draft-ietf-rtcweb-audio-codecs-for-interop-01.txt		draft-ietf-rtcweb-audio-codecs-for-interop-02.txt
	Network Working Group S. Proust		Network Working Group S. Proust
	Internet-Draft Orange		Internet-Draft Orange
	Intended status: Informational E. Berger		Intended status: Informational E. Berger
	Expires: July 20, 2015 Cisco		Expires: February 8, 2016 Cisco
	B. Feiten		B. Feiten
	Deutsche Telekom		Deutsche Telekom
	B. Burman		B. Burman
	Ericsson		Ericsson
	K. Bogineni		K. Bogineni
	Verizon Wireless		Verizon Wireless
	M. Lei		M. Lei
	Huawei		Huawei
	E. Marocco		E. Marocco
	Telecom Italia		Telecom Italia
	January 16, 2015		August 7, 2015
	Additional WebRTC audio codecs for interoperability.		Additional WebRTC audio codecs for interoperability.
	draft-ietf-rtcweb-audio-codecs-for-interop-01		draft-ietf-rtcweb-audio-codecs-for-interop-02
	Abstract		Abstract
	To ensure a baseline level of interoperability between WebRTC		To ensure a baseline level of interoperability between WebRTC
	clients, [I-D.ietf-rtcweb-audio] requires a minimum set of codecs.		clients, [I-D.ietf-rtcweb-audio] requires a minimum set of codecs.
	However, to maximize the possibility to establish the session without		However, to maximize the possibility to establish the session without
	the need for audio transcoding, it is also recommended to include in		the need for audio transcoding, it is also recommended to include in
	the offer other suitable audio codecs that are available to the		the offer other suitable audio codecs that are available to the
	browser.		browser.
	skipping to change at page 2, line 4		skipping to change at page 2, line 4
	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 July 20, 2015.		This Internet-Draft will expire on February 8, 2016.
	Copyright Notice		Copyright Notice
	Copyright (c) 2015 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
	skipping to change at page 4, line 18		skipping to change at page 4, line 18
	use cases with AMR-WB at 12.65 kbit/s and in the same range for		use cases with AMR-WB at 12.65 kbit/s and in the same range for
	other wideband transcoding cases. It should be stressed that if		other wideband transcoding cases. It should be stressed that if
	G.711 is used as a fall back codec for interoperation, wideband		G.711 is used as a fall back codec for interoperation, wideband
	voice quality will be lost. Such bandwidth reduction effect down		voice quality will be lost. Such bandwidth reduction effect down
	to narrow band clearly degrades the user perceived quality of		to narrow band clearly degrades the user perceived quality of
	service leading to shorter and less frequent calls. Such a switch		service leading to shorter and less frequent calls. Such a switch
	to G.711 is less than desirable or acceptable choice for		to G.711 is less than desirable or acceptable choice for
	customers. If transcoding is performed between OPUS and any other		customers. If transcoding is performed between OPUS and any other
	wideband codec, wideband communication could be maintained but		wideband codec, wideband communication could be maintained but
	with degraded quality (MOS scores of transcoding between AMR-WB		with degraded quality (MOS scores of transcoding between AMR-WB
	12.65kbit/s and OPUS at 16 kbit/s in both directions are		12.65 kbit/s and OPUS at 16 kbit/s in both directions are
	significantly lower than those of AMR-WB at 12.65kbit/s or OPUS at		significantly lower than those of AMR-WB at 12.65 kbit/s or OPUS
	16 kbit/s). Furthermore, in degraded conditions, the addition of		at 16 kbit/s). Furthermore, in degraded conditions, the addition
	defects, like audio artifacts due to packet losses, and the audio		of defects, like audio artifacts due to packet losses, and the
	effects resulting from the cascading of different packet loss		audio effects resulting from the cascading of different packet
	recovery algorithms may result in a quality below the acceptable		loss recovery algorithms may result in a quality below the
	limit for the customers.		acceptable limit for the customers.
	o Degraded user experience with respect to conversational		o Degraded user experience with respect to conversational
	interactivity: the degradation of conversational interactivity is		interactivity: the degradation of conversational interactivity is
	due to the increase of end to end latency for both directions that		due to the increase of end to end latency for both directions that
	is introduced by the transcoding operations. Transcoding requires		is introduced by the transcoding operations. Transcoding requires
	full de-packetization for decoding of the media stream (including		full de-packetization for decoding of the media stream (including
	mechanisms of de-jitter buffering and packet loss recovery) then		mechanisms of de-jitter buffering and packet loss recovery) then
	re-encoding, re-packetization and re-sending. The delays produced		re-encoding, re-packetization and re-sending. The delays produced
	by all these operations are additive and may increase the end to		by all these operations are additive and may increase the end to
	end delay beyond acceptable limits like with more than 1s end to		end delay beyond acceptable limits like with more than 1s end to
	skipping to change at page 5, line 25		skipping to change at page 5, line 25
	switched mobile telephony services and new multimedia telephony		switched mobile telephony services and new multimedia telephony
	services over IP/IMS and 4G/VoLTE, specified by GSMA as voice IMS		services over IP/IMS and 4G/VoLTE, specified by GSMA as voice IMS
	profile for VoLTE in [IR.92]. More detailed information on AMR-WB		profile for VoLTE in [IR.92]. More detailed information on AMR-WB
	can be found in [IR.36]. [IR.36] includes references for all 3GPP		can be found in [IR.36]. [IR.36] includes references for all 3GPP
	AMR-WB related specifications including detailed codec description		AMR-WB related specifications including detailed codec description
	and Source code.		and Source code.
	4.1.2. WebRTC relevant use case for AMR-WB		4.1.2. WebRTC relevant use case for AMR-WB
	The market of voice personal communication is driven by mobile		The market of voice personal communication is driven by mobile
	terminals. AMR-WB is now implemented in more than 200 devices models		terminals. AMR-WB is now implemented in several hundreds of devices
	and 85 HD mobile networks in 60 countries with a customer base of		models and more than 130 HD mobile networks in 80 countries with a
	more than 100 million. A high number of calls are consequently		customer base of more than 300 millions. A high number of calls are
	likely to occur between WebRTC clients and mobile 3GPP terminals.		consequently likely to occur between WebRTC clients and mobile 3GPP
	The use of AMR-WB by WebRTC clients would consequently allow		terminals. The use of AMR-WB by WebRTC clients would consequently
	transcoding free interoperation with all mobile 3GPP wideband		allow transcoding free interoperation with all mobile 3GPP wideband
	terminal. Besides, WebRTC clients running on mobile terminals		terminal. Besides, WebRTC clients running on mobile terminals
	(smartphones) may reuse the AMR-WB codec already implemented on these		(smartphones) may reuse the AMR-WB codec already implemented on these
	devices.		devices.
	4.1.3. Guidelines for AMR-WB usage and implementation with WebRTC		4.1.3. Guidelines for AMR-WB usage and implementation with WebRTC
	Guidelines for implementing and using AMR-WB and ensuring		Guidelines for implementing and using AMR-WB and ensuring
	interoperability with 3GPP mobile services can be found in		interoperability with 3GPP mobile services can be found in
	[TS26.114]. In order to ensure interoperability with 4G/VoLTE as		[TS26.114]. In order to ensure interoperability with 4G/VoLTE as
	specified by GSMA, the more specific IMS profile for voice derived		specified by GSMA, the more specific IMS profile for voice derived
	skipping to change at page 7, line 50		skipping to change at page 8, line 4
	Guidelines for implementing and using G.722 with purpose to ensure		Guidelines for implementing and using G.722 with purpose to ensure
	interoperability with Multimedia Telephony services overs IMS can be		interoperability with Multimedia Telephony services overs IMS can be
	found in section 7 of [TS26.114]. Additional information of G.722		found in section 7 of [TS26.114]. Additional information of G.722
	implementation in DECT can be found in [EN300175-8] and full codec		implementation in DECT can be found in [EN300175-8] and full codec
	description and C source code in [G.722].		description and C source code in [G.722].
	4.4. Other codecs		4.4. Other codecs
	Other interoperability use cases may justify the use of other codecs.		Other interoperability use cases may justify the use of other codecs.
	Some further update of this Draft may provide under this section
	additional use case description and codec implementation guidelines
	for these codecs.
	5. Security Considerations		5. Security Considerations
	6. IANA Considerations		6. IANA Considerations
	None.		None.
	7. Acknowledgements		7. Acknowledgements
	Thanks to Milan Patel for his review.		None.
	8. References		8. References
	8.1. 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, March 1997.		Requirement Levels", BCP 14, RFC 2119,
			DOI 10.17487/RFC2119, March 1997,
			<http://www.rfc-editor.org/info/rfc2119>.
	8.2. Informative references		8.2. Informative references
	[EN300175-8]		[EN300175-8]
	ETSI, "ETSI EN 300 175-8, v2.5.1: "Digital Enhanced		ETSI, "ETSI EN 300 175-8, v2.5.1: "Digital Enhanced
	Cordless Telecommunications (DECT); Common Interface (CI);		Cordless Telecommunications (DECT); Common Interface (CI);
	Part 8: Speech and audio coding and transmission".", 2009.		Part 8: Speech and audio coding and transmission".", 2009.
	[G.722] ITU, "Recommendation ITU-T G.722 (2012): "7 kHz audio-		[G.722] ITU, "Recommendation ITU-T G.722 (2012): "7 kHz audio-
	coding within 64 kbit/s".", 2012.		coding within 64 kbit/s".", 2012.
	[I-D.ietf-rtcweb-audio]		[I-D.ietf-rtcweb-audio]
	Valin, J. and C. Bran, "WebRTC Audio Codec and Processing		Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
	Requirements", draft-ietf-rtcweb-audio-07 (work in		Requirements", draft-ietf-rtcweb-audio-08 (work in
	progress), October 2014.		progress), April 2015.
	[I-D.ietf-rtcweb-overview]		[I-D.ietf-rtcweb-overview]
	Alvestrand, H., "Overview: Real Time Protocols for		Alvestrand, H., "Overview: Real Time Protocols for
	Browser-based Applications", draft-ietf-rtcweb-overview-13		Browser-based Applications", draft-ietf-rtcweb-overview-14
	(work in progress), November 2014.		(work in progress), June 2015.
	[I-D.ietf-rtcweb-use-cases-and-requirements]
	Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
	Time Communication Use-cases and Requirements", draft-
	ietf-rtcweb-use-cases-and-requirements-15 (work in
	progress), December 2014.
	[IR.36] GSMA, "Adaptive Multirate Wide Band", 2013.		[IR.36] GSMA, "Adaptive Multirate Wide Band V3.0", September 2014.
	[IR.92] GSMA, "IMS Profile for Voice and SMS", 2013.		[IR.92] GSMA, "IMS Profile for Voice and SMS V9.0", April 2015.
	[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, DOI 10.17487/RFC6716,
			September 2012, <http://www.rfc-editor.org/info/rfc6716>.
			[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
			Time Communication Use Cases and Requirements", RFC 7478,
			DOI 10.17487/RFC7478, March 2015,
			<http://www.rfc-editor.org/info/rfc7478>.
	[TS181005]		[TS181005]
	ETSI, "Telecommunications and Internet converged Services		ETSI, "Telecommunications and Internet converged Services
	and Protocols for Advanced Networking (TISPAN); Service		and Protocols for Advanced Networking (TISPAN); Service
	and Capability Requirements V3.3.1 (2009-12)", 2009.		and Capability Requirements V3.3.1 (2009-12)", 2009.
	[TS26.114]		[TS26.114]
	3GPP, "IP Multimedia Subsystem (IMS); Multimedia		3GPP, "IP Multimedia Subsystem (IMS); Multimedia
	telephony; Media handling and interaction", 2011.		telephony; Media handling and interaction V13.0.0", June
			2015.
	Authors' Addresses		Authors' Addresses
	Stephane Proust		Stephane Proust
	Orange		Orange
	2, avenue Pierre Marzin		2, avenue Pierre Marzin
	Lannion 22307		Lannion 22307
	France		France
	Email: stephane.proust@orange.com		Email: stephane.proust@orange.com
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