draft-ietf-rtcweb-audio-00.txt		draft-ietf-rtcweb-audio-01.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: March 11, 2013 Plantronics		Expires: May 27, 2013 Plantronics
	September 7, 2012		November 23, 2012
	WebRTC Audio Codec and Processing Requirements		WebRTC Audio Codec and Processing Requirements
	draft-ietf-rtcweb-audio-00		draft-ietf-rtcweb-audio-01
	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.
	skipping to change at page 1, line 32		skipping to change at page 1, line 32
	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 March 11, 2013.		This Internet-Draft will expire on May 27, 2013.
	Copyright Notice		Copyright Notice
	Copyright (c) 2012 IETF Trust and the persons identified as the		Copyright (c) 2012 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 3, line 34		skipping to change at page 3, line 34
	WebRTC client implementations, it leaves the question of supporting		WebRTC client implementations, it leaves the question of supporting
	additional codecs to the will of the implementer.		additional codecs to the will of the implementer.
	WebRTC clients are REQUIRED to implement the following audio codecs.		WebRTC clients are REQUIRED to implement the following audio codecs.
	o Opus [RFC6716], with any ptime value up to 120 ms		o Opus [RFC6716], with 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 a
	ptime of 20 - see section 4.5.14 of [RFC3551]		ptime of 20 - see section 4.5.14 of [RFC3551]
	o Telephone Event - [RFC4734]		o Telephone Event - [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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	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.
	SHOULD include an AEC and if not, SHOULD ensure that the speaker-to-		WebRTC clients SHOULD include an AEC and if that is not possible, the
	microphone gain is below unity at all frequencies to avoid		clients SHOULD ensure that the speaker-to-microphone gain is below
	instability when none of the client has echo cancellation. For		unity at all frequencies to avoid instability when none of the client
	clients that do not control the audio capture and playback devices		has echo cancellation. For clients that do not control the audio
	directly, it is RECOMMENDED to support echo cancellation between		capture and playback devices directly, it is RECOMMENDED to support
	devices running at slight different sampling rates, such as when a		echo cancellation between devices running at slight different
	webcam is used for microphone.		sampling rates, such as when a webcam is used for microphone.
	The client SHOULD allow either the entire AEC or the non-linear		The client SHOULD allow either the entire AEC or the non-linear
	processing (NLP) to be turned off for applications, such as music,		processing (NLP) to be turned off for applications, such as music,
	that do not behave well with the spectral attenuation methods		that do not behave well with the spectral attenuation methods
	typically used in NLPs. It SHOULD have the ability to detect the		typically used in NLPs. It 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 it 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.
	skipping to change at page 6, line 24		skipping to change at page 6, line 24
	Rescorla, E., "Security Considerations for RTC-Web",		Rescorla, E., "Security Considerations for RTC-Web",
	May 2011.		May 2011.
	[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, March 1997.		Requirement Levels", BCP 14, RFC 2119, March 1997.
	[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,
	July 2003.		July 2003.
	[RFC4734] Schulzrinne, H. and T. Taylor, "Definition of Events for		[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
	Modem, Fax, and Text Telephony Signals", RFC 4734,		Digits, Telephony Tones, and Telephony Signals", RFC 4733,
	December 2006.		December 2006.
	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
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	15 lines changed or deleted		15 lines changed or added
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