically used in NLPs. It SHOULD have the ability to detect the
presence of a headset and disable echo cancellation. presence of a headset and disable echo cancellation.
For some applications where the remote client may not have an echo For some applications where the remote client may not have an echo
canceller, the local client MAY include a far-end echo canceller, but canceller, the local client MAY include a far-end echo canceller, but
if that it the case, it SHOULD be disabled by default. if that is the case, it SHOULD be disabled by default.
6. Legacy VoIP Interoperability 6. Legacy VoIP Interoperability
The codec requirements above will ensure, at a minimum, voice The codec requirements above will ensure, at a minimum, voice
interoperability capabilities between WebRTC client applications and interoperability capabilities between WebRTC client applications and
legacy phone systems. legacy phone systems.
7. IANA Considerations 7. IANA Considerations
This document makes no request of IANA. This document makes no request of IANA.
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Rescorla, E., "Security Considerations for RTC-Web", Rescorla, E., "Security Considerations for RTC-Web",
May 2011. May 2011.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003. July 2003.
[RFC4734] Schulzrinne, H. and T. Taylor, "Definition of Events for [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Modem, Fax, and Text Telephony Signals", RFC 4734, Digits, Telephony Tones, and Telephony Signals", RFC 4733,
December 2006. December 2006.
Authors' Addresses Authors' Addresses
Jean-Marc Valin Jean-Marc Valin
Mozilla Mozilla
650 Castro Street 650 Castro Street
Mountain View, CA 94041 Mountain View, CA 94041
USA USA
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