--- 1/draft-ietf-rtcweb-rtp-usage-16.txt 2014-08-25 09:14:35.749846585 -0700 +++ 2/draft-ietf-rtcweb-rtp-usage-17.txt 2014-08-25 09:14:35.845848892 -0700 @@ -1,21 +1,21 @@ RTCWEB Working Group C. S. Perkins Internet-Draft University of Glasgow Intended status: Standards Track M. Westerlund -Expires: January 24, 2015 Ericsson +Expires: February 26, 2015 Ericsson J. Ott Aalto University - July 23, 2014 + August 25, 2014 Web Real-Time Communication (WebRTC): Media Transport and Use of RTP - draft-ietf-rtcweb-rtp-usage-16 + draft-ietf-rtcweb-rtp-usage-17 Abstract The Web Real-Time Communication (WebRTC) framework provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web-browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context, and gives requirements for which RTP features, profiles, and extensions need to be supported. @@ -28,21 +28,21 @@ 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 and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference 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 (c) 2014 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents @@ -217,21 +217,21 @@ Also described are the core extensions providing essential features that all WebRTC implementations need to implement to function effectively on today's networks. 4.1. RTP and RTCP The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be implemented as the media transport protocol for WebRTC. RTP itself comprises two parts: the RTP data transfer protocol, and the RTP 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 functionality implementations of RTP, but are REQUIRED in all WebRTC implementations: o Support for use of multiple simultaneous SSRC values in a single RTP session, including support for RTP end-points that send many SSRC values simultaneously, following [RFC3550] and [I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for multi-SSRC sessions defined in @@ -1928,59 +1928,59 @@ 16.2. Informative References [I-D.ietf-avtcore-multiplex-guidelines] Westerlund, M., Perkins, C., and H. Alvestrand, "Guidelines for using the Multiplexing Features of RTP to Support Multiple Media Streams", draft-ietf-avtcore- multiplex-guidelines-02 (work in progress), January 2014. [I-D.ietf-avtcore-rtp-topologies-update] Westerlund, M. and S. Wenger, "RTP Topologies", draft- - ietf-avtcore-rtp-topologies-update-02 (work in progress), - May 2014. + ietf-avtcore-rtp-topologies-update-04 (work in progress), + August 2014. [I-D.ietf-avtext-rtp-grouping-taxonomy] Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro, "A Taxonomy of Grouping Semantics and Mechanisms for Real- Time Transport Protocol (RTP) Sources", draft-ietf-avtext- rtp-grouping-taxonomy-02 (work in progress), June 2014. [I-D.ietf-mmusic-msid] Alvestrand, H., "WebRTC MediaStream Identification in the Session Description Protocol", draft-ietf-mmusic-msid-06 (work in progress), June 2014. [I-D.ietf-mmusic-sdp-bundle-negotiation] Holmberg, C., Alvestrand, H., and C. Jennings, "Negotiating Media Multiplexing Using the Session 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] Westerlund, M., "How to Write an RTP Payload Format", draft-ietf-payload-rtp-howto-13 (work in progress), January 2014. [I-D.ietf-rmcat-cc-requirements] Jesup, R., "Congestion Control Requirements For RMCAT", draft-ietf-rmcat-cc-requirements-05 (work in progress), July 2014. [I-D.ietf-rtcweb-audio] Valin, J. and C. Bran, "WebRTC Audio Codec and Processing Requirements", draft-ietf-rtcweb-audio-05 (work in progress), February 2014. [I-D.ietf-rtcweb-overview] Alvestrand, H., "Overview: Real Time Protocols for - Browser-based Applications", draft-ietf-rtcweb-overview-10 - (work in progress), June 2014. + Browser-based Applications", draft-ietf-rtcweb-overview-11 + (work in progress), August 2014. [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-14 (work in progress), February 2014. [I-D.ietf-tsvwg-rtcweb-qos] Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. Polk, "DSCP and other packet markings for RTCWeb QoS",