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Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-21

Document type: Active Internet-Draft (rtcweb WG)
Document stream: IETF
Last updated: 2014-11-26
Intended RFC status: Unknown
Other versions: plain text, pdf, html

IETF State: WG Consensus: Waiting for Write-Up Apr 2014
Document shepherd: Cullen Jennings

IESG State: I-D Exists
Responsible AD: (None)
Send notices to: No addresses provided

RTCWEB Working Group                                       C. S. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Standards Track                           M. Westerlund
Expires: May 30, 2015                                           Ericsson
                                                                  J. Ott
                                                        Aalto University
                                                       November 26, 2014

  Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
                     draft-ietf-rtcweb-rtp-usage-21

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.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   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 May 30, 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

Perkins, et al.           Expires May 30, 2015                  [Page 1]
Internet-Draft               RTP for WebRTC                November 2014

   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Rationale . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   4.  WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . .   5
     4.1.  RTP and RTCP  . . . . . . . . . . . . . . . . . . . . . .   5
     4.2.  Choice of the RTP Profile . . . . . . . . . . . . . . . .   7
     4.3.  Choice of RTP Payload Formats . . . . . . . . . . . . . .   8
     4.4.  Use of RTP Sessions . . . . . . . . . . . . . . . . . . .  10
     4.5.  RTP and RTCP Multiplexing . . . . . . . . . . . . . . . .  10
     4.6.  Reduced Size RTCP . . . . . . . . . . . . . . . . . . . .  11
     4.7.  Symmetric RTP/RTCP  . . . . . . . . . . . . . . . . . . .  11
     4.8.  Choice of RTP Synchronisation Source (SSRC) . . . . . . .  12
     4.9.  Generation of the RTCP Canonical Name (CNAME) . . . . . .  12
     4.10. Handling of Leap Seconds  . . . . . . . . . . . . . . . .  13
   5.  WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . .  13
     5.1.  Conferencing Extensions and Topologies  . . . . . . . . .  13
       5.1.1.  Full Intra Request (FIR)  . . . . . . . . . . . . . .  15
       5.1.2.  Picture Loss Indication (PLI) . . . . . . . . . . . .  15
       5.1.3.  Slice Loss Indication (SLI) . . . . . . . . . . . . .  16
       5.1.4.  Reference Picture Selection Indication (RPSI) . . . .  16
       5.1.5.  Temporal-Spatial Trade-off Request (TSTR) . . . . . .  16
       5.1.6.  Temporary Maximum Media Stream Bit Rate Request
               (TMMBR) . . . . . . . . . . . . . . . . . . . . . . .  16
     5.2.  Header Extensions . . . . . . . . . . . . . . . . . . . .  17
       5.2.1.  Rapid Synchronisation . . . . . . . . . . . . . . . .  17
       5.2.2.  Client-to-Mixer Audio Level . . . . . . . . . . . . .  17
       5.2.3.  Mixer-to-Client Audio Level . . . . . . . . . . . . .  18
       5.2.4.  Media Stream Identification . . . . . . . . . . . . .  18
       5.2.5.  Coordination of Video Orientation . . . . . . . . . .  18
   6.  WebRTC Use of RTP: Improving Transport Robustness . . . . . .  19

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