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
	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 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
	(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
	skipping to change at page 2, line 17		skipping to change at page 2, line 17
	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
	skipping to change at page 4, line 12		skipping to change at page 4, line 12
	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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