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.
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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
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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
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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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