draft-ietf-rtcweb-audio-03.txt   draft-ietf-rtcweb-audio-04.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 18, 2014 Plantronics Expires: July 31, 2014 Plantronics
October 15, 2013 January 27, 2014
WebRTC Audio Codec and Processing Requirements WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-03 draft-ietf-rtcweb-audio-04
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 April 18, 2014. This Internet-Draft will expire on July 31, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2013 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
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
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 . . . . . . . . . . . . . . . . . . . . . . . . . 3
5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . 4 5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . 4
6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 4 6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 4
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 4 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5
8. Security Considerations . . . . . . . . . . . . . . . . . . . 5 8. Security Considerations . . . . . . . . . . . . . . . . . . . 5
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5
10. Normative References . . . . . . . . . . . . . . . . . . . . 5 10. Normative References . . . . . . . . . . . . . . . . . . . . 5
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 5 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.
2. Terminology 2. Terminology
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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 [Opus-RTP]
and any ptime value up to 120 ms and 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 any
ptime of 20 - see section 4.5.14 of [RFC3551] ptime value up to 120 ms - see section 4.5.14 of [RFC3551]
o Telephone Event - [RFC4733] o The audio/telephone-event media format as specified in [RFC4733].
WebRTC clients are REQUIRED to be able to generate and consume the
following events:
+------------+--------------------------------+-----------+
|Event Code | Event Name | Reference |
+------------+--------------------------------+-----------+
| 0 | DTMF digit "0" | RFC4733 |
| 1 | DTMF digit "1" | RFC4733 |
| 2 | DTMF digit "2" | RFC4733 |
| 3 | DTMF digit "3" | RFC4733 |
| 4 | DTMF digit "4" | RFC4733 |
| 5 | DTMF digit "5" | RFC4733 |
| 6 | DTMF digit "6" | RFC4733 |
| 7 | DTMF digit "7" | RFC4733 |
| 8 | DTMF digit "8" | RFC4733 |
| 9 | DTMF digit "9" | RFC4733 |
| 10 | 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.
4. Audio Level 4. Audio Level
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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
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.
AUTHORS' NOTE: The idea of using the same level as what the ITU-T
recommends is that it should improve inter-operability while at the
same time maintaining sufficient dynamic range and reducing the risk
of clipping. The main drawbacks are that the resulting level is
about 12 dB lower than typical "commercial music" levels and it
leaves room for ill-behaved clients to be much louder than a normal
client. While using music-type levels is not really an option (it
would require using the same compressor-limitors that studios use),
it would be possible to have a level slightly higher (e.g. 3 dB)
than what is recommended above without causing interoperability
problems.
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 client 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 client 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 client, then the
client SHOULD use automatic gain control (AGC) to dynamically adjust client SHOULD use automatic gain control (AGC) to dynamically adjust
the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop sharing the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop sharing
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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.
WebRTC clients SHOULD include an AEC and if that is not possible, the WebRTC clients SHOULD include an AEC or some other form of echo
clients SHOULD ensure that the speaker-to-microphone gain is below control and if that is not possible, the clients SHOULD ensure that
unity at all frequencies to avoid instability when none of the client the speaker-to-microphone gain is below unity at all frequencies to
has echo cancellation. For clients that do not control the audio avoid instability when none of the client has echo control. For
capture and playback devices directly, it is RECOMMENDED to support clients that do not control the audio capture and playback hardware,
echo cancellation between devices running at slight different it is RECOMMENDED to support echo cancellation between devices
sampling rates, such as when a webcam is used for microphone. running at slightly different sampling rates, such as when a webcam
is used for microphone.
The client SHOULD allow either the entire AEC or the non-linear Clients SHOULD allow the entire AEC and/or the non-linear 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. It SHOULD have the ability to detect the NLPs. Similarly, clients 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 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.
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.
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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
The codec requirements have no additional security considerations Implementers should consider whether the use of VBR is appropriate
other than those captured in for their application based on [RFC6562]. Encryption and
[I-D.ekr-security-considerations-for-rtc-web]. 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. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
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Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003. July 2003.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
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
Variable Bit Rate Audio with Secure RTP", RFC 6562, March
2012.
[Opus-RTP] [Opus-RTP]
Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for Opus Codec", August 2013. for Opus Codec", August 2013.
[I-D.ekr-security-considerations-for-rtc-web]
Rescorla, E.K., "Security Considerations for RTC-Web", May
2011.
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
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 USA
Phone: +1 206 661-2398 Phone: +1 206 661-2398
Email: cary.bran@plantronics.com Email: cary.bran@plantronics.com
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