draft-ietf-rtcweb-audio-07.txt   draft-ietf-rtcweb-audio-08.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: April 27, 2015 Plantronics Expires: November 1, 2015 Plantronics
October 24, 2014 April 30, 2015
WebRTC Audio Codec and Processing Requirements WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-07 draft-ietf-rtcweb-audio-08
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 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.
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
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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 April 27, 2015. This Internet-Draft will expire on November 1, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2015 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
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the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
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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. References . . . . . . . . . . . . . . . . . . . . . . . . . 5 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 5
10.1. Normative References . . . . . . . . . . . . . . . . . . 5 10.1. Normative References . . . . . . . . . . . . . . . . . . 6
10.2. Informative References . . . . . . . . . . . . . . . . . 6 10.2. Informative References . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 6 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 client audio processing and codec requirements for WebRTC endpoint
implementations. implementations.
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", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
document are to be interpreted as described in RFC 2119 [RFC2119]. "OPTIONAL" in this document are to be interpreted as described in RFC
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
clients, 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 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 endpoints are REQUIRED to implement the following audio
codecs:
o Opus [RFC6716] with the payload format specified in o Opus [RFC6716] with the payload format specified in
[I-D.ietf-payload-rtp-opus]. [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.
o The audio/telephone-event media format as specified in [RFC4733]. o The audio/telephone-event media format as specified in [RFC4733].
WebRTC clients are REQUIRED to be able to generate and consume the WebRTC endpoints are REQUIRED to be able to generate and consume
following events: 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 |
+------------+--------------------------------+-----------+ +------------+--------------------------------+-----------+
For all cases where the client is able to process audio at a sampling For all cases where the endpoint is able to process audio at a
rate higher than 8 kHz, it is RECOMMENDED that Opus be offered before sampling rate higher than 8 kHz, it is RECOMMENDED that Opus be
PCMA/PCMU. For Opus, all modes MUST be supported on the decoder offered before PCMA/PCMU. For Opus, all modes MUST be supported on
side. The choice of encoder-side modes is left to the implementer. the decoder side. The choice of encoder-side modes is left to the
Clients MAY use the offer/answer mechanism to signal a preference for implementer. Endpoints MAY use the offer/answer mechanism to signal
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
[I-D.ietf-rtcweb-audio-codecs-for-interop]. [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
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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
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 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to
a root mean square (RMS) level of 2600. Only active speech should be a root mean square (RMS) level of 2600. Only active speech should be
considered in the RMS calculation. If the client has control over considered in the RMS calculation. If the endpoint has control over
the entire audio capture path, as is typically the case for a regular the entire audio capture path, as is typically the case for a regular
phone, then it is RECOMMENDED that the gain be adjusted in such a way phone, then it is RECOMMENDED that the gain be adjusted in such a way
that active speech have a level of 2600 (-19 dBm0) for an average that active speech have a level of 2600 (-19 dBm0) for an average
speaker. If the client does not have control over the entire audio speaker. If the endpoint does not have control over the entire audio
capture, as is typically the case for a software client, then the capture, as is typically the case for a software endpoint, then the
client SHOULD use automatic gain control (AGC) to dynamically adjust endpoint SHOULD use automatic gain control (AGC) to dynamically
the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop sharing adjust the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop
applications, the level SHOULD NOT be automatically adjusted and the sharing applications, the level SHOULD NOT be automatically adjusted
client SHOULD allow the user to set the gain manually. 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, clients 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 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 endpoints, echo cancellation will improve the user
experience and should be implemented by the endpoint device. experience and should be implemented by the endpoint device.
WebRTC clients SHOULD include an AEC or some other form of echo WebRTC endpoints SHOULD include an AEC or some other form of echo
control and if that is not possible, the clients SHOULD ensure that control. On general purpose platforms (e.g. PC), it is common for
the speaker-to-microphone gain is below unity at all frequencies to the audio capture ADC and the audio playback DAC to use different
avoid instability when none of the client has echo control. For clocks. In these cases, such as when a webcam is used for capture
clients that do not control the audio capture and playback hardware, and a separate soundcard is used for playback, the sampling rates are
it is RECOMMENDED to support echo cancellation between devices likely to differ slightly. Endpoint AECs SHOULD be robust to such
running at slightly different sampling rates, such as when a webcam conditions, unless they are shipped along with hardware that
is used for microphone. guarantees capture and playback to be sampled from the same clock.
Clients SHOULD allow the entire AEC and/or the non-linear processing Endpoints SHOULD allow the entire AEC and/or the non-linear
(NLP) to be turned off for applications, such as music, that do not processing (NLP) to be turned off for applications, such as music,
behave well with the spectral attenuation methods typically used in that do not behave well with the spectral attenuation methods
NLPs. Similarly, clients SHOULD have the ability to detect the typically used in NLPs. Similarly, endpoints SHOULD have the ability
presence of a headset and disable echo cancellation. to detect the 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 endpoint may not have an echo
canceller, the local client MAY include a far-end echo canceller, but canceller, the local endpoint MAY include a far-end echo canceller,
if that is the case, it SHOULD be disabled by default. but 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 endpoints applications
legacy phone systems that support G.711. and 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
For security considerations regarding the codecs themselves please
refer their specifications, including [RFC6716],
[I-D.ietf-payload-rtp-opus], [RFC3551], [RFC3389], and [RFC4733].
Likewise, consult the RTP base specification for security RTP-based
security considerations. WebRTC security is further discussed in
[I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch] and
[I-D.ietf-rtcweb-rtp-usage].
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.
skipping to change at page 6, line 18 skipping to change at page 6, line 29
[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.
[I-D.ietf-payload-rtp-opus] [I-D.ietf-payload-rtp-opus]
Spittka, J., Vos, K., and J. Valin, "RTP Payload Format Spittka, J., Vos, K., and J. Valin, "RTP Payload Format
for Opus Speech and Audio Codec", draft-ietf-payload-rtp- for the Opus Speech and Audio Codec", draft-ietf-payload-
opus-03 (work in progress), July 2014. rtp-opus-11 (work in progress), April 2015.
10.2. Informative References 10.2. Informative References
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-08 (work in progress), February 2015.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-11 (work in progress), March 2015.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-23 (work in progress), March
2015.
[I-D.ietf-rtcweb-audio-codecs-for-interop] [I-D.ietf-rtcweb-audio-codecs-for-interop]
Proust, S., Berger, E., Feiten, B., Bogineni, K., Lei, M., Proust, S., Berger, E., Feiten, B., Bogineni, K., Lei, M.,
and E. Marocco, "Additional WebRTC audio codecs for and E. Marocco, "Additional WebRTC audio codecs for
interoperability with legacy networks.", draft-ietf- interoperability.", draft-ietf-rtcweb-audio-codecs-for-
rtcweb-audio-codecs-for-interop-00 (work in progress), interop-01 (work in progress), January 2015.
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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