draft-ietf-rtcweb-audio-11.txt   rfc7874.txt 
Network Working Group JM. Valin Internet Engineering Task Force (IETF) JM. Valin
Internet-Draft Mozilla Request for Comments: 7874 Mozilla
Intended status: Standards Track C. Bran Category: Standards Track C. Bran
Expires: October 23, 2016 Plantronics ISSN: 2070-1721 Plantronics
April 21, 2016 May 2016
WebRTC Audio Codec and Processing Requirements WebRTC Audio Codec and Processing Requirements
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 is an Internet Standards Track document.
provisions of BCP 78 and BCP 79.
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 This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 5741.
This Internet-Draft will expire on October 23, 2016. 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 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
skipping to change at page 2, line 13 skipping to change at page 2, line 13
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 . . . . . . . . . . . . . . . . . . . . . . . . . 4 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. Security Considerations . . . . . . . . . . . . . . . . . . . 5
8. Security Considerations . . . . . . . . . . . . . . . . . . . 5 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 6
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 6 8.1. Normative References . . . . . . . . . . . . . . . . . . 6
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 6 8.2. Informative References . . . . . . . . . . . . . . . . . 6
10.1. Normative References . . . . . . . . . . . . . . . . . . 6 Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 7
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 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
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in RFC "OPTIONAL" in this document are to be interpreted as described in RFC
2119 [RFC2119]. 2119 [RFC2119].
3. Codec Requirements 3. Codec Requirements
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 also be included in the offer in
order to maximize the possibility to establish the session without order to maximize the possibility of establishing 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 [RFC7587]. o Opus [RFC6716] with the payload format specified in [RFC7587].
o G.711 PCMA and PCMU with the payload format specified in section o PCMA and PCMU (as specified in ITU-T Recommendation G.711 [G.711])
4.5.14 of [RFC3551]. with the payload format specified in Section 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 Discontinuous Transmission (DTX) / CN by
senders is OPTIONAL.
o The audio/telephone-event media format as specified in [RFC4733]. o the 'audio/telephone-event' media type 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. DTMF events generated by a WebRTC in-band dual-tone multi-frequency (DTMF) tones, if any. DTMF
endpoint MUST have a duration of no more than 8000 ms and no less events generated by a WebRTC endpoint MUST have a duration of no
than 40 ms. The recommended default duration is 100 ms for each more than 8000 ms and no less than 40 ms. The recommended default
tone. The gap between events MUST be no less than 30 ms; the duration is 100 ms for each tone. The gap between events MUST be
recommended default gap duration is 70 ms. WebRTC endpoints are no less than 30 ms; the recommended default gap duration is 70 ms.
not required to do anything with RFC 4733 tones sent to them, WebRTC endpoints are not required to do anything with tones (as
except gracefully drop them. There is currently no API to inform specified in RFC 4733) sent to them, except gracefully drop them.
JavaScript about the received DTMF or other RFC 4733 tones. There is currently no API to inform JavaScript about the received
WebRTC endpoints are REQUIRED to be able to generate and consume DTMF or other tones (as specified in RFC 4733). WebRTC endpoints
the following events: 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] |
| 6 | DTMF digit "6" | RFC4733 | | 6 | DTMF digit "6" | [RFC4733] |
| 7 | DTMF digit "7" | RFC4733 | | 7 | DTMF digit "7" | [RFC4733] |
| 8 | DTMF digit "8" | RFC4733 | | 8 | DTMF digit "8" | [RFC4733] |
| 9 | DTMF digit "9" | RFC4733 | | 9 | DTMF digit "9" | [RFC4733] |
| 10 | DTMF digit "*" | RFC4733 | | 10 | DTMF digit "*" | [RFC4733] |
| 11 | DTMF digit "#" | RFC4733 | | 11 | DTMF digit "#" | [RFC4733] |
| 12 | DTMF digit "A" | RFC4733 | | 12 | DTMF digit "A" | [RFC4733] |
| 13 | DTMF digit "B" | RFC4733 | | 13 | DTMF digit "B" | [RFC4733] |
| 14 | DTMF digit "C" | RFC4733 | | 14 | DTMF digit "C" | [RFC4733] |
| 15 | DTMF digit "D" | RFC4733 | | 15 | DTMF digit "D" | [RFC4733] |
+------------+--------------------------------+-----------+ +------------+--------------------------------+-----------+
For all cases where the endpoint is able to process audio at a For all cases where the endpoint is able to process audio at a
sampling rate higher than 8 kHz, it is RECOMMENDED that Opus be sampling rate higher than 8 kHz, it is RECOMMENDED that Opus be
offered before PCMA/PCMU. For Opus, all modes MUST be supported on offered before PCMA/PCMU. For Opus, all modes MUST be supported on
the decoder side. The choice of encoder-side modes is left to the the decoder side. The choice of encoder-side modes is left to the
implementer. Endpoints MAY use the offer/answer mechanism to signal implementer. Endpoints MAY use the offer/answer mechanism to signal
a preference for a particular mode or ptime. a preference for a particular mode or ptime.
For additional information on implementing codecs other than the For additional information on implementing codecs other than the
mandatory-to-implement codecs listed above, refer to mandatory-to-implement codecs listed above, refer to [RFC7875].
[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 to 48 kHz and higher. For this
reason, the level SHOULD be normalized by only considering reason, the level SHOULD be normalized by only considering
frequencies above 300 Hz, regardless of the sampling rate used. The frequencies above 300 Hz, regardless of the sampling rate used. The
level SHOULD also be adapted to avoid clipping, either by lowering level SHOULD also be adapted to avoid clipping, either by lowering
the gain to a level below -19 dBm0, or through the use of a the gain to a level below -19 dBm0 or through the use of a
compressor. compressor.
Assuming 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to Assuming linear 16-bit PCM with a value of +/-32767, -19 dBm0
a root mean square (RMS) level of 2600. Only active speech should be corresponds to a root mean square (RMS) level of 2600. Only active
considered in the RMS calculation. If the endpoint has control over speech should be considered in the RMS calculation. If the endpoint
the entire audio capture path, as is typically the case for a regular has control over the entire audio-capture path, as is typically the
phone, then it is RECOMMENDED that the gain be adjusted in such a way case for a regular phone, then it is RECOMMENDED that the gain be
that active speech have a level of 2600 (-19 dBm0) for an average adjusted in such a way that an average speaker would have a level of
speaker. If the endpoint does not have control over the entire audio 2600 (-19 dBm0) for active speech. If the endpoint does not have
capture, as is typically the case for a software endpoint, then the control over the entire audio capture, as is typically the case for a
endpoint SHOULD use automatic gain control (AGC) to dynamically software endpoint, then the endpoint SHOULD use automatic gain
adjust the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop control (AGC) to dynamically adjust the level to 2600 (-19 dBm0) +/-
sharing applications, the level SHOULD NOT be automatically adjusted 6 dB. For music- or desktop-sharing applications, the level SHOULD
and the endpoint SHOULD allow the user to set the gain manually. NOT be automatically adjusted, and the endpoint SHOULD allow the user
to set the gain manually.
The RECOMMENDED filter for normalizing the signal energy is a second- The RECOMMENDED filter for normalizing the signal energy is a second-
order Butterworth filter with a 300 Hz cutoff frequency. order Butterworth filter with a 300 Hz cutoff frequency.
It is common for the audio output on some devices to be "calibrated" It is common for the audio output on some devices to be "calibrated"
for playing back pre-recorded "commercial" music, which is typically for playing back pre-recorded "commercial" music, which is typically
around 12 dB louder than the level recommended in this section. around 12 dB louder than the level recommended in this section.
Because of this, endpoints MAY increase the gain before playback. Because of this, endpoints MAY increase the gain before playback.
5. Acoustic Echo Cancellation (AEC) 5. Acoustic Echo Cancellation (AEC)
It is plausible that the dominant near to mid-term WebRTC usage model It is plausible that the dominant near-to-medium-term WebRTC usage
will be people using the interactive audio and video capabilities to model will be people using the interactive audio and video
communicate with each other via web browsers running on a notebook capabilities to communicate with each other via web browsers running
computer that has built-in microphone and speakers. The notebook-as- on a notebook computer that has a built-in microphone and speakers.
communication-device paradigm presents challenging echo cancellation The notebook-as-communication-device paradigm presents challenging
problems, the specific remedy of which will not be mandated here. echo cancellation problems, the specific remedy of which will not be
However, while no specific algorithm or standard will be required by mandated here. However, while no specific algorithm or standard will
WebRTC-compatible endpoints, echo cancellation will improve the user be required by WebRTC-compatible endpoints, echo cancellation will
experience and should be implemented by the endpoint device. improve the user experience and should be implemented by the endpoint
device.
WebRTC endpoints SHOULD include an AEC or some other form of echo WebRTC endpoints SHOULD include an AEC or some other form of echo
control. On general purpose platforms (e.g. PC), it is common for control. On general-purpose platforms (e.g., a PC), it is common for
the audio capture ADC and the audio playback DAC to use different the analog-to-digital converter (ADC) for audio capture and the
digital-to-analog converter (DAC) for audio playback to use different
clocks. In these cases, such as when a webcam is used for capture clocks. In these cases, such as when a webcam is used for capture
and a separate soundcard is used for playback, the sampling rates are and a separate soundcard is used for playback, the sampling rates are
likely to differ slightly. Endpoint AECs SHOULD be robust to such likely to differ slightly. Endpoint AECs SHOULD be robust to such
conditions, unless they are shipped along with hardware that conditions, unless they are shipped along with hardware that
guarantees capture and playback to be sampled from the same clock. guarantees capture and playback to be sampled from the same clock.
Endpoints SHOULD allow the entire AEC and/or the non-linear Endpoints SHOULD allow the entire AEC and/or the nonlinear processing
processing (NLP) to be turned off for applications, such as music, (NLP) to be turned off for applications, such as music, that do not
that do not behave well with the spectral attenuation methods behave well with the spectral attenuation methods typically used in
typically used in NLPs. Similarly, endpoints SHOULD have the ability NLP. Similarly, endpoints SHOULD have the ability to detect the
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 endpoint may not have an echo For some applications where the remote endpoint may not have an echo
canceller, the local endpoint MAY include a far-end echo canceller, canceller, the local endpoint MAY include a far-end echo canceller,
but if that is the case, it SHOULD be disabled by default. but when included, 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 endpoints and legacy interoperability capabilities between WebRTC endpoints and legacy
phone systems that support G.711. phone systems that support G.711.
7. IANA Considerations 7. Security Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
8. Security Considerations
For security considerations regarding the codecs themselves please For security considerations regarding the codecs themselves, please
refer their specifications, including [RFC6716], [RFC7587], refer to their specifications, including [RFC6716], [RFC7587],
[RFC3551], [RFC3389], and [RFC4733]. Likewise, consult the RTP base [RFC3551], [RFC3389], and [RFC4733]. Likewise, consult the RTP base
specification for RTP-based security considerations. WebRTC security specification for RTP-based security considerations. WebRTC security
is further discussed in [I-D.ietf-rtcweb-security] and is further discussed in [WebRTC-SEC], [WebRTC-SEC-ARCH], and
[I-D.ietf-rtcweb-security-arch] and [I-D.ietf-rtcweb-rtp-usage]. [WebRTC-RTP-USAGE].
Implementers should consider whether the use of variable bitrate is
appropriate for their application based on [RFC6562]. Encryption and
authentication issues are beyond the scope of this document.
9. Acknowledgements
This draft incorporates ideas and text from various other drafts. In Using the guidelines in [RFC6562], implementers should consider
particular we would like to acknowledge, and say thanks for, work we whether the use of variable bitrate is appropriate for their
incorporated from Harald Alvestrand and Cullen Jennings. application. Encryption and authentication issues are beyond the
scope of this document.
10. References 8. References
10.1. Normative References 8.1. 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>.
[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,
DOI 10.17487/RFC3551, July 2003, DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>. <http://www.rfc-editor.org/info/rfc3551>.
skipping to change at page 7, line 5 skipping to change at page 6, line 42
[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>.
[RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format [RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for the Opus Speech and Audio Codec", RFC 7587, for the Opus Speech and Audio Codec", RFC 7587,
DOI 10.17487/RFC7587, June 2015, DOI 10.17487/RFC7587, June 2015,
<http://www.rfc-editor.org/info/rfc7587>. <http://www.rfc-editor.org/info/rfc7587>.
10.2. Informative References [G.711] ITU-T, "Pulse code modulation (PCM) of voice frequencies",
ITU-T Recommendation G.711, November 1988,
<http://www.itu.int/rec/T-REC-G.711-198811-I/en>.
[I-D.ietf-rtcweb-security] 8.2. Informative References
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-08 (work in progress), February 2015.
[I-D.ietf-rtcweb-security-arch] [WebRTC-SEC]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "Security Considerations for WebRTC", Work
rtcweb-security-arch-11 (work in progress), March 2015. in Progress, draft-ietf-rtcweb-security-08, February 2015.
[I-D.ietf-rtcweb-rtp-usage] [WebRTC-SEC-ARCH]
Rescorla, E., "WebRTC Security Architecture", Work in
Progress, draft-ietf-rtcweb-security-arch-11, March 2015.
[WebRTC-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-25 (work in progress), June Work in Progress, draft-ietf-rtcweb-rtp-usage-26, March
2015. 2016.
[I-D.ietf-rtcweb-audio-codecs-for-interop] [RFC7875] Proust, S., Ed., "Additional WebRTC Audio Codecs for
Proust, S., "Additional WebRTC audio codecs for Interoperability", RFC 7875, DOI 10.17487/RFC7875, May
interoperability.", draft-ietf-rtcweb-audio-codecs-for- 2016, <http://www.rfc-editor.org/info/rfc7875>.
interop-05 (work in progress), February 2016.
Acknowledgements
This document incorporates ideas and text from various other
documents. In particular, we would like to acknowledge, and say
thanks for, work we incorporated from Harald Alvestrand and Cullen
Jennings.
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 United States
Email: jmvalin@jmvalin.ca Email: jmvalin@jmvalin.ca
Cary Bran Cary Bran
Plantronics Plantronics
345 Encinial Street 345 Encinial Street
Santa Cruz, CA 95060 Santa Cruz, CA 95060
USA United States
Phone: +1 206 661-2398 Phone: +1 206 661-2398
Email: cary.bran@plantronics.com Email: cary.bran@plantronics.com
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