draft-ietf-rtcweb-audio-10.txt   draft-ietf-rtcweb-audio-11.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: August 12, 2016 Plantronics Expires: October 23, 2016 Plantronics
February 9, 2016 April 21, 2016
WebRTC Audio Codec and Processing Requirements WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-10 draft-ietf-rtcweb-audio-11
Abstract Abstract
This document outlines the audio codec and processing requirements This document outlines the audio codec and processing requirements
for WebRTC endpoints. for WebRTC endpoints.
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 August 12, 2016. This Internet-Draft will expire on October 23, 2016.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2016 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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to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
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 . . . . . . . . . . . . . . . . . . . . . . . . . 4
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 . . . . . . . . . . . . . . . . . . . . . . 6
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 6 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 6
10.1. Normative References . . . . . . . . . . . . . . . . . . 6 10.1. Normative References . . . . . . . . . . . . . . . . . . 6
10.2. Informative References . . . . . . . . . . . . . . . . . 6 10.2. Informative References . . . . . . . . . . . . . . . . . 7
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 7 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 7
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 endpoints. audio processing and codec requirements for WebRTC endpoints.
2. Terminology 2. Terminology
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To ensure a baseline level of interoperability between WebRTC To ensure a baseline level of interoperability between WebRTC
endpoints, a minimum set of required codecs are specified below. If endpoints, a minimum set of required codecs are specified below. If
other suitable audio codecs are available for the WebRTC endpoint to other suitable audio codecs are available for the WebRTC endpoint to
use, it is RECOMMENDED that they are also be included in the offer in use, it is RECOMMENDED that they are also be included in the offer in
order to maximize the possibility to establish the session without order to maximize the possibility to establish the session without
the need for audio transcoding. the need for audio transcoding.
WebRTC endpoints are REQUIRED to implement the following audio WebRTC endpoints are REQUIRED to implement the following audio
codecs: codecs:
o Opus [RFC6716] with the payload format specified in o Opus [RFC6716] with the payload format specified in [RFC7587].
[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). WebRTC endpoints MUST support o [RFC3389] comfort noise (CN). WebRTC endpoints MUST support
RFC3389 CN for streams encoded with G.711 or any other supported RFC3389 CN for streams encoded with G.711 or any other supported
codec that does not provide its own CN. Since Opus provides its codec that does not provide its own CN. Since Opus provides its
own CN mechanism, the use of RFC3389 CN with Opus is NOT own CN mechanism, the use of RFC3389 CN with Opus is NOT
RECOMMENDED. Use of DTX/CN by senders is OPTIONAL. RECOMMENDED. Use of DTX/CN by senders is OPTIONAL.
o The audio/telephone-event media format as specified in [RFC4733]. o The audio/telephone-event media format as specified in [RFC4733].
The endpoints MAY send DTMF events at any time and SHOULD suppress The endpoints MAY send DTMF events at any time and SHOULD suppress
in-band DTMF tones, if any. WebRTC endpoints are REQUIRED to be in-band DTMF tones, if any. DTMF events generated by a WebRTC
able to generate and consume the following events: endpoint MUST have a duration of no more than 8000 ms and no less
than 40 ms. The recommended default duration is 100 ms for each
tone. The gap between events MUST be no less than 30 ms; the
recommended default gap duration is 70 ms. WebRTC endpoints are
not required to do anything with RFC 4733 tones sent to them,
except gracefully drop them. There is currently no API to inform
JavaScript about the received DTMF or other RFC 4733 tones.
WebRTC endpoints are REQUIRED to be able to generate and consume
the following events:
+------------+--------------------------------+-----------+ +------------+--------------------------------+-----------+
|Event Code | Event Name | Reference | |Event Code | Event Name | Reference |
+------------+--------------------------------+-----------+ +------------+--------------------------------+-----------+
| 0 | DTMF digit "0" | RFC4733 | | 0 | DTMF digit "0" | RFC4733 |
| 1 | DTMF digit "1" | RFC4733 | | 1 | DTMF digit "1" | RFC4733 |
| 2 | DTMF digit "2" | RFC4733 | | 2 | DTMF digit "2" | RFC4733 |
| 3 | DTMF digit "3" | RFC4733 | | 3 | DTMF digit "3" | RFC4733 |
| 4 | DTMF digit "4" | RFC4733 | | 4 | DTMF digit "4" | RFC4733 |
| 5 | DTMF digit "5" | RFC4733 | | 5 | DTMF digit "5" | RFC4733 |
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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
For security considerations regarding the codecs themselves please For security considerations regarding the codecs themselves please
refer their specifications, including [RFC6716], refer their specifications, including [RFC6716], [RFC7587],
[I-D.ietf-payload-rtp-opus], [RFC3551], [RFC3389], and [RFC4733]. [RFC3551], [RFC3389], and [RFC4733]. Likewise, consult the RTP base
Likewise, consult the RTP base specification for security RTP-based specification for RTP-based security considerations. WebRTC security
security considerations. WebRTC security is further discussed in is further discussed in [I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch] and [I-D.ietf-rtcweb-security-arch] and [I-D.ietf-rtcweb-rtp-usage].
[I-D.ietf-rtcweb-rtp-usage].
Implementers should consider whether the use of VBR is appropriate Implementers should consider whether the use of variable bitrate is
for their application based on [RFC6562]. Encryption and appropriate 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 particular we would like to acknowledge, and say thanks for, work we
we incorporated from Harald Alvestrand and Cullen Jennings. incorporated from Harald Alvestrand and Cullen Jennings.
10. References 10. References
10.1. Normative 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, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>. <http://www.rfc-editor.org/info/rfc2119>.
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[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, DOI 10.17487/RFC6716, Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <http://www.rfc-editor.org/info/rfc6716>. September 2012, <http://www.rfc-editor.org/info/rfc6716>.
[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, Variable Bit Rate Audio with Secure RTP", RFC 6562,
DOI 10.17487/RFC6562, March 2012, DOI 10.17487/RFC6562, March 2012,
<http://www.rfc-editor.org/info/rfc6562>. <http://www.rfc-editor.org/info/rfc6562>.
[I-D.ietf-payload-rtp-opus] [RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
Spittka, J., Vos, K., and J. Valin, "RTP Payload Format for the Opus Speech and Audio Codec", RFC 7587,
for the Opus Speech and Audio Codec", draft-ietf-payload- DOI 10.17487/RFC7587, June 2015,
rtp-opus-11 (work in progress), April 2015. <http://www.rfc-editor.org/info/rfc7587>.
10.2. Informative References 10.2. Informative References
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-08 (work in progress), February 2015. ietf-rtcweb-security-08 (work in progress), February 2015.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-11 (work in progress), March 2015. rtcweb-security-arch-11 (work in progress), March 2015.
[I-D.ietf-rtcweb-rtp-usage] [I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP", Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-23 (work in progress), March draft-ietf-rtcweb-rtp-usage-25 (work in progress), June
2015. 2015.
[I-D.ietf-rtcweb-audio-codecs-for-interop] [I-D.ietf-rtcweb-audio-codecs-for-interop]
Proust, S., Berger, E., Feiten, B., Burman, B., Bogineni, Proust, S., "Additional WebRTC audio codecs for
K., Lei, M., and E. Marocco, "Additional WebRTC audio interoperability.", draft-ietf-rtcweb-audio-codecs-for-
codecs for interoperability.", draft-ietf-rtcweb-audio- interop-05 (work in progress), February 2016.
codecs-for-interop-01 (work in progress), January 2015.
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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