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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