draft-ietf-rtcweb-rtp-usage-06.txt   draft-ietf-rtcweb-rtp-usage-07.txt 
Network Working Group C. 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: August 29, 2013 Ericsson Expires: January 16, 2014 Ericsson
J. Ott J. Ott
Aalto University Aalto University
February 25, 2013 July 15, 2013
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-06 draft-ietf-rtcweb-rtp-usage-07
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.
Status of this Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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 August 29, 2013. This Internet-Draft will expire on January 16, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 6 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . . 6 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 6
4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7
4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 8 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 10 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 10
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 10 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 10
4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 11 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12
5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 11 5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . . 12 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 13
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 13 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 13
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 13 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 14
5.1.6. Temporary Maximum Media Stream Bit Rate Request 5.1.6. Temporary Maximum Media Stream Bit Rate Request
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 13 (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 14
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 14 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 15
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 14 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 15
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 15 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 16
6.1. Negative Acknowledgements and RTP Retransmission . . . . . 15 6.1. Negative Acknowledgements and RTP Retransmission . . . . 16
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 16 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 17
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 17 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 17
7.1. Boundary Conditions and Circuit Breakers . . . . . . . . . 17 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 18
7.2. RTCP Extensions for Congestion Control . . . . . . . . . . 18 7.2. RTCP Limitations for Congestion Control . . . . . . . . . 19
7.3. RTCP Limitations for Congestion Control . . . . . . . . . 18 7.3. Congestion Control Interoperability and Legacy Systems . 19
7.4. Congestion Control Interoperability With Legacy Systems . 19
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 20 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 20
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 20 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 21
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 20 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 22 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 23
12. RTP Implementation Considerations . . . . . . . . . . . . . . 23 12. RTP Implementation Considerations . . . . . . . . . . . . . . 23
12.1. RTP Sessions and PeerConnections . . . . . . . . . . . . . 23 12.1. RTP Sessions and PeerConnections . . . . . . . . . . . . 24
12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 24 12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . 25
12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 25 12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . 25
12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 26 12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . 27
12.5. Contributing Sources and the CSRC List . . . . . . . . . . 27 12.5. Contributing Sources and the CSRC List . . . . . . . . . 28
12.6. Media Synchronization . . . . . . . . . . . . . . . . . . 27 12.6. Media Synchronization . . . . . . . . . . . . . . . . . 28
12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . . 28 12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . 29
12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 29 12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . 30
12.9. Differentiated Treatment of Flows . . . . . . . . . . . . 29 12.9. Differentiated Treatment of Flows . . . . . . . . . . . 30
13. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 31 13. Security Considerations . . . . . . . . . . . . . . . . . . . 32
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 32 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
15. Security Considerations . . . . . . . . . . . . . . . . . . . 32 15. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 33
16. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 33 16. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33
17. References . . . . . . . . . . . . . . . . . . . . . . . . . . 33 17. References . . . . . . . . . . . . . . . . . . . . . . . . . 33
17.1. Normative References . . . . . . . . . . . . . . . . . . . 33 17.1. Normative References . . . . . . . . . . . . . . . . . . 34
17.2. Informative References . . . . . . . . . . . . . . . . . . 36 17.2. Informative References . . . . . . . . . . . . . . . . . 37
Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 38 Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 38
A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 38 A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 39
A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 41 A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . 41
A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 44 A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 44
A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 44 A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . 45
A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 47 A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 47
A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . . 50 A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . 50
A.4. Translator Based . . . . . . . . . . . . . . . . . . . . . 53 A.4. Translator Based . . . . . . . . . . . . . . . . . . . . 52
A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . . 53 A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . 52
A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 54 A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 53
A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 56 A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 55
A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . . 60 A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . 58
A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 61 A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 60
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 62 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 60
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] provides a framework The Real-time Transport Protocol (RTP) [RFC3550] provides a framework
for delivery of audio and video teleconferencing data and other real- for delivery of audio and video teleconferencing data and other real-
time media applications. Previous work has defined the RTP protocol, time media applications. Previous work has defined the RTP protocol,
along with numerous profiles, payload formats, and other extensions. along with numerous profiles, payload formats, and other extensions.
When combined with appropriate signalling, these form the basis for When combined with appropriate signalling, these form the basis for
many teleconferencing systems. many teleconferencing systems.
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two peers' web-browsers. This memo describes how the RTP framework two peers' web-browsers. This memo describes how the RTP framework
is to be used in the WebRTC context. It proposes a baseline set of is to be used in the WebRTC context. It proposes a baseline set of
RTP features that are to be implemented by all WebRTC-aware end- RTP features that are to be implemented by all WebRTC-aware end-
points, along with suggested extensions for enhanced functionality. points, along with suggested extensions for enhanced functionality.
The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete
WebRTC framework, of which this memo is a part. WebRTC framework, of which this memo is a part.
The structure of this memo is as follows. Section 2 outlines our The structure of this memo is as follows. Section 2 outlines our
rationale in preparing this memo and choosing these RTP features. rationale in preparing this memo and choosing these RTP features.
Section 3 defines requirement terminology. Requirements for core RTP Section 3 defines terminology. Requirements for core RTP protocols
protocols are described in Section 4 and suggested RTP extensions are are described in Section 4 and suggested RTP extensions are described
described in Section 5. Section 6 outlines mechanisms that can in Section 5. Section 6 outlines mechanisms that can increase
increase robustness to network problems, while Section 7 describes robustness to network problems, while Section 7 describes congestion
congestion control and rate adaptation mechanisms. The discussion of control and rate adaptation mechanisms. The discussion of mandated
mandated RTP mechanisms concludes in Section 8 with a review of RTP mechanisms concludes in Section 8 with a review of performance
performance monitoring and network management tools that can be used monitoring and network management tools that can be used in the
in the WebRTC context. Section 9 gives some guidelines for future WebRTC context. Section 9 gives some guidelines for future
incorporation of other RTP and RTP Control Protocol (RTCP) extensions incorporation of other RTP and RTP Control Protocol (RTCP) extensions
into this framework. Section 10 describes requirements placed on the into this framework. Section 10 describes requirements placed on the
signalling channel. Section 11 discusses the relationship between signalling channel. Section 11 discusses the relationship between
features of the RTP framework and the WebRTC application programming features of the RTP framework and the WebRTC application programming
interface (API), and Section 12 discusses RTP implementation interface (API), and Section 12 discusses RTP implementation
considerations. This memo concludes with an appendix discussing considerations. This memo concludes with an appendix discussing
several different RTP Topologies, and how they affect the RTP several different RTP Topologies, and how they affect the RTP
session(s) and various implementation details of possible realization session(s) and various implementation details of possible realization
of central nodes. of central nodes.
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The RTP framework comprises the RTP data transfer protocol, the RTP The RTP framework comprises the RTP data transfer protocol, the RTP
control protocol, and numerous RTP payload formats, profiles, and control protocol, and numerous RTP payload formats, profiles, and
extensions. This range of add-ons has allowed RTP to meet various extensions. This range of add-ons has allowed RTP to meet various
needs that were not envisaged by the original protocol designers, and needs that were not envisaged by the original protocol designers, and
to support many new media encodings, but raises the question of what to support many new media encodings, but raises the question of what
extensions are to be supported by new implementations. The extensions are to be supported by new implementations. The
development of the WebRTC framework provides an opportunity for us to development of the WebRTC framework provides an opportunity for us to
review the available RTP features and extensions, and to define a review the available RTP features and extensions, and to define a
common baseline feature set for all WebRTC implementations of RTP. common baseline feature set for all WebRTC implementations of RTP.
This builds on the past 15 years development of RTP to mandate the This builds on the past 20 years development of RTP to mandate the
use of extensions that have shown widespread utility, while still use of extensions that have shown widespread utility, while still
remaining compatible with the wide installed base of RTP remaining compatible with the wide installed base of RTP
implementations where possible. implementations where possible.
Other RTP and RTCP extensions not discussed in this document can be Other RTP and RTCP extensions not discussed in this document can be
implemented by WebRTC end-points if they are beneficial for new use implemented by WebRTC end-points if they are beneficial for new use
cases. However, they are not necessary to address the WebRTC use cases. However, they are not necessary to address the WebRTC use
cases and requirements identified to date cases and requirements identified to date
[I-D.ietf-rtcweb-use-cases-and-requirements]. [I-D.ietf-rtcweb-use-cases-and-requirements].
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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 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. SSRC values simultaneously, following [RFC3550] and
[I-D.ietf-avtcore-rtp-multi-stream]. Support for the RTCP
optimisations for multi-SSRC sessions defined in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] is RECOMMENDED.
* (tbd: is draft-westerlund-mmusic-max-ssrc-01 needed?)
o Random choice of SSRC on joining a session; collision detection o Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (but see also Section 4.8). and resolution for SSRC values (see also Section 4.8).
o Support for reception of RTP data packets containing CSRC lists, o Support for reception of RTP data packets containing CSRC lists,
as generated by RTP mixers, and RTCP packets relating to CSRCs. as generated by RTP mixers, and RTCP packets relating to CSRCs.
o Support for sending correct synchronization information in the o Support for sending correct synchronization information in the
RTCP Sender Reports, to allow a receiver to implement lip-sync, RTCP Sender Reports, to allow a receiver to implement lip-sync,
with RECOMMENDED support for the rapid RTP synchronisation with RECOMMENDED support for the rapid RTP synchronisation
extensions (see Section 5.2.1). extensions (see Section 5.2.1).
o Support for sending and receiving RTCP SR, RR, SDES, and BYE o Support for sending and receiving RTCP SR, RR, SDES, and BYE
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especially those targeted at VoIP-only systems, do not support all of especially those targeted at VoIP-only systems, do not support all of
the above features, and in some cases do not support RTCP at all. the above features, and in some cases do not support RTCP at all.
Implementers are advised to consider the requirements for graceful Implementers are advised to consider the requirements for graceful
degradation when interoperating with legacy implementations. degradation when interoperating with legacy implementations.
Other implementation considerations are discussed in Section 12. Other implementation considerations are discussed in Section 12.
4.2. Choice of the RTP Profile 4.2. Choice of the RTP Profile
The complete specification of RTP for a particular application domain The complete specification of RTP for a particular application domain
requires the choice of an RTP Profile. For WebRTC use, the "Extended requires the choice of an RTP Profile. For WebRTC use, the Extended
Secure RTP Profile for Real-time Transport Control Protocol (RTCP)- Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as
Based Feedback (RTP/SAVPF)" [RFC5124] as extended by extended by [I-D.ietf-avtcore-avp-codecs], MUST be implemented. This
[I-D.ietf-avtcore-avp-codecs] MUST be implemented. This builds on builds on the basic RTP/AVP profile [RFC3551], the RTP profile for
the basic RTP/AVP profile [RFC3551], the RTP profile for RTCP-based RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP profile
feedback (RTP/AVPF) [RFC4585], and the secure RTP profile (RTP/SAVP) (RTP/SAVP) [RFC3711].
[RFC3711].
The RTCP-based feedback extensions [RFC4585] are needed for the The RTCP-based feedback extensions [RFC4585] are needed for the
improved RTCP timer model, that allows more flexible transmission of improved RTCP timer model, that allows more flexible transmission of
RTCP packets in response to events, rather than strictly according to RTCP packets in response to events, rather than strictly according to
bandwidth. This is vital for being able to report congestion events. bandwidth. This is vital for being able to report congestion events.
These extensions also save RTCP bandwidth, and will commonly only use These extensions also save RTCP bandwidth, and will commonly only use
the full RTCP bandwidth allocation if there are many events that the full RTCP bandwidth allocation if there are many events that
require feedback. They are also needed to make use of the RTP require feedback. They are also needed to make use of the RTP
conferencing extensions discussed in Section 5.1. conferencing extensions discussed in Section 5.1.
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packets using the basic RTP/AVP profile or the RTP/AVPF profile; they packets using the basic RTP/AVP profile or the RTP/AVPF profile; they
MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP
packets that are generated. The default and mandatory to implement packets that are generated. The default and mandatory to implement
transforms listed in Section 5 of [RFC3711] SHALL apply. transforms listed in Section 5 of [RFC3711] SHALL apply.
Implementations MUST support DTLS-SRTP [RFC5764] for key-management. Implementations MUST support DTLS-SRTP [RFC5764] for key-management.
Other key management schemes MAY be supported. Other key management schemes MAY be supported.
4.3. Choice of RTP Payload Formats 4.3. Choice of RTP Payload Formats
Implementations MUST follow the WebRTC Audio Codec and Processing The set of mandatory to implement codecs and RTP payload formats for
Requirements [I-D.ietf-rtcweb-audio] and SHOULD follow the updated WebRTC is not specified in this memo. Implementations can support
recommendations for audio codecs in the RTP/AVP Profile any codec for which an RTP payload format and associated signalling
[I-D.ietf-avtcore-avp-codecs]. Support for other audio codecs is is defined. Implementation cannot assume that the other participants
OPTIONAL. in an RTP session understand any RTP payload format, no matter how
common; support for all RTP payload formats MUST be negotiated before
they are used.
(tbd: the mandatory to implement video codec is not yet decided) Endpoints can signal support for multiple RTP payload formats, or
multiple configurations of a single RTP payload format, as long as
each unique RTP payload format configuration uses a different RTP
payload type number. As outlined in Section 4.8, the RTP payload
type number is sometimes used to associate an RTP media stream with a
signalling context. This association is possible provided unique RTP
payload type numbers are used in each context. For example, an RTP
media stream can be associated with an SDP "m=" line by comparing the
RTP payload type numbers used by the media stream with payload types
signalled in the "a=rtpmap:" lines in the media sections of the SDP.
If RTP media streams are being associated with signalling contexts
based on the RTP payload type, then the assignment of RTP payload
type numbers MUST be unique across signalling contexts; if the same
RTP payload format configuration is used in multiple contexts, then a
different RTP payload type number has to be assigned in each context
to ensure uniqueness. If the RTP payload type number is not being
used to associated RTP media streams with a signalling context, then
the same RTP payload type number can be used to indicate the exact
same RTP payload format configuration in multiple contexts.
Endpoints MAY signal support for multiple RTP payload formats, or An endpoint that has signalled support for multiple RTP payload
multiple configurations of a single RTP payload format, provided each formats SHOULD accept data in any of those payload formats at any
payload format uses a different RTP payload type number. An endpoint time, unless it has previously signalled limitations on its decoding
that has signalled support for multiple RTP payload formats SHOULD capability. This requirement is constrained if several types of
accept data in any of those payload formats at any time, unless it media (e.g., audio and video) are sent in the same RTP session. In
has previously signalled limitations on its decoding capability. such a case, a source (SSRC) is restricted to switching only between
This requirement is constrained if several media types are sent in the RTP payload formats signalled for the type of media that is being
the same RTP session. In such a case, a source (SSRC) is restricted sent by that source; see Section 4.4. To support rapid rate
to switching only between the RTP payload formats signalled for the adaptation by changing codec, RTP does not require advance signalling
media type that is being sent by that source; see Section 4.4. To for changes between RTP payload formats that were signalled during
support rapid rate adaptation by changing codec, RTP does not require session set-up.
advance signalling for changes between RTP payload formats that were
signalled during session set-up.
An RTP sender that changes between two RTP payload types that use An RTP sender that changes between two RTP payload types that use
different RTP clock rates MUST follow the recommendations in Section different RTP clock rates MUST follow the recommendations in
4.1 of [I-D.ietf-avtext-multiple-clock-rates]. RTP receivers MUST Section 4.1 of [I-D.ietf-avtext-multiple-clock-rates]. RTP receivers
follow the recommendations in Section 4.3 of MUST follow the recommendations in Section 4.3 of
[I-D.ietf-avtext-multiple-clock-rates], in order to support sources [I-D.ietf-avtext-multiple-clock-rates], in order to support sources
that switch between clock rates in an RTP session (these that switch between clock rates in an RTP session (these
recommendations for receivers are backwards compatible with the case recommendations for receivers are backwards compatible with the case
where senders use only a single clock rate). where senders use only a single clock rate).
4.4. RTP Session Multiplexing 4.4. Use of RTP Sessions
An association amongst a set of participants communicating with RTP
is known as an RTP session. A participant can be involved in
multiple RTP sessions at the same time. In a multimedia session,
each medium has typically been carried in a separate RTP session with
its own RTCP packets (i.e., one RTP session for the audio, with a
separate RTP session using a different transport address for the
video; if SDP is used, this corresponds to one RTP session for each
"m=" line in the SDP). WebRTC implementations of RTP are REQUIRED to
implement support for multimedia sessions in this way, for
compatibility with legacy systems.
In today's networks, however, with the widespread use of Network An association amongst a set of participants communicating using RTP
Address/Port Translators (NAT/NAPT) and Firewalls (FW), it is is known as an RTP session. A participant can be involved in several
desirable to reduce the number of transport addresses used by real- RTP sessions at the same time. In a multimedia session, each type of
time media applications using RTP by combining all RTP media streams media has typically been carried in a separate RTP session (e.g.,
in a single RTP session. Using a single RTP session also effects the using one RTP session for the audio, and a separate RTP session using
possibility for differentiated treatment of media flows. This is different transport addresses for the video). WebRTC implementations
further discussed in Section 12.9. WebRTC implementations of RTP are of RTP are REQUIRED to implement support for multimedia sessions in
REQUIRED to support transport of all RTP media streams, independent this way, separating each session using different transport-layer
of media type, in a single RTP session according to addresses (e.g., different UDP ports) for compatibility with legacy
[I-D.ietf-avtcore-multi-media-rtp-session]. If such RTP session systems.
set-up is to be used, this MUST be negotiated during the signalling
phase [I-D.ietf-mmusic-sdp-bundle-negotiation].
Support for multiple RTP sessions over a single UDP flow as defined In modern day networks, however, with the widespread use of network
by [I-D.westerlund-avtcore-transport-multiplexing] is RECOMMENDED/ address/port translators (NAT/NAPT) and firewalls, it is desirable to
OPTIONAL. If multiple RTP sessions are to be multiplexed onto a reduce the number of transport-layer flows used by RTP applications.
single UDP flow, this MUST be negotiated during the signalling phase. This can be done by sending all the RTP media streams in a single RTP
session, which will comprise a single transport-layer flow (this will
prevent the use of some quality-of-service mechanisms, as discussed
in Section 12.9). Implementations are REQUIRED to support transport
of all RTP media streams, independent of media type, in a single RTP
session according to [I-D.ietf-avtcore-multi-media-rtp-session]. If
such RTP session set-up is to be used, this MUST be negotiated during
the signalling phase [I-D.ietf-mmusic-sdp-bundle-negotiation].
(tbd: No consensus on the level of support of Multiple RTP It is also possible to use a shim-based approach to run multiple RTP
sessions over a single UDP flow.) sessions on a single transport-layer flow. This gives advantages in
some gateway scenarios, and makes it easy to distinguish groups of
RTP media streams that might need distinct processing. One way of
doing this is described in
[I-D.westerlund-avtcore-transport-multiplexing]. At the time of this
writing, there is no consensus to use a shim-based approach in WebRTC
implementations.
Further discussion about when different RTP session structures and Further discussion about when different RTP session structures and
multiplexing methods are suitable can be found in the memo on multiplexing methods are suitable can be found in
Guidelines for using the Multiplexing Features of RTP
[I-D.westerlund-avtcore-multiplex-architecture]. [I-D.westerlund-avtcore-multiplex-architecture].
4.5. RTP and RTCP Multiplexing 4.5. RTP and RTCP Multiplexing
Historically, RTP and RTCP have been run on separate transport layer Historically, RTP and RTCP have been run on separate transport layer
addresses (e.g., two UDP ports for each RTP session, one port for RTP addresses (e.g., two UDP ports for each RTP session, one port for RTP
and one port for RTCP). With the increased use of Network Address/ and one port for RTCP). With the increased use of Network Address/
Port Translation (NAPT) this has become problematic, since Port Translation (NAPT) this has become problematic, since
maintaining multiple NAT bindings can be costly. It also complicates maintaining multiple NAT bindings can be costly. It also complicates
firewall administration, since multiple ports need to be opened to firewall administration, since multiple ports need to be opened to
allow RTP traffic. To reduce these costs and session set-up times, allow RTP traffic. To reduce these costs and session set-up times,
support for multiplexing RTP data packets and RTCP control packets on support for multiplexing RTP data packets and RTCP control packets on
a single port for each RTP session is REQUIRED, as specified in a single port for each RTP session is REQUIRED, as specified in
[RFC5761]. For backwards compatibility, implementations are also [RFC5761]. For backwards compatibility, implementations are also
REQUIRED to support sending of RTP and RTCP to separate destination REQUIRED to support RTP and RTCP sent on separate transport-layer
ports. addresses.
Note that the use of RTP and RTCP multiplexed onto a single transport Note that the use of RTP and RTCP multiplexed onto a single transport
port ensures that there is occasional traffic sent on that port, even port ensures that there is occasional traffic sent on that port, even
if there is no active media traffic. This can be useful to keep NAT if there is no active media traffic. This can be useful to keep NAT
bindings alive, and is the recommend method for application level bindings alive, and is the recommend method for application level
keep-alives of RTP sessions [RFC6263]. keep-alives of RTP sessions [RFC6263].
4.6. Reduced Size RTCP 4.6. Reduced Size RTCP
RTCP packets are usually sent as compound RTCP packets, and [RFC3550] RTCP packets are usually sent as compound RTCP packets, and [RFC3550]
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or Receiver Report (RR) packet. When using frequent RTCP feedback or Receiver Report (RR) packet. When using frequent RTCP feedback
messages under the RTP/AVPF Profile [RFC4585] these statistics are messages under the RTP/AVPF Profile [RFC4585] these statistics are
not needed in every packet, and unnecessarily increase the mean RTCP not needed in every packet, and unnecessarily increase the mean RTCP
packet size. This can limit the frequency at which RTCP packets can packet size. This can limit the frequency at which RTCP packets can
be sent within the RTCP bandwidth share. be sent within the RTCP bandwidth share.
To avoid this problem, [RFC5506] specifies how to reduce the mean To avoid this problem, [RFC5506] specifies how to reduce the mean
RTCP message size and allow for more frequent feedback. Frequent RTCP message size and allow for more frequent feedback. Frequent
feedback, in turn, is essential to make real-time applications feedback, in turn, is essential to make real-time applications
quickly aware of changing network conditions, and to allow them to quickly aware of changing network conditions, and to allow them to
adapt their transmission and encoding behaviour. Support for sending adapt their transmission and encoding behaviour. Support for non-
RTCP feedback packets as [RFC5506] non-compound packets is REQUIRED, compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be
but MUST be negotiated using the signalling channel before use. For negotiated using the signalling channel before use. For backwards
backwards compatibility, implementations are also REQUIRED to support compatibility, implementations are also REQUIRED to support the use
the use of compound RTCP feedback packets if the remote endpoint does of compound RTCP feedback packets if the remote endpoint does not
not agree to the use of non-compound RTCP in the signalling exchange. agree to the use of non-compound RTCP in the signalling exchange.
4.7. Symmetric RTP/RTCP 4.7. Symmetric RTP/RTCP
To ease traversal of NAT and firewall devices, implementations are To ease traversal of NAT and firewall devices, implementations are
REQUIRED to implement and use Symmetric RTP [RFC4961]. This requires REQUIRED to implement and use Symmetric RTP [RFC4961]. The reasons
that the IP address and port used for sending and receiving RTP and for using symmetric RTP is primarily to avoid issues with NAT and
RTCP packets are identical. The reasons for using symmetric RTP is Firewalls by ensuring that the flow is actually bi-directional and
primarily to avoid issues with NAT and Firewalls by ensuring that the thus kept alive and registered as flow the intended recipient
flow is actually bi-directional and thus kept alive and registered as actually wants. In addition, it saves resources, specifically ports
flow the intended recipient actually wants. In addition, it saves at the end-points, but also in the network as NAT mappings or
resources, specifically ports at the end-points, but also in the firewall state is not unnecessary bloated. Also the amount of QoS
network as NAT mappings or firewall state is not unnecessary bloated. state is reduced.
Also the amount of QoS state is reduced.
4.8. Choice of RTP Synchronisation Source (SSRC) 4.8. Choice of RTP Synchronisation Source (SSRC)
Implementations are REQUIRED to support signalled RTP SSRC values, Implementations are REQUIRED to support signalled RTP synchronisation
using the "a=ssrc:" SDP attribute defined in Sections 4.1 and 5 of source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined
[RFC5576], and MUST also support the "previous-ssrc" source attribute in Section 4.1 and Section 5 of [RFC5576]. Implementations MUST also
defined in Section 6.2 of [RFC5576]. Other attributes defined in support the "previous-ssrc" source attribute defined in Section 6.2
[RFC5576] MAY be supported. of [RFC5576]. Other per-SSRC attributes defined in [RFC5576] MAY be
supported.
Use of the "a=ssrc:" attribute is OPTIONAL. Implementations MUST Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP
support random SSRC assignment, and MUST support SSRC collision session is OPTIONAL. Implementations MUST be prepared to accept RTP
detection and resolution, both according to [RFC3550]. and RTCP packets using SSRCs that have not been explicitly signalled
ahead of time. Implementations MUST support random SSRC assignment,
and MUST support SSRC collision detection and resolution, according
to [RFC3550]. When using signalled SSRC values, collision detection
MUST be performed as described in Section 5 of [RFC5576].
It is often desirable to associate an RTP media stream with a non-RTP
context (e.g., to associate an RTP media stream with an "m=" line in
a session description formatted using SDP). If SSRCs are signalled
this is straightforward (in SDP the "a=ssrc:" line will be at the
media level, allowing a direct association with an "m=" line). If
SSRCs are not signalled, the RTP payload type numbers used in an RTP
media stream are often sufficient to associate that media stream with
a signalling context (e.g., if RTP payload type numbers are assigned
as described in Section 4.3 of this memo, the RTP payload types used
by an RTP media stream can be compared with values in SDP "a=rtpmap:"
lines, which are at the media level in SDP, and so map to an "m="
line).
4.9. Generation of the RTCP Canonical Name (CNAME) 4.9. Generation of the RTCP Canonical Name (CNAME)
The RTCP Canonical Name (CNAME) provides a persistent transport-level The RTCP Canonical Name (CNAME) provides a persistent transport-level
identifier for an RTP endpoint. While the Synchronisation Source identifier for an RTP endpoint. While the Synchronisation Source
(SSRC) identifier for an RTP endpoint can change if a collision is (SSRC) identifier for an RTP endpoint can change if a collision is
detected, or when the RTP application is restarted, its RTCP CNAME is detected, or when the RTP application is restarted, its RTCP CNAME is
meant to stay unchanged, so that RTP endpoints can be uniquely meant to stay unchanged, so that RTP endpoints can be uniquely
identified and associated with their RTP media streams within a set identified and associated with their RTP media streams within a set
of related RTP sessions. For proper functionality, each RTP endpoint of related RTP sessions. For proper functionality, each RTP endpoint
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full functionality, or extremely useful to improve on the baseline full functionality, or extremely useful to improve on the baseline
performance, in the WebRTC application context. One set of these performance, in the WebRTC application context. One set of these
extensions is related to conferencing, while others are more generic extensions is related to conferencing, while others are more generic
in nature. The following subsections describe the various RTP in nature. The following subsections describe the various RTP
extensions mandated or suggested for use within the WebRTC context. extensions mandated or suggested for use within the WebRTC context.
5.1. Conferencing Extensions 5.1. Conferencing Extensions
RTP is inherently a group communication protocol. Groups can be RTP is inherently a group communication protocol. Groups can be
implemented using a centralised server, multi-unicast, or using IP implemented using a centralised server, multi-unicast, or using IP
multicast. While IP multicast was popular in early deployments, in multicast. While IP multicast is popular in IPTV systems, overlay-
today's practice, overlay-based conferencing dominates, typically based topologies dominate in interactive conferencing environments.
using one or more central servers to connect endpoints in a star or Such overlay-based topologies typically use one or more central
flat tree topology. These central servers can be implemented in a servers to connect end-points in a star or flat tree topology. These
number of ways as discussed in Appendix A, and in the memo on RTP central servers can be implemented in a number of ways as discussed
Topologies [I-D.westerlund-avtcore-rtp-topologies-update]. in Appendix A, and in the memo on RTP Topologies
[I-D.westerlund-avtcore-rtp-topologies-update].
As discussed in Section 3.7 of Not all of the possible the overlay-based topologies are suitable for
[I-D.westerlund-avtcore-rtp-topologies-update], the use of a video use in the WebRTC environment. Specifically:
switching MCU makes the use of RTCP for congestion control, or any
type of quality reports, very problematic. Also, as discussed in
section 3.8 of [I-D.westerlund-avtcore-rtp-topologies-update], the
use of a content modifying MCU with RTCP termination breaks RTP loop
detection and removes the ability for receivers to identify active
senders. RTP Transport Translators (Topo-Translator) are not of
immediate interest to WebRTC, although the main difference compared
to point to point is the possibility of seeing multiple different
transport paths in any RTCP feedback. Accordingly, only Point to
Point (Topo-Point-to-Point), Multiple concurrent Point to Point
(Mesh) and RTP Mixers (Topo-Mixer) topologies are needed to achieve
the use-cases to be supported in WebRTC initially. These RECOMMENDED
topologies are expected to be supported by all WebRTC end-points
(these topologies require no special RTP-layer support in the end-
point if the RTP features mandated in this memo are implemented).
The RTP extensions described below to be used with centralised o The use of video switching MCUs makes the use of RTCP for
conferencing -- where one RTP Mixer (e.g., a conference bridge) congestion control and quality of service reports problematic (see
receives a participant's RTP media streams and distributes them to Section 3.7 of [I-D.westerlund-avtcore-rtp-topologies-update]).
the other participants -- are not necessary for interoperability; an
RTP endpoint that does not implement these extensions will work o The use of content modifying MCUs with RTCP termination breaks RTP
correctly, but might offer poor performance. Support for the listed loop detection, and prevents receivers from identifying active
extensions will greatly improve the quality of experience and, to senders (see section 3.8 of
provide a reasonable baseline quality, some these extensions are [I-D.westerlund-avtcore-rtp-topologies-update]).
mandatory to be supported by WebRTC end-points.
o RTP Transport Translators (Topo-Translator) are not of immediate
interest to WebRTC, although the main difference compared to point
to point is the possibility of seeing multiple different transport
paths in any RTCP feedback.
Accordingly, only Point to Point (Topo-Point-to-Point), Multiple
concurrent Point to Point (Mesh) and RTP Mixers (Topo-Mixer)
topologies are needed to achieve the use-cases to be supported in
WebRTC initially. These RECOMMENDED topologies are expected to be
supported by all WebRTC end-points (these topologies require no
special RTP-layer support in the end-point if the RTP features
mandated in this memo are implemented).
The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
designed to be used with centralised conferencing, where an RTP
middlebox (e.g., a conference bridge) receives a participant's RTP
media streams and distributes them to the other participants. These
extensions are not necessary for interoperability; an RTP endpoint
that does not implement these extensions will work correctly, but
might offer poor performance. Support for the listed extensions will
greatly improve the quality of experience and, to provide a
reasonable baseline quality, some these extensions are mandatory to
be supported by WebRTC end-points.
The RTCP conferencing extensions are defined in Extended RTP Profile The RTCP conferencing extensions are defined in Extended RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/ for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio- AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124]. usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
5.1.1. Full Intra Request (FIR) 5.1.1. Full Intra Request (FIR)
The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of the The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of the
Codec Control Messages [RFC5104]. This message is used to make the Codec Control Messages [RFC5104]. This message is used to make the
mixer request a new Intra picture from a participant in the session. mixer request a new Intra picture from a participant in the session.
This is used when switching between sources to ensure that the This is used when switching between sources to ensure that the
receivers can decode the video or other predictive media encoding receivers can decode the video or other predictive media encoding
with long prediction chains. It is REQUIRED that WebRTC senders with long prediction chains. WebRTC senders MUST understand and
understand the react to this feedback message since it greatly react to the FIR feedback message since it greatly improves the user
improves the user experience when using centralised mixer-based experience when using centralised mixer-based conferencing; support
conferencing; support for sending the FIR message is OPTIONAL. for sending the FIR message is OPTIONAL.
5.1.2. Picture Loss Indication (PLI) 5.1.2. Picture Loss Indication (PLI)
The Picture Loss Indication is defined in Section 6.3.1 of the RTP/ The Picture Loss Indication is defined in Section 6.3.1 of the RTP/
AVPF profile [RFC4585]. It is used by a receiver to tell the sending AVPF profile [RFC4585]. It is used by a receiver to tell the sending
encoder that it lost the decoder context and would like to have it encoder that it lost the decoder context and would like to have it
repaired somehow. This is semantically different from the Full Intra repaired somehow. This is semantically different from the Full Intra
Request above as there could be multiple ways to fulfil the request. Request above as there could be multiple ways to fulfil the request.
It is REQUIRED that WebRTC senders understand and react to this WebRTC senders MUST understand and react to this feedback message as
feedback message as a loss tolerance mechanism; receivers MAY send a loss tolerance mechanism; receivers MAY send PLI messages.
PLI messages.
5.1.3. Slice Loss Indication (SLI) 5.1.3. Slice Loss Indication (SLI)
The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF
profile [RFC4585]. It is used by a receiver to tell the encoder that profile [RFC4585]. It is used by a receiver to tell the encoder that
it has detected the loss or corruption of one or more consecutive it has detected the loss or corruption of one or more consecutive
macro blocks, and would like to have these repaired somehow. The use macro blocks, and would like to have these repaired somehow. Support
of this feedback message is OPTIONAL as a loss tolerance mechanism. for this feedback message is OPTIONAL as a loss tolerance mechanism.
5.1.4. Reference Picture Selection Indication (RPSI) 5.1.4. Reference Picture Selection Indication (RPSI)
Reference Picture Selection Indication (RPSI) is defined in
Reference Picture Selection Indication (RPSI) is defined in Section Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video coding
6.3.3 of the RTP/AVPF profile [RFC4585]. Some video coding standards standards allow the use of older reference pictures than the most
allow the use of older reference pictures than the most recent one recent one for predictive coding. If such a codec is in used, and if
for predictive coding. If such a codec is in used, and if the the encoder has learned about a loss of encoder-decoder
encoder has learned about a loss of encoder-decoder synchronisation, synchronisation, a known-as-correct reference picture can be used for
a known-as-correct reference picture can be used for future coding. future coding. The RPSI message allows this to be signalled.
The RPSI message allows this to be signalled. Support for RPSI Support for RPSI messages is OPTIONAL.
messages is OPTIONAL.
5.1.5. Temporal-Spatial Trade-off Request (TSTR) 5.1.5. Temporal-Spatial Trade-off Request (TSTR)
The temporal-spatial trade-off request and notification are defined The temporal-spatial trade-off request and notification are defined
in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used
to ask the video encoder to change the trade-off it makes between to ask the video encoder to change the trade-off it makes between
temporal and spatial resolution, for example to prefer high spatial temporal and spatial resolution, for example to prefer high spatial
image quality but low frame rate. Support for TSTR requests and image quality but low frame rate. Support for TSTR requests and
notifications is OPTIONAL. notifications is OPTIONAL.
5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR) 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
This feedback message is defined in Sections 3.5.4 and 4.2.1 of the This feedback message is defined in Sections 3.5.4 and 4.2.1 of the
Codec Control Messages [RFC5104]. This message and its notification Codec Control Messages [RFC5104]. This message and its notification
message are used by a media receiver to inform the sending party that message are used by a media receiver to inform the sending party that
there is a current limitation on the amount of bandwidth available to there is a current limitation on the amount of bandwidth available to
this receiver. This can be various reasons for this: for example, an this receiver. This can be various reasons for this: for example, an
RTP mixer can use this message to limit the media rate of the sender RTP mixer can use this message to limit the media rate of the sender
being forwarded by the mixer (without doing media transcoding) to fit being forwarded by the mixer (without doing media transcoding) to fit
the bottlenecks existing towards the other session participants. It the bottlenecks existing towards the other session participants.
is REQUIRED that this feedback message is supported. WebRTC senders WebRTC senders are REQUIRED to implement support for TMMBR messages,
are REQUIRED to implement support for TMMBR messages, and MUST follow and MUST follow bandwidth limitations set by a TMMBR message received
bandwidth limitations set by a TMMBR message received for their SSRC. for their SSRC. The sending of TMMBR requests is OPTIONAL.
The sending of TMMBR requests is OPTIONAL.
5.2. Header Extensions 5.2. Header Extensions
The RTP specification [RFC3550] provides the capability to include The RTP specification [RFC3550] provides the capability to include
RTP header extensions containing in-band data, but the format and RTP header extensions containing in-band data, but the format and
semantics of the extensions are poorly specified. The use of header semantics of the extensions are poorly specified. The use of header
extensions is OPTIONAL in the WebRTC context, but if they are used, extensions is OPTIONAL in the WebRTC context, but if they are used,
they MUST be formatted and signalled following the general mechanism they MUST be formatted and signalled following the general mechanism
for RTP header extensions defined in [RFC5285], since this gives for RTP header extensions defined in [RFC5285], since this gives
well-defined semantics to RTP header extensions. well-defined semantics to RTP header extensions.
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might include metadata that is additional to the usual RTP might include metadata that is additional to the usual RTP
information. information.
5.2.1. Rapid Synchronisation 5.2.1. Rapid Synchronisation
Many RTP sessions require synchronisation between audio, video, and Many RTP sessions require synchronisation between audio, video, and
other content. This synchronisation is performed by receivers, using other content. This synchronisation is performed by receivers, using
information contained in RTCP SR packets, as described in the RTP information contained in RTCP SR packets, as described in the RTP
specification [RFC3550]. This basic mechanism can be slow, however, specification [RFC3550]. This basic mechanism can be slow, however,
so it is RECOMMENDED that the rapid RTP synchronisation extensions so it is RECOMMENDED that the rapid RTP synchronisation extensions
described in [RFC6051] be implemented. The rapid synchronisation described in [RFC6051] be implemented in addition to RTCP SR-based
extensions use the general RTP header extension mechanism [RFC5285], synchronisation. The rapid synchronisation extensions use the
which requires signalling, but are otherwise backwards compatible. general RTP header extension mechanism [RFC5285], which requires
signalling, but are otherwise backwards compatible.
5.2.2. Client-to-Mixer Audio Level 5.2.2. Client-to-Mixer Audio Level
The Client to Mixer Audio Level extension [RFC6464] is an RTP header The Client to Mixer Audio Level extension [RFC6464] is an RTP header
extension used by a client to inform a mixer about the level of audio extension used by a client to inform a mixer about the level of audio
activity in the packet to which the header is attached. This enables activity in the packet to which the header is attached. This enables
a central node to make mixing or selection decisions without decoding a central node to make mixing or selection decisions without decoding
or detailed inspection of the payload, reducing the complexity in or detailed inspection of the payload, reducing the complexity in
some types of central RTP nodes. It can also save decoding resources some types of central RTP nodes. It can also save decoding resources
in receivers, which can choose to decode only the most relevant RTP in receivers, which can choose to decode only the most relevant RTP
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the relative activity level of each session participant, rather than the relative activity level of each session participant, rather than
just being included or not based on the CSRC field. This is a pure just being included or not based on the CSRC field. This is a pure
optimisations of non critical functions, and is hence OPTIONAL to optimisations of non critical functions, and is hence OPTIONAL to
implement. If it is implemented, it is REQUIRED that the header implement. If it is implemented, it is REQUIRED that the header
extensions are encrypted according to extensions are encrypted according to
[I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information [I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
contained in these header extensions can be considered sensitive. contained in these header extensions can be considered sensitive.
6. WebRTC Use of RTP: Improving Transport Robustness 6. WebRTC Use of RTP: Improving Transport Robustness
There are some tools that can make RTP flows robust against Packet There are tools that can make RTP media streams robust against packet
loss and reduce the impact on media quality. However, they all add loss and reduce the impact of loss on media quality. However, they
extra bits compared to a non-robust stream. These extra bits need to all add extra bits compared to a non-robust stream. The overhead of
be considered, and the aggregate bit-rate MUST be rate-controlled. these extra bits needs to be considered, and the aggregate bit-rate
Thus, improving robustness might require a lower base encoding MUST be rate controlled to avoid causing network congestion (see
quality, but has the potential to deliver that quality with fewer Section 7). As a result, improving robustness might require a lower
errors. The mechanisms described in the following sub-sections can base encoding quality, but has the potential to deliver that quality
be used to improve tolerance to packet loss. with fewer errors. The mechanisms described in the following sub-
sections can be used to improve tolerance to packet loss.
6.1. Negative Acknowledgements and RTP Retransmission 6.1. Negative Acknowledgements and RTP Retransmission
As a consequence of supporting the RTP/SAVPF profile, implementations As a consequence of supporting the RTP/SAVPF profile, implementations
will support negative acknowledgements (NACKs) for RTP data packets will support negative acknowledgements (NACKs) for RTP data packets
[RFC4585]. This feedback can be used to inform a sender of the loss [RFC4585]. This feedback can be used to inform a sender of the loss
of particular RTP packets, subject to the capacity limitations of the of particular RTP packets, subject to the capacity limitations of the
RTCP feedback channel. A sender can use this information to optimise RTCP feedback channel. A sender can use this information to optimise
the user experience by adapting the media encoding to compensate for the user experience by adapting the media encoding to compensate for
known lost packets, for example. known lost packets, for example.
skipping to change at page 16, line 13 skipping to change at page 16, line 43
sending NACK feedback, and the importance of the lost packet, to make sending NACK feedback, and the importance of the lost packet, to make
an informed decision on whether it is worth telling the sender about an informed decision on whether it is worth telling the sender about
a packet loss event. a packet loss event.
The RTP Retransmission Payload Format [RFC4588] offers the ability to The RTP Retransmission Payload Format [RFC4588] offers the ability to
retransmit lost packets based on NACK feedback. Retransmission needs retransmit lost packets based on NACK feedback. Retransmission needs
to be used with care in interactive real-time applications to ensure to be used with care in interactive real-time applications to ensure
that the retransmitted packet arrives in time to be useful, but can that the retransmitted packet arrives in time to be useful, but can
be effective in environments with relatively low network RTT (an RTP be effective in environments with relatively low network RTT (an RTP
sender can estimate the RTT to the receivers using the information in sender can estimate the RTT to the receivers using the information in
RTCP SR and RR packets). The use of retransmissions can also RTCP SR and RR packets, as described at the end of Section 6.4.1 of
increase the forward RTP bandwidth, and can potentially worsen the [RFC3550]). The use of retransmissions can also increase the forward
problem if the packet loss was caused by network congestion. We RTP bandwidth, and can potentially worsen the problem if the packet
note, however, that retransmission of an important lost packet to loss was caused by network congestion. We note, however, that
repair decoder state can have lower cost than sending a full intra retransmission of an important lost packet to repair decoder state
frame. It is not appropriate to blindly retransmit RTP packets in can have lower cost than sending a full intra frame. It is not
response to a NACK. The importance of lost packets and the appropriate to blindly retransmit RTP packets in response to a NACK.
likelihood of them arriving in time to be useful needs to be The importance of lost packets and the likelihood of them arriving in
considered before RTP retransmission is used. time to be useful needs to be considered before RTP retransmission is
used.
Receivers are REQUIRED to implement support for RTP retransmission Receivers are REQUIRED to implement support for RTP retransmission
packets [RFC4588]. Senders MAY send RTP retransmission packets in packets [RFC4588]. Senders MAY send RTP retransmission packets in
response to NACKs if the RTP retransmission payload format has been response to NACKs if the RTP retransmission payload format has been
negotiated for the session, and if the sender believes it is useful negotiated for the session, and if the sender believes it is useful
to send a retransmission of the packet(s) referenced in the NACK. An to send a retransmission of the packet(s) referenced in the NACK. An
RTP sender is not expected to retransmit every NACKed packet. RTP sender is not expected to retransmit every NACKed packet.
6.2. Forward Error Correction (FEC) 6.2. Forward Error Correction (FEC)
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this memo makes no recommendation on the choice of block-based FEC this memo makes no recommendation on the choice of block-based FEC
for WebRTC use. for WebRTC use.
7. WebRTC Use of RTP: Rate Control and Media Adaptation 7. WebRTC Use of RTP: Rate Control and Media Adaptation
WebRTC will be used in heterogeneous network environments using a WebRTC will be used in heterogeneous network environments using a
variety set of link technologies, including both wired and wireless variety set of link technologies, including both wired and wireless
links, to interconnect potentially large groups of users around the links, to interconnect potentially large groups of users around the
world. As a result, the network paths between users can have widely world. As a result, the network paths between users can have widely
varying one-way delays, available bit-rates, load levels, and traffic varying one-way delays, available bit-rates, load levels, and traffic
mixtures. Individual end-points can open one or more RTP sessions to mixtures. Individual end-points can send one or more RTP media
each participant in a WebRTC conference, and there can be several streams to each participant in a WebRTC conference, and there can be
participants. Each of these RTP sessions can contain different types several participants. Each of these RTP media streams can contain
of media, and the type of media, bit rate, and number of flows can be different types of media, and the type of media, bit rate, and number
highly asymmetric. Non-RTP traffic can share the network paths RTP of flows can be highly asymmetric. Non-RTP traffic can share the
flows. Since the network environment is not predictable or stable, network paths RTP flows. Since the network environment is not
WebRTC endpoints MUST ensure that the RTP traffic they generate can predictable or stable, WebRTC endpoints MUST ensure that the RTP
adapt to match changes in the available network capacity. traffic they generate can adapt to match changes in the available
network capacity.
The quality of experience for users of WebRTC implementation is very The quality of experience for users of WebRTC implementation is very
dependent on effective adaptation of the media to the limitations of dependent on effective adaptation of the media to the limitations of
the network. End-points have to be designed so they do not transmit the network. End-points have to be designed so they do not transmit
significantly more data than the network path can support, except for significantly more data than the network path can support, except for
very short time periods, otherwise high levels of network packet loss very short time periods, otherwise high levels of network packet loss
or delay spikes will occur, causing media quality degradation. The or delay spikes will occur, causing media quality degradation. The
limiting factor on the capacity of the network path might be the link limiting factor on the capacity of the network path might be the link
bandwidth, or it might be competition with other traffic on the link bandwidth, or it might be competition with other traffic on the link
(this can be non-WebRTC traffic, traffic due to other WebRTC flows, (this can be non-WebRTC traffic, traffic due to other WebRTC flows,
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media codecs provides upper- and lower-bounds on the supported bit- media codecs provides upper- and lower-bounds on the supported bit-
rates that the application can utilise to provide useful quality, and rates that the application can utilise to provide useful quality, and
the packetization choices that exist. In addition, the signalling the packetization choices that exist. In addition, the signalling
channel can establish maximum media bit-rate boundaries using the SDP channel can establish maximum media bit-rate boundaries using the SDP
"b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media "b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media
Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo). Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo).
The combination of media codec choice and signalled bandwidth limits The combination of media codec choice and signalled bandwidth limits
SHOULD be used to limit traffic based on known bandwidth limitations, SHOULD be used to limit traffic based on known bandwidth limitations,
for example the capacity of the edge links, to the extent possible. for example the capacity of the edge links, to the extent possible.
7.2. RTCP Extensions for Congestion Control 7.2. RTCP Limitations for Congestion Control
As described in Section 5.1.6, the Temporary Maximum Media Stream Bit
Rate (TMMBR) request is supported by WebRTC senders. This request
can be used by a media receiver to impose limitations on the media
sender based on the receiver's determined bit-rate limitations, to
provide a limited means of congestion control.
(tbd: What other RTP/RTCP extensions are needed?)
With proprietary congestion control algorithms issues can arise when
different algorithms and implementations interact in a communication
session. If the different implementations have made different
choices in regards to the type of adaptation, for example one sender
based, and one receiver based, then one could end up in situation
where one direction is dual controlled, when the other direction is
not controlled.
(tbd: How to ensure that both paths and sender and receiver based
solutions can interact)
7.3. RTCP Limitations for Congestion Control
Experience with the congestion control algorithms of TCP [RFC5681], Experience with the congestion control algorithms of TCP [RFC5681],
TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
that feedback on packet arrivals needs to be sent roughly once per that feedback on packet arrivals needs to be sent roughly once per
round trip time. We note that the real-time media traffic might not round trip time. We note that the real-time media traffic might not
have to adapt to changing path conditions as rapidly as needed for have to adapt to changing path conditions as rapidly as needed for
the elastic applications TCP was designed for, but frequent feedback the elastic applications TCP was designed for, but frequent feedback
is still needed to allow the congestion control algorithm to track is still needed to allow the congestion control algorithm to track
the path dynamics. the path dynamics.
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In group communication, the share of RTCP bandwidth needs to be In group communication, the share of RTCP bandwidth needs to be
shared by all group members, reducing the capacity and thus the shared by all group members, reducing the capacity and thus the
reporting frequency per node. reporting frequency per node.
Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP
bandwidth, split across two entities in a point-to-point session. An bandwidth, split across two entities in a point-to-point session. An
endpoint could thus send a report of 100 bytes about every 70ms or endpoint could thus send a report of 100 bytes about every 70ms or
for every other frame in a 30 fps video. for every other frame in a 30 fps video.
7.4. Congestion Control Interoperability With Legacy Systems 7.3. Congestion Control Interoperability and Legacy Systems
There are legacy implementations that do not implement RTCP, and There are legacy implementations that do not implement RTCP, and
hence do not provide any congestion feedback. Congestion control hence do not provide any congestion feedback. Congestion control
cannot be performed with these end-points. WebRTC implementations cannot be performed with these end-points. WebRTC implementations
that need to interwork with such end-points MUST limit their that need to interwork with such end-points MUST limit their
transmission to a low rate, equivalent to a VoIP call using a low transmission to a low rate, equivalent to a VoIP call using a low
bandwidth codec, that is unlikely to cause any significant bandwidth codec, that is unlikely to cause any significant
congestion. congestion.
When interworking with legacy implementations that support RTCP using When interworking with legacy implementations that support RTCP using
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important parameters for frequent reporting, such as the "trr-int" important parameters for frequent reporting, such as the "trr-int"
parameter, and the possibility that the end-point supports some parameter, and the possibility that the end-point supports some
useful feedback format for congestion control purpose such as TMMBR useful feedback format for congestion control purpose such as TMMBR
[RFC5104]. Implementations that have to interwork with such end- [RFC5104]. Implementations that have to interwork with such end-
points MUST ensure that they stay within the RTP circuit breaker points MUST ensure that they stay within the RTP circuit breaker
[I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the
congestion they can cause, but might find that they can achieve congestion they can cause, but might find that they can achieve
better congestion response depending on the amount of feedback that better congestion response depending on the amount of feedback that
is available. is available.
With proprietary congestion control algorithms issues can arise when
different algorithms and implementations interact in a communication
session. If the different implementations have made different
choices in regards to the type of adaptation, for example one sender
based, and one receiver based, then one could end up in situation
where one direction is dual controlled, when the other direction is
not controlled. This memo cannot mandate behaviour for proprietary
congestion control algorithms, but implementations that use such
algorithms ought to be aware of this issue, and try to ensure that
both effective congestion control is negotiated for media flowing in
both directions. If the IETF were to standardise both sender- and
receiver-based congestion control algorithms for WebRTC traffic in
the future, the issues of interoperability, control, and ensuring
that both directions of media flow are congestion controlled would
also need to be considered.
8. WebRTC Use of RTP: Performance Monitoring 8. WebRTC Use of RTP: Performance Monitoring
RTCP does contains a basic set of RTP flow monitoring metrics like As described in Section 4.1, implementations are REQUIRED to generate
packet loss and jitter. There are a number of extensions that could RTCP Sender Report (SR) and Reception Report (RR) packets relating to
be included in the set to be supported. However, in most cases which the RTP media streams they send and receive. These RTCP reports can
RTP monitoring that is needed depends on the application, which makes be used for performance monitoring purposes, since they include basic
it difficult to select which to include when the set of applications packet loss and jitter statistics.
is very large.
Exposing some metrics in the WebRTC API needs to be considered A large number of additional performance metrics are supported by the
allowing the application to gather the measurements of interest. RTCP Extended Reports (XR) framework [RFC3611]. It is not yet clear
However, security implications for the different data sets exposed what extended metrics are appropriate for use in the WebRTC context,
will need to be considered in this. so implementations are not expected to generate any RTCP XR packets.
However, implementations that can use detailed performance monitoring
data MAY generate RTCP XR packets as appropriate; the use of such
packets SHOULD be signalled in advance.
(tbd: If any RTCP XR metrics need to be added is still an open All WebRTC implementations MUST be prepared to receive RTP XR report
question, but possible to extend at a later stage) packets, whether or not they were signalled. There is no requirement
that the data contained in such reports be used, or exposed to the
Javascript application, however.
9. WebRTC Use of RTP: Future Extensions 9. WebRTC Use of RTP: Future Extensions
It is possible that the core set of RTP protocols and RTP extensions It is possible that the core set of RTP protocols and RTP extensions
specified in this memo will prove insufficient for the future needs specified in this memo will prove insufficient for the future needs
of WebRTC applications. In this case, future updates to this memo of WebRTC applications. In this case, future updates to this memo
MUST be made following the Guidelines for Writers of RTP Payload MUST be made following the Guidelines for Writers of RTP Payload
Format Specifications [RFC2736] and Guidelines for Extending the RTP Format Specifications [RFC2736] and Guidelines for Extending the RTP
Control Protocol [RFC5968], and SHOULD take into account any future Control Protocol [RFC5968], and SHOULD take into account any future
guidelines for extending RTP and related protocols that have been guidelines for extending RTP and related protocols that have been
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Authors of future extensions are urged to consider the wide range of Authors of future extensions are urged to consider the wide range of
environments in which RTP is used when recommending extensions, since environments in which RTP is used when recommending extensions, since
extensions that are applicable in some scenarios can be problematic extensions that are applicable in some scenarios can be problematic
in others. Where possible, the WebRTC framework will adopt RTP in others. Where possible, the WebRTC framework will adopt RTP
extensions that are of general utility, to enable easy implementation extensions that are of general utility, to enable easy implementation
of a gateway to other applications using RTP, rather than adopt of a gateway to other applications using RTP, rather than adopt
mechanisms that are narrowly targeted at specific WebRTC use cases. mechanisms that are narrowly targeted at specific WebRTC use cases.
10. Signalling Considerations 10. Signalling Considerations
RTP is built with the assumption of an external signalling channel RTP is built with the assumption that an external signalling channel
that can be used to configure the RTP sessions and their features. exists, and can be used to configure RTP sessions and their features.
The basic configuration of an RTP session consists of the following The basic configuration of an RTP session consists of the following
parameters: parameters:
RTP Profile: The name of the RTP profile to be used in session. The RTP Profile: The name of the RTP profile to be used in session. The
RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
on basic level, as can their secure variants RTP/SAVP [RFC3711] on basic level, as can their secure variants RTP/SAVP [RFC3711]
and RTP/SAVPF [RFC5124]. The secure variants of the profiles do and RTP/SAVPF [RFC5124]. The secure variants of the profiles do
not directly interoperate with the non-secure variants, due to the not directly interoperate with the non-secure variants, due to the
presence of additional header fields in addition to any presence of additional header fields for authentication in SRTP
cryptographic transformation of the packet content. As WebRTC packets and cryptographic transformation of the payload. WebRTC
requires the usage of the RTP/SAVPF profile this can be inferred requires the use of the RTP/SAVPF profile, and this MUST be
as there is only a single profile, but in SDP this is still signalled if SDP is used. Interworking functions might transform
information that has to be signalled. Interworking functions this into the RTP/SAVP profile for a legacy use case, by
might transform this into RTP/SAVP for a legacy use case by indicating to the WebRTC end-point that the RTP/SAVPF is used, and
indicating to the WebRTC end-point a RTP/SAVPF end-point and limiting the usage of the "a=rtcp:" attribute to indicate a trr-
limiting the usage of the a=rtcp attribute to indicate a trr-int int value of 4 seconds.
value of 4 seconds.
Transport Information: Source and destination IP address(s) and Transport Information: Source and destination IP address(s) and
ports for RTP and RTCP MUST be signalled for each RTP session. In ports for RTP and RTCP MUST be signalled for each RTP session. In
WebRTC these transport addresses will be provided by ICE that WebRTC these transport addresses will be provided by ICE that
signals candidates and arrives at nominated candidate address signals candidates and arrives at nominated candidate address
pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such
that a single port is used for RTP and RTCP flows, this MUST be that a single port is used for RTP and RTCP flows, this MUST be
signalled (see Section 4.5). If several RTP sessions are to be signalled (see Section 4.5). If several RTP sessions are to be
multiplexed onto a single transport layer flow, this MUST also be multiplexed onto a single transport layer flow, this MUST also be
signalled (see Section 4.4). signalled (see Section 4.4).
RTP Payload Types, media formats, and media format RTP Payload Types, media formats, and format parameters: The mapping
parameters: The mapping between media type names (and hence the RTP between media type names (and hence the RTP payload formats to be
payload formats to be used) and the RTP payload type numbers MUST used), and the RTP payload type numbers MUST be signalled. Each
be signalled. Each media type MAY also have a number of media media type MAY also have a number of media type parameters that
type parameters that MUST also be signalled to configure the codec MUST also be signalled to configure the codec and RTP payload
and RTP payload format (the "a=fmtp:" line from SDP). format (the "a=fmtp:" line from SDP). Section 4.3 of this memo
discusses requirements for uniqueness of payload types.
RTP Extensions: The RTP extensions to be used SHOULD be agreed upon, RTP Extensions: The RTP extensions to be used SHOULD be agreed upon,
including any parameters for each respective extension. At the including any parameters for each respective extension. At the
very least, this will help avoiding using bandwidth for features very least, this will help avoiding using bandwidth for features
that the other end-point will ignore. But for certain mechanisms that the other end-point will ignore. But for certain mechanisms
there is requirement for this to happen as interoperability there is requirement for this to happen as interoperability
failure otherwise happens. failure otherwise happens.
RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the
end-points will be necessary. This SHALL be done as described in end-points will be necessary. This SHALL be done as described in
"Session Description Protocol (SDP) Bandwidth Modifiers for RTP "Session Description Protocol (SDP) Bandwidth Modifiers for RTP
Control Protocol (RTCP) Bandwidth" [RFC3556], or something Control Protocol (RTCP) Bandwidth" [RFC3556], or something
semantically equivalent. This also ensures that the end-points semantically equivalent. This also ensures that the end-points
have a common view of the RTCP bandwidth, this is important as too have a common view of the RTCP bandwidth, this is important as too
different view of the bandwidths can lead to failure to different view of the bandwidths can lead to failure to
interoperate. interoperate.
These parameters are often expressed in SDP messages conveyed within These parameters are often expressed in SDP messages conveyed within
an offer/answer exchange. RTP does not depend on SDP or on the an offer/answer exchange. RTP does not depend on SDP or on the offer
offer/answer model, but does require all the necessary parameters to /answer model, but does require all the necessary parameters to be
be agreed upon, and provided to the RTP implementation. We note that agreed upon, and provided to the RTP implementation. We note that in
in the WebRTC context it will depend on the signalling model and API the WebRTC context it will depend on the signalling model and API how
how these parameters need to be configured but they will be need to these parameters need to be configured but they will be need to
either set in the API or explicitly signalled between the peers. either set in the API or explicitly signalled between the peers.
11. WebRTC API Considerations 11. WebRTC API Considerations
The WebRTC API and its media function have the concept of a WebRTC The WebRTC API and its media function have the concept of a WebRTC
MediaStream that consists of zero or more tracks. A track is an MediaStream that consists of zero or more tracks. A track is an
individual stream of media from any type of media source like a individual stream of media from any type of media source like a
microphone or a camera, but also conceptual sources, like a audio mix microphone or a camera, but also conceptual sources, like a audio mix
or a video composition, are possible. The tracks within a WebRTC or a video composition, are possible. The tracks within a WebRTC
MediaStream are expected to be synchronized. MediaStream are expected to be synchronized.
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As a result, a WebRTC MediaStream is a collection of SSRCs carrying As a result, a WebRTC MediaStream is a collection of SSRCs carrying
the different media included in the synchronised aggregate. the different media included in the synchronised aggregate.
Therefore, also the synchronization state associated with the Therefore, also the synchronization state associated with the
included SSRCs are part of concept. It is important to consider that included SSRCs are part of concept. It is important to consider that
there can be multiple different WebRTC MediaStreams containing a there can be multiple different WebRTC MediaStreams containing a
given Track (SSRC). To avoid unnecessary duplication of media at the given Track (SSRC). To avoid unnecessary duplication of media at the
transport level in such cases, a need arises for a binding defining transport level in such cases, a need arises for a binding defining
which WebRTC MediaStreams a given SSRC is associated with at the which WebRTC MediaStreams a given SSRC is associated with at the
signalling level. signalling level.
The API also needs to be capable of handling when new SSRCs are
received but not previously signalled by signalling in some fashion.
Note, that not all SSRCs carries media directly associated with a
media source, instead they can be repair or redundancy information
for one or a set of SSRCs.
A proposal for how the binding between WebRTC MediaStreams and SSRC A proposal for how the binding between WebRTC MediaStreams and SSRC
can be done is specified in "Cross Session Stream Identification in can be done is specified in "Cross Session Stream Identification in
the Session Description Protocol" [I-D.alvestrand-rtcweb-msid]. the Session Description Protocol" [I-D.alvestrand-rtcweb-msid].
(tbd: This text needs to be improved and achieved consensus on. (tbd: This text needs to be improved and achieved consensus on.
Interim meeting in June 2012 shows large differences in opinions.) Interim meeting in June 2012 shows large differences in opinions.)
(tbd: It is an open question whether these considerations are best (tbd: It is an open question whether these considerations are best
discussed in this draft, in the W3C WebRTC API spec, or elsewhere. discussed in this draft, in the W3C WebRTC API spec, or elsewhere.
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does not have multiple media sources of the same media type it has to does not have multiple media sources of the same media type it has to
support transmission using multiple SSRCs concurrently in the same support transmission using multiple SSRCs concurrently in the same
RTP session. This is due to the requirement on an WebRTC end-point RTP session. This is due to the requirement on an WebRTC end-point
to support multiple media types in one RTP session. For example, one to support multiple media types in one RTP session. For example, one
audio and one video source can result in the end-point sending with audio and one video source can result in the end-point sending with
two different SSRCs in the same RTP session. As multi-party two different SSRCs in the same RTP session. As multi-party
conferences are supported, as discussed below in Section 12.3, a conferences are supported, as discussed below in Section 12.3, a
WebRTC end-point will need to be capable of receiving, decoding and WebRTC end-point will need to be capable of receiving, decoding and
play out multiple RTP media streams of the same type concurrently. play out multiple RTP media streams of the same type concurrently.
tbd: there needs to be a way of indicating how RTP stream relate when
there are multiple sources, possibly with simulcast or layered
coding, and different types of mixer or other middlebox. It is
possible that the various BUNDLE/Plan-X proposals will solve this,
but it might also need RTP-level stream identification. To be
resolved once the outcome of the BUNDLE and plan-X discussions is
known.
tbd: Are any mechanism needed to signal limitations in the number of tbd: Are any mechanism needed to signal limitations in the number of
active SSRC that an end-point can handle? active SSRC that an end-point can handle?
12.3. Multiparty 12.3. Multiparty
There are numerous situations and clear use cases for WebRTC There are numerous situations and clear use cases for WebRTC
supporting RTP sessions supporting multi-party. This can be realized supporting RTP sessions supporting multi-party. This can be realized
in a number of ways using a number of different implementation in a number of ways using a number of different implementation
strategies. In the following, the focus is on the different set of strategies. In the following, the focus is on the different set of
WebRTC end-point requirements that arise from different sets of WebRTC end-point requirements that arise from different sets of
multi-party topologies. multi-party topologies.
The multi-unicast mesh (Figure 1)-based multi-party topology The multi-unicast mesh (Figure 1)-based multi-party topology
discussed above provides a non-centralized solution but can incur a discussed above provides a non-centralized solution but can incur a
heavy tax on the end-points' outgoing paths. It can also consume heavy tax on the end-points' outgoing paths. It can also consume
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particular RTP media flow need to be marked. RTCP compound packets particular RTP media flow need to be marked. RTCP compound packets
with Sender Reports (SR), ought to be marked with the same priority with Sender Reports (SR), ought to be marked with the same priority
as the RTP media flow itself, so the RTCP-based round-trip time (RTT) as the RTP media flow itself, so the RTCP-based round-trip time (RTT)
measurements are done using the same flow priority as the media flow measurements are done using the same flow priority as the media flow
experiences. RTCP compound packets containing RR packet ought to be experiences. RTCP compound packets containing RR packet ought to be
sent with the priority used by the majority of the RTP media flows sent with the priority used by the majority of the RTP media flows
reported on. RTCP packets containing time-critical feedback packets reported on. RTCP packets containing time-critical feedback packets
can use higher priority to improve the timeliness and likelihood of can use higher priority to improve the timeliness and likelihood of
delivery of such feedback. delivery of such feedback.
13. Open Issues 13. Security Considerations
This section contains a summary of the open issues or to be done
things noted in the document:
1. Need to add references to the RTP payload format for the Video
Codec chosen in Section 4.3.
2. The methods and solutions for RTP multiplexing over a single
transport is not yet finalized in Section 4.4.
3. RTP congestion control algorithms will probably require some
feedback information to be conveyed in RTCP. Are the tools that
are mandated by this memo sufficient, or do we need additional
information Section 7.2?
4. RTP congestion control could be implementing using either a
sender-based algorithm or a receiver-based algorithm. To ensure
interoperability, does this memo need to mandate which end is in
charge of congestion control for a path Section 7.2?
5. Still open if any RTCP XR performance metrics are needed, as
discussed in Section 8.
6. The API mapping to RTP level concepts has to be agreed and
documented in Section 11.
7. An open question if any requirements are needed to agree and
limit the number of simultaneously used media sources (SSRCs)
within an RTP session. See Section 12.2.
8. The method for achieving simulcast of a media source has to be
decided as discussed in Section 12.8.
9. Possible documentation of what support for differentiated
treatment that are needed on RTP level as the API and the
network level specification matures as discussed in
Section 12.9.
10. Editing of Appendix A to remove redundancy between this and the
update of RTP Topologies
[I-D.westerlund-avtcore-rtp-topologies-update].
14. IANA Considerations
This memo makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as
an RFC.
15. Security Considerations
The overall security architecture for WebRTC is described in The overall security architecture for WebRTC is described in
[I-D.ietf-rtcweb-security-arch], and security considerations for the [I-D.ietf-rtcweb-security-arch], and security considerations for the
WebRTC framework are described in [I-D.ietf-rtcweb-security]. These WebRTC framework are described in [I-D.ietf-rtcweb-security]. These
considerations apply to this memo also. considerations apply to this memo also.
The security considerations of the RTP specification, the RTP/SAVPF The security considerations of the RTP specification, the RTP/SAVPF
profile, and the various RTP/RTCP extensions and RTP payload formats profile, and the various RTP/RTCP extensions and RTP payload formats
that form the complete protocol suite described in this memo apply. that form the complete protocol suite described in this memo apply.
We do not believe there are any new security considerations resulting We do not believe there are any new security considerations resulting
skipping to change at page 33, line 13 skipping to change at page 33, line 9
provides guidelines for generation of untraceable CNAME values that provides guidelines for generation of untraceable CNAME values that
alleviate this risk. alleviate this risk.
The guidelines in [RFC6562] apply when using variable bit rate (VBR) The guidelines in [RFC6562] apply when using variable bit rate (VBR)
audio codecs such as Opus (see Section 4.3 for discussion of mandated audio codecs such as Opus (see Section 4.3 for discussion of mandated
audio codecs). These guidelines in [RFC6562] also apply, but are of audio codecs). These guidelines in [RFC6562] also apply, but are of
lesser importance, when using the client-to-mixer audio level header lesser importance, when using the client-to-mixer audio level header
extensions (Section 5.2.2) or the mixer-to-client audio level header extensions (Section 5.2.2) or the mixer-to-client audio level header
extensions (Section 5.2.3). extensions (Section 5.2.3).
14. IANA Considerations
This memo makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as
an RFC.
15. Open Issues
This section contains a summary of the open issues or to be done
things noted in the document:
1. tbd: The API mapping to RTP level concepts has to be agreed and
documented in Section 11.
2. tbd: An open question if any requirements are needed to agree and
limit the number of simultaneously used media sources (SSRCs)
within an RTP session. See Section 12.2 and Section 4.1.
3. tbd: The method for achieving simulcast of a media source has to
be decided as discussed in Section 12.8.
4. tbd: Possible documentation of what support for differentiated
treatment that are needed on RTP level as the API and the network
level specification matures as discussed in Section 12.9.
5. tbd: Editing of Appendix A to remove redundancy between this and
the update of RTP Topologies
[I-D.westerlund-avtcore-rtp-topologies-update].
16. Acknowledgements 16. Acknowledgements
The authors would like to thank Harald Alvestrand, Cary Bran, Charles The authors would like to thank Harald Alvestrand, Cary Bran, Charles
Eckel and Cullen Jennings for valuable feedback. Eckel, Cullen Jennings, Bernard Aboba, and the other members of the
IETF RTCWEB working group for their valuable feedback.
17. References 17. References
17.1. Normative References 17.1. Normative References
[I-D.ietf-avtcore-6222bis] [I-D.ietf-avtcore-6222bis]
Rescorla, E. and A. Begen, "Guidelines for Choosing RTP Begen, A., Perkins, C., Wing, D., and E. Rescorla,
Control Protocol (RTCP) Canonical Names (CNAMEs)", "Guidelines for Choosing RTP Control Protocol (RTCP)
draft-ietf-avtcore-6222bis-00 (work in progress), Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
December 2012. (work in progress), July 2013.
[I-D.ietf-avtcore-avp-codecs] [I-D.ietf-avtcore-avp-codecs]
Terriberry, T., "Update to Recommended Codecs for the AVP Terriberry, T., "Update to Remove DVI4 from the
RTP Profile", draft-ietf-avtcore-avp-codecs-00 (work in Recommended Codecs for the RTP Profile for Audio and Video
progress), January 2013. Conferences with Minimal Control (RTP/AVP)", draft-ietf-
avtcore-avp-codecs-03 (work in progress), July 2013.
[I-D.ietf-avtcore-multi-media-rtp-session] [I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Multiple Westerlund, M., Perkins, C., and J. Lennox, "Sending
Media Types in an RTP Session", Multiple Types of Media in a Single RTP Session", draft-
draft-ietf-avtcore-multi-media-rtp-session-01 (work in ietf-avtcore-multi-media-rtp-session-03 (work in
progress), October 2012. progress), July 2013.
[I-D.ietf-avtcore-rtp-circuit-breakers] [I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "Multimedia Congestion Control: Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", Circuit Breakers for Unicast RTP Sessions", draft-ietf-
draft-ietf-avtcore-rtp-circuit-breakers-02 (work in avtcore-rtp-circuit-breakers-02 (work in progress),
progress), February 2013. February 2013.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback ",
draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work
in progress), July 2013.
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-01 (work in progress),
July 2013.
[I-D.ietf-avtcore-srtp-encrypted-header-ext] [I-D.ietf-avtcore-srtp-encrypted-header-ext]
Lennox, J., "Encryption of Header Extensions in the Secure Lennox, J., "Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)", Real-Time Transport Protocol (SRTP)", draft-ietf-avtcore-
draft-ietf-avtcore-srtp-encrypted-header-ext-05 (work in srtp-encrypted-header-ext-05 (work in progress), February
progress), February 2013. 2013.
[I-D.ietf-avtext-multiple-clock-rates] [I-D.ietf-avtext-multiple-clock-rates]
Petit-Huguenin, M. and G. Zorn, "Support for Multiple Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", Clock Rates in an RTP Session", draft-ietf-avtext-
draft-ietf-avtext-multiple-clock-rates-08 (work in multiple-clock-rates-09 (work in progress), April 2013.
progress), November 2012.
[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,
"Multiplexing Negotiation Using Session Description "Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
draft-ietf-mmusic-sdp-bundle-negotiation-03 (work in bundle-negotiation-04 (work in progress), June 2013.
progress), February 2013.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-01 (work in
progress), November 2012.
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower- Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-06 (work based Applications", draft-ietf-rtcweb-overview-06 (work
in progress), February 2013. in progress), February 2013.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web",
draft-ietf-rtcweb-security-04 (work in progress),
January 2013.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "RTCWEB Security Architecture", Rescorla, E., "WebRTC Security Architecture", draft-ietf-
draft-ietf-rtcweb-security-arch-06 (work in progress), rtcweb-security-arch-07 (work in progress), July 2013.
January 2013.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-05 (work in progress), July 2013.
[I-D.westerlund-avtcore-transport-multiplexing] [I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
Single Lower-Layer Transport", Single Lower-Layer Transport", draft-westerlund-avtcore-
draft-westerlund-avtcore-transport-multiplexing-04 (work transport-multiplexing-05 (work in progress), February
in progress), October 2012. 2013.
[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, March 1997.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736, Payload Format Specifications", BCP 36, RFC 2736, December
December 1999. 1999.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003. July 2003.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
RFC 3556, July 2003. 3556, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004. RFC 3711, March 2004.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
July 2006. 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006. July 2006.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, July 2007. BCP 131, RFC 4961, July 2007.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile "Codec Control Messages in the RTP Audio-Visual Profile
skipping to change at page 36, line 17 skipping to change at page 37, line 6
[RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time [RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to- Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, December 2011. Mixer Audio Level Indication", RFC 6464, December 2011.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to- Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 2011. Client Audio Level Indication", RFC 6465, December 2011.
[RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP", RFC 6562, Variable Bit Rate Audio with Secure RTP", RFC 6562, March
March 2012. 2012.
17.2. Informative References 17.2. Informative References
[I-D.alvestrand-rtcweb-msid] [I-D.alvestrand-rtcweb-msid]
Alvestrand, H., "Cross Session Stream Identification in Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol", the Session Description Protocol", draft-alvestrand-
draft-alvestrand-rtcweb-msid-02 (work in progress), rtcweb-msid-02 (work in progress), May 2012.
May 2012.
[I-D.ietf-avt-srtp-ekt] [I-D.ietf-avt-srtp-ekt]
Wing, D., McGrew, D., and K. Fischer, "Encrypted Key Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03 Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
(work in progress), October 2011. (work in progress), October 2011.
[I-D.ietf-rtcweb-qos] [I-D.ietf-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS", other packet markings for RTCWeb QoS", draft-ietf-rtcweb-
draft-ietf-rtcweb-qos-00 (work in progress), October 2012. qos-00 (work in progress), October 2012.
[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", Time Communication Use-cases and Requirements", draft-
draft-ietf-rtcweb-use-cases-and-requirements-10 (work in ietf-rtcweb-use-cases-and-requirements-11 (work in
progress), December 2012. progress), June 2013.
[I-D.jesup-rtp-congestion-reqs] [I-D.jesup-rtp-congestion-reqs]
Jesup, R. and H. Alvestrand, "Congestion Control Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media", Requirements For Real Time Media", draft-jesup-rtp-
draft-jesup-rtp-congestion-reqs-00 (work in progress), congestion-reqs-00 (work in progress), March 2012.
March 2012.
[I-D.westerlund-avtcore-multiplex-architecture] [I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., Perkins, C., and H. Westerlund, M., Perkins, C., and H. Alvestrand,
Alvestrand, "Guidelines for using the Multiplexing "Guidelines for using the Multiplexing Features of RTP",
Features of RTP", draft-westerlund-avtcore-multiplex-architecture-03 (work
draft-westerlund-avtcore-multiplex-architecture-02 (work in progress), February 2013.
in progress), July 2012.
[I-D.westerlund-avtcore-rtp-topologies-update] [I-D.westerlund-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", Westerlund, M. and S. Wenger, "RTP Topologies", draft-
draft-westerlund-avtcore-rtp-topologies-update-02 (work in westerlund-avtcore-rtp-topologies-update-02 (work in
progress), February 2013. progress), February 2013.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006. Congestion Control", RFC 4341, March 2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
March 2006. March 2006.
[RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient
Stream Loss-Tolerant Authentication (TESLA) in the Secure Stream Loss-Tolerant Authentication (TESLA) in the Secure
Real-time Transport Protocol (SRTP)", RFC 4383, Real-time Transport Protocol (SRTP)", RFC 4383, February
February 2006. 2006.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control [RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828, (TFRC): The Small-Packet (SP) Variant", RFC 4828, April
April 2007. 2007.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", Friendly Rate Control (TFRC): Protocol Specification", RFC
RFC 5348, September 2008. 5348, September 2008.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009. (SDP)", RFC 5576, June 2009.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009. Control", RFC 5681, September 2009.
[RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP
Control Protocol (RTCP)", RFC 5968, September 2010. Control Protocol (RTCP)", RFC 5968, September 2010.
skipping to change at page 60, line 47 skipping to change at page 59, line 17
such functionality, which on the API level is extremely simplistic to such functionality, which on the API level is extremely simplistic to
perform. perform.
+---+ +---+ +---+ +---+ +---+ +---+
| A |--->| B |--->| C | | A |--->| B |--->| C |
+---+ +---+ +---+ +---+ +---+ +---+
Figure 19: MediaStream Forwarding Figure 19: MediaStream Forwarding
There exist two main approaches to how B forwards the media from A to There exist two main approaches to how B forwards the media from A to
C. The first one is to simply relay the RTP media stream. The second C. The first one is to simply relay the RTP media stream. The
one is for B to act as a transcoder. Lets consider both approaches. second one is for B to act as a transcoder. Lets consider both
approaches.
A relay approach will result in that the WebRTC end-points will have A relay approach will result in that the WebRTC end-points will have
to have the same capabilities as being discussed in Relay to have the same capabilities as being discussed in Relay
(Appendix A.4.3). Thus A will see an RTP session that is extended (Appendix A.4.3). Thus A will see an RTP session that is extended
beyond the PeerConnection and see two different receiving end-points beyond the PeerConnection and see two different receiving end-points
with different path characteristics (B and C). Thus A's congestion with different path characteristics (B and C). Thus A's congestion
control needs to be capable of handling this. The security solution control needs to be capable of handling this. The security solution
can either support mechanism that allows A to inform C about the key can either support mechanism that allows A to inform C about the key
A is using despite B and C having agreed on another set of keys. A is using despite B and C having agreed on another set of keys.
Alternatively B will decrypt and then re-encrypt using a new key. Alternatively B will decrypt and then re-encrypt using a new key.
 End of changes. 92 change blocks. 
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