draft-ietf-rtcweb-rtp-usage-16.txt		draft-ietf-rtcweb-rtp-usage-17.txt
	RTCWEB Working Group C. S. Perkins		RTCWEB Working Group C. S. Perkins
	Internet-Draft University of Glasgow		Internet-Draft University of Glasgow
	Intended status: Standards Track M. Westerlund		Intended status: Standards Track M. Westerlund
	Expires: January 24, 2015 Ericsson		Expires: February 26, 2015 Ericsson
	J. Ott		J. Ott
	Aalto University		Aalto University
	July 23, 2014		August 25, 2014
	Web Real-Time Communication (WebRTC): Media Transport and Use of RTP		Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
	draft-ietf-rtcweb-rtp-usage-16		draft-ietf-rtcweb-rtp-usage-17
	Abstract		Abstract
	The Web Real-Time Communication (WebRTC) framework provides support		The Web Real-Time Communication (WebRTC) framework provides support
	for direct interactive rich communication using audio, video, text,		for direct interactive rich communication using audio, video, text,
	collaboration, games, etc. between two peers' web-browsers. This		collaboration, games, etc. between two peers' web-browsers. This
	memo describes the media transport aspects of the WebRTC framework.		memo describes the media transport aspects of the WebRTC framework.
	It specifies how the Real-time Transport Protocol (RTP) is used in		It specifies how the Real-time Transport Protocol (RTP) is used in
	the WebRTC context, and gives requirements for which RTP features,		the WebRTC context, and gives requirements for which RTP features,
	profiles, and extensions need to be supported.		profiles, and extensions need to be supported.
	skipping to change at page 1, line 39		skipping to change at page 1, line 39
	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 January 24, 2015.		This Internet-Draft will expire on February 26, 2015.
	Copyright Notice		Copyright Notice
	Copyright (c) 2014 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
	skipping to change at page 5, line 40		skipping to change at page 5, line 40
	Also described are the core extensions providing essential features		Also described are the core extensions providing essential features
	that all WebRTC implementations need to implement to function		that all WebRTC implementations need to implement to function
	effectively on today's networks.		effectively on today's networks.
	4.1. RTP and RTCP		4.1. RTP and RTCP
	The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be		The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be
	implemented as the media transport protocol for WebRTC. RTP itself		implemented as the media transport protocol for WebRTC. RTP itself
	comprises two parts: the RTP data transfer protocol, and the RTP		comprises two parts: the RTP data transfer protocol, and the RTP
	control protocol (RTCP). RTCP is a fundamental and integral part of		control protocol (RTCP). RTCP is a fundamental and integral part of
	RTP, and MUST be implemented in all WebRTC applications.		RTP, and MUST be implemented and used in all WebRTC applications.
	The following RTP and RTCP features are sometimes omitted in limited		The following RTP and RTCP features are sometimes omitted in limited
	functionality implementations of RTP, but are REQUIRED in all WebRTC		functionality implementations of RTP, but are REQUIRED in all WebRTC
	implementations:		implementations:
	o Support for use of multiple simultaneous SSRC values in a single		o Support for use of multiple simultaneous SSRC values in a single
	RTP session, including support for RTP end-points that send many		RTP session, including support for RTP end-points that send many
	SSRC values simultaneously, following [RFC3550] and		SSRC values simultaneously, following [RFC3550] and
	[I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for		[I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for
	multi-SSRC sessions defined in		multi-SSRC sessions defined in
	skipping to change at page 41, line 30		skipping to change at page 41, line 30
	16.2. Informative References		16.2. Informative References
	[I-D.ietf-avtcore-multiplex-guidelines]		[I-D.ietf-avtcore-multiplex-guidelines]
	Westerlund, M., Perkins, C., and H. Alvestrand,		Westerlund, M., Perkins, C., and H. Alvestrand,
	"Guidelines for using the Multiplexing Features of RTP to		"Guidelines for using the Multiplexing Features of RTP to
	Support Multiple Media Streams", draft-ietf-avtcore-		Support Multiple Media Streams", draft-ietf-avtcore-
	multiplex-guidelines-02 (work in progress), January 2014.		multiplex-guidelines-02 (work in progress), January 2014.
	[I-D.ietf-avtcore-rtp-topologies-update]		[I-D.ietf-avtcore-rtp-topologies-update]
	Westerlund, M. and S. Wenger, "RTP Topologies", draft-		Westerlund, M. and S. Wenger, "RTP Topologies", draft-
	ietf-avtcore-rtp-topologies-update-02 (work in progress),		ietf-avtcore-rtp-topologies-update-04 (work in progress),
	May 2014.		August 2014.
	[I-D.ietf-avtext-rtp-grouping-taxonomy]		[I-D.ietf-avtext-rtp-grouping-taxonomy]
	Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,		Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,
	"A Taxonomy of Grouping Semantics and Mechanisms for Real-		"A Taxonomy of Grouping Semantics and Mechanisms for Real-
	Time Transport Protocol (RTP) Sources", draft-ietf-avtext-		Time Transport Protocol (RTP) Sources", draft-ietf-avtext-
	rtp-grouping-taxonomy-02 (work in progress), June 2014.		rtp-grouping-taxonomy-02 (work in progress), June 2014.
	[I-D.ietf-mmusic-msid]		[I-D.ietf-mmusic-msid]
	Alvestrand, H., "WebRTC MediaStream Identification in the		Alvestrand, H., "WebRTC MediaStream Identification in the
	Session Description Protocol", draft-ietf-mmusic-msid-06		Session Description Protocol", draft-ietf-mmusic-msid-06
	(work in progress), June 2014.		(work in progress), June 2014.
	[I-D.ietf-mmusic-sdp-bundle-negotiation]		[I-D.ietf-mmusic-sdp-bundle-negotiation]
	Holmberg, C., Alvestrand, H., and C. Jennings,		Holmberg, C., Alvestrand, H., and C. Jennings,
	"Negotiating Media Multiplexing Using the Session		"Negotiating Media Multiplexing Using the Session
	Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-		Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
	negotiation-07 (work in progress), April 2014.		negotiation-08 (work in progress), August 2014.
	[I-D.ietf-payload-rtp-howto]		[I-D.ietf-payload-rtp-howto]
	Westerlund, M., "How to Write an RTP Payload Format",		Westerlund, M., "How to Write an RTP Payload Format",
	draft-ietf-payload-rtp-howto-13 (work in progress),		draft-ietf-payload-rtp-howto-13 (work in progress),
	January 2014.		January 2014.
	[I-D.ietf-rmcat-cc-requirements]		[I-D.ietf-rmcat-cc-requirements]
	Jesup, R., "Congestion Control Requirements For RMCAT",		Jesup, R., "Congestion Control Requirements For RMCAT",
	draft-ietf-rmcat-cc-requirements-05 (work in progress),		draft-ietf-rmcat-cc-requirements-05 (work in progress),
	July 2014.		July 2014.
	[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-05 (work in		Requirements", draft-ietf-rtcweb-audio-05 (work in
	progress), February 2014.		progress), February 2014.
	[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-10		Browser-based Applications", draft-ietf-rtcweb-overview-11
	(work in progress), June 2014.		(work in progress), August 2014.
	[I-D.ietf-rtcweb-use-cases-and-requirements]		[I-D.ietf-rtcweb-use-cases-and-requirements]
	Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-		Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
	Time Communication Use-cases and Requirements", draft-		Time Communication Use-cases and Requirements", draft-
	ietf-rtcweb-use-cases-and-requirements-14 (work in		ietf-rtcweb-use-cases-and-requirements-14 (work in
	progress), February 2014.		progress), February 2014.
	[I-D.ietf-tsvwg-rtcweb-qos]		[I-D.ietf-tsvwg-rtcweb-qos]
	Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J.		Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J.
	Polk, "DSCP and other packet markings for RTCWeb QoS",		Polk, "DSCP and other packet markings for RTCWeb QoS",
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