draft-ietf-rtcweb-audio-06.txt   draft-ietf-rtcweb-audio-07.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 9, 2015 Plantronics Expires: April 27, 2015 Plantronics
September 5, 2014 October 24, 2014
WebRTC Audio Codec and Processing Requirements WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-06 draft-ietf-rtcweb-audio-07
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 9, 2015. This Internet-Draft will expire on April 27, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 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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1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . 2 3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . 2
4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . 3 4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . 3
5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . 4 5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . 4
6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 5 6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 5
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5
8. Security Considerations . . . . . . . . . . . . . . . . . . . 5 8. Security Considerations . . . . . . . . . . . . . . . . . . . 5
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5
10. Normative References . . . . . . . . . . . . . . . . . . . . 5 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 5
10.1. Normative References . . . . . . . . . . . . . . . . . . 5
10.2. Informative References . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 6 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 6
1. Introduction 1. Introduction
An integral part of the success and adoption of the Web Real Time An integral part of the success and adoption of the Web Real Time
Communications (WebRTC) will be the voice and video interoperability Communications (WebRTC) will be the voice and video interoperability
between WebRTC applications. This specification will outline the between WebRTC applications. This specification will outline the
audio processing and codec requirements for WebRTC client audio processing and codec requirements for WebRTC client
implementations. implementations.
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To ensure a baseline level of interoperability between WebRTC To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs are specified below. If clients, a minimum set of required codecs are specified below. If
other suitable audio codecs are available for the browser to use, it 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 is RECOMMENDED that they are also be included in the offer in order
to maximize the possibility to establish the session without the need to maximize the possibility to establish the session without the need
for audio transcoding. for audio transcoding.
WebRTC clients are REQUIRED to implement the following audio codecs: WebRTC clients are REQUIRED to implement the following audio codecs:
o Opus [RFC6716] with the payload format specified in [Opus-RTP]. o Opus [RFC6716] with the payload format specified in
[I-D.ietf-payload-rtp-opus].
o G.711 PCMA and PCMU with the payload format specified in section o G.711 PCMA and PCMU with the payload format specified in section
4.5.14 of [RFC3551]. 4.5.14 of [RFC3551].
o [RFC3389] comfort noise (CN). Receivers MUST support RFC3389 CN o [RFC3389] comfort noise (CN). Receivers MUST support RFC3389 CN
for streams encoded with G.711 or any other supported codec that for streams encoded with G.711 or any other supported codec that
does not provide its own CN. Since Opus provides its own CN does not provide its own CN. Since Opus provides its own CN
mechanism, the use of RFC3389 CN with Opus is NOT RECOMMENDED. mechanism, the use of RFC3389 CN with Opus is NOT RECOMMENDED.
Use of DTX/CN by senders is OPTIONAL. Use of DTX/CN by senders is OPTIONAL.
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| 11 | DTMF digit "#" | RFC4733 | | 11 | DTMF digit "#" | 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.
For additional information on implementing codecs other than the
mandatory-to-implement codecs listed above, refer to
[I-D.ietf-rtcweb-audio-codecs-for-interop].
4. Audio Level 4. Audio Level
It is desirable to standardize the "on the wire" audio level for It is desirable to standardize the "on the wire" audio level for
speech transmission to avoid users having to manually adjust the speech transmission to avoid users having to manually adjust the
playback and to facilitate mixing in conferencing applications. It playback and to facilitate mixing in conferencing applications. It
is also desirable to be consistent with ITU-T recommendations G.169 is also desirable to be consistent with ITU-T recommendations G.169
and G.115, which recommend an active audio level of -19 dBm0. and G.115, which recommend an active audio level of -19 dBm0.
However, unlike G.169 and G.115, the audio for WebRTC is not However, unlike G.169 and G.115, the audio for WebRTC is not
constrained to have a passband specified by G.712 and can in fact be constrained to have a passband specified by G.712 and can in fact be
sampled at any sampling rate from 8 kHz to 48 kHz and up. For this sampled at any sampling rate from 8 kHz to 48 kHz and up. For this
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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 is 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 that support G.711.
7. IANA Considerations 7. IANA Considerations
This document makes no request of IANA. This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. RFC.
8. Security Considerations 8. Security Considerations
Implementers should consider whether the use of VBR is appropriate Implementers should consider whether the use of VBR is appropriate
for their application based on [RFC6562]. Encryption and for their application based on [RFC6562]. Encryption and
authentication issues are beyond the scope of this document. authentication issues are beyond the scope of this document.
9. Acknowledgements 9. Acknowledgements
This draft incorporates ideas and text from various other drafts. In This draft incorporates ideas and text from various other drafts. In
particularly we would like to acknowledge, and say thanks for, work particularly we would like to acknowledge, and say thanks for, work
we incorporated from Harald Alvestrand and Cullen Jennings. we incorporated from Harald Alvestrand and Cullen Jennings.
10. Normative References 10. References
10.1. Normative References
[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.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002. Comfort Noise (CN)", RFC 3389, September 2002.
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Digits, Telephony Tones, and Telephony Signals", RFC 4733, Digits, Telephony Tones, and Telephony Signals", RFC 4733,
December 2006. December 2006.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, September 2012. Opus Audio Codec", RFC 6716, September 2012.
[RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP", RFC 6562, March Variable Bit Rate Audio with Secure RTP", RFC 6562, March
2012. 2012.
[Opus-RTP] [I-D.ietf-payload-rtp-opus]
Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format Spittka, J., Vos, K., and J. Valin, "RTP Payload Format
for Opus Codec", August 2013. for Opus Speech and Audio Codec", draft-ietf-payload-rtp-
opus-03 (work in progress), July 2014.
10.2. Informative References
[I-D.ietf-rtcweb-audio-codecs-for-interop]
Proust, S., Berger, E., Feiten, B., Bogineni, K., Lei, M.,
and E. Marocco, "Additional WebRTC audio codecs for
interoperability with legacy networks.", draft-ietf-
rtcweb-audio-codecs-for-interop-00 (work in progress),
September 2014.
Authors' Addresses Authors' Addresses
Jean-Marc Valin Jean-Marc Valin
Mozilla Mozilla
331 E. Evelyn Avenue 331 E. Evelyn Avenue
Mountain View, CA 94041 Mountain View, CA 94041
USA USA
Email: jmvalin@jmvalin.ca Email: jmvalin@jmvalin.ca
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