draft-ietf-rtcweb-rtp-usage-03.txt   draft-ietf-rtcweb-rtp-usage-04.txt 
Network Working Group C. Perkins Network Working Group C. Perkins
Internet-Draft University of Glasgow Internet-Draft University of Glasgow
Intended status: Standards Track M. Westerlund Intended status: Standards Track M. Westerlund
Expires: December 6, 2012 Ericsson Expires: January 17, 2013 Ericsson
J. Ott J. Ott
Aalto University Aalto University
June 4, 2012 July 16, 2012
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-03 draft-ietf-rtcweb-rtp-usage-04
Abstract Abstract
The Web Real-Time Communication (WebRTC) framework provides support The Web Real-Time Communication (WebRTC) framework provides support
for direct interactive rich communication using audio, video, text, for direct interactive rich communication using audio, video, text,
collaboration, games, etc. between two peers' web-browsers. This collaboration, games, etc. between two peers' web-browsers. This
memo describes the media transport aspects of the WebRTC framework. memo describes the media transport aspects of the WebRTC framework.
It specifies how the Real-time Transport Protocol (RTP) is used in It specifies how the Real-time Transport Protocol (RTP) is used in
the WebRTC context, and gives requirements for which RTP features, the WebRTC context, and gives requirements for which RTP features,
profiles, and extensions need to be supported. profiles, and extensions need to be supported.
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on December 6, 2012. This Internet-Draft will expire on January 17, 2013.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
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 . . . . . . . . . . . . . . 6
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . . 6 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . . 6
4.2. Choice of RTP Profile . . . . . . . . . . . . . . . . . . 7 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8
4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 8 4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 9
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 8 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 9 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 9 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 11
4.8. Generation of the RTCP Canonical Name (CNAME) . . . . . . 10 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 10 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11
5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 10 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12
5.1.1. Full Intra Request . . . . . . . . . . . . . . . . . . 11 5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12
5.1.2. Picture Loss Indication . . . . . . . . . . . . . . . 11 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . . 13
5.1.3. Slice Loss Indication . . . . . . . . . . . . . . . . 11 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13
5.1.4. Reference Picture Selection Indication . . . . . . . . 12 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13
5.1.5. Temporary Maximum Media Stream Bit Rate Request . . . 12 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 14
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 12 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 14
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 12 5.1.6. Temporary Maximum Media Stream Bit Rate Request . . . 14
5.2.2. Client to Mixer Audio Level . . . . . . . . . . . . . 13 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14
5.2.3. Mixer to Client Audio Level . . . . . . . . . . . . . 13 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 15
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 13 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 15
6.1. Retransmission . . . . . . . . . . . . . . . . . . . . . . 14 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 15 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 16
6.2.1. Basic Redundancy . . . . . . . . . . . . . . . . . . . 15 6.1. Negative Acknowledgements and RTP Retransmission . . . . . 16
6.2.2. Block Based FEC . . . . . . . . . . . . . . . . . . . 16 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 17
6.2.3. Recommendations for 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. Congestion Control Requirements . . . . . . . . . . . . . 19 7.1. Congestion Control Requirements . . . . . . . . . . . . . 18
7.2. Rate Control Boundary Conditions . . . . . . . . . . . . . 19 7.2. Rate Control Boundary Conditions . . . . . . . . . . . . . 19
7.3. RTCP Limiations . . . . . . . . . . . . . . . . . . . . . 19 7.3. RTCP Limitations for Congestion Control . . . . . . . . . 19
7.4. Legacy Interop Limitations . . . . . . . . . . . . . . . . 20 7.4. Congestion Control Interoperability With Legacy Systems . 20
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 21 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 20
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 21 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 21
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 23 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 22
11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 23 11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 22
12. RTP Implementation Considerations . . . . . . . . . . . . . . 23 12. RTP Implementation Considerations . . . . . . . . . . . . . . 23
12.1. RTP Sessions and PeerConnection . . . . . . . . . . . . . 24 12.1. RTP Sessions and PeerConnection . . . . . . . . . . . . . 23
12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 25 12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 25
12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 25 12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 25
12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 27 12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 26
12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 28 12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 27
12.6. Media Synchronization . . . . . . . . . . . . . . . . . . 29 12.6. Media Synchronization . . . . . . . . . . . . . . . . . . 28
12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . . 29 12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . . 28
12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 30 12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 29
12.9. Differentiated Treatment of Flows . . . . . . . . . . . . 30 12.9. Differentiated Treatment of Flows . . . . . . . . . . . . 29
13. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31 13. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31
14. Security Considerations . . . . . . . . . . . . . . . . . . . 32 14. Security Considerations . . . . . . . . . . . . . . . . . . . 31
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 32 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 31
16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32 16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32
16.1. Normative References . . . . . . . . . . . . . . . . . . . 32 16.1. Normative References . . . . . . . . . . . . . . . . . . . 32
16.2. Informative References . . . . . . . . . . . . . . . . . . 35 16.2. Informative References . . . . . . . . . . . . . . . . . . 34
Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 37 Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 36
A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 37 A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 36
A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 40 A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 39
A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 43 A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 42
A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 43 A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 42
A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 46 A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 45
A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . . 49 A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . . 48
A.4. Translator Based . . . . . . . . . . . . . . . . . . . . . 52 A.4. Translator Based . . . . . . . . . . . . . . . . . . . . . 51
A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . . 52 A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . . 51
A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 53 A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 52
A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 55 A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 54
A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . . 59 A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . . 58
A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 60 A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 59
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61 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.
The Web Real-Time communication (WebRTC) framework is a new protocol The Web Real-Time communication (WebRTC) framework provides the
framework that provides support for direct, interactive, real-time protocol building blocks to support direct, interactive, real-time
communication using audio, video, collaboration, games, etc., between communication using audio, video, collaboration, games, etc., between
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 must be implemented by all WebRTC-aware browsers, RTP features that are to be implemented by all WebRTC-aware end-
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 requirement terminology. Requirements for core RTP
protocols are described in Section 4 and recommended RTP extensions protocols are described in Section 4 and recommended RTP extensions
are described in Section 5. Section 6 outlines mechanisms that can are described in Section 5. Section 6 outlines mechanisms that can
increase robustness to network problems, while Section 7 describes increase robustness to network problems, while Section 7 describes
the required congestion control and rate adaptation mechanisms. The the required congestion control and rate adaptation mechanisms. The
discussion of required RTP mechanisms concludes in Section 8 with a discussion of mandated RTP mechanisms concludes in Section 8 with a
review of performance monitoring and network management tools that review of performance monitoring and network management tools that
can be used in the WebRTC context. Section 9 gives some guidelines can be used in the WebRTC context. Section 9 gives some guidelines
for future incorporation of other RTP and RTP Control Protocol (RTCP) for future incorporation of other RTP and RTP Control Protocol (RTCP)
extensions into this framework. Section 10 describes requirements extensions into this framework. Section 10 describes requirements
placed on the signalling channel. Section 11 discusses the placed on the signalling channel. Section 11 discusses the
relationship between features of the RTP framework and the WebRTC relationship between features of the RTP framework and the WebRTC
application programming interface (API), and Section 12 discusses RTP application programming interface (API), and Section 12 discusses RTP
implementation considerations. This memo concludes with an appendix implementation considerations. This memo concludes with an appendix
discussing several different RTP Topologies, and how they affect the discussing several different RTP Topologies, and how they affect the
RTP session(s) and various implementation details of possible RTP session(s) and various implementation details of possible
realization of central nodes. realization of central nodes.
2. Rationale 2. Rationale
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
features should be supported by new implementations? The development extensions are to be supported by new implementations. The
of the WebRTC framework provides an opportunity for us to review the development of the WebRTC framework provides an opportunity for us to
available RTP features and extensions, and to define a common review the available RTP features and extensions, and to define a
baseline feature set for all WebRTC implementations of RTP. This common baseline feature set for all WebRTC implementations of RTP.
builds on the past 15 years development of RTP to mandate the use of This builds on the past 15 years development of RTP to mandate the
extensions that have shown widespread utility, while still remaining use of extensions that have shown widespread utility, while still
compatible with the wide installed base of RTP implementations where remaining compatible with the wide installed base of RTP
possible. implementations where possible.
RTP and RTCP extensions not discussed in this document can still be
implemented by a WebRTC end-point, but they are considered optional,
are not required for interoperability, and do not provide features
needed to address the WebRTC use cases and requirements
[I-D.ietf-rtcweb-use-cases-and-requirements].
While the baseline set of RTP features and extensions defined in this While the baseline set of RTP features and extensions defined in this
memo is targetted at the requirements of the WebRTC framework, it is memo is targeted at the requirements of the WebRTC framework, it is
expected to be broadly useful for other conferencing-related uses of expected to be broadly useful for other conferencing-related uses of
RTP. In particular, it is likely that this set of RTP features and RTP. In particular, it is likely that this set of RTP features and
extensions will be apppropriate for other desktop or mobile video extensions will be appropriate for other desktop or mobile video
conferencing systems, or for room-based high-quality telepresence conferencing systems, or for room-based high-quality telepresence
applications. applications.
3. Terminology 3. Terminology
This memo specifies various requirements levels for implementation or This memo specifies various requirements levels for implementation or
use of RTP features and extensions. When we describe the importance use of RTP features and extensions. When we describe the importance
of RTP extensions, or the need for implementation support, we use the of RTP extensions, or the need for implementation support, we use the
following requirement levels to specify the importance of the feature following requirement levels to specify the importance of the feature
in the WebRTC framework: in the WebRTC framework:
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that it enhances the product while another vendor may omit the that it enhances the product while another vendor may omit the
same item. An implementation which does not include a particular same item. An implementation which does not include a particular
option MUST be prepared to interoperate with another option MUST be prepared to interoperate with another
implementation which does include the option, though perhaps with implementation which does include the option, though perhaps with
reduced functionality. In the same vein an implementation which reduced functionality. In the same vein an implementation which
does include a particular option MUST be prepared to interoperate does include a particular option MUST be prepared to interoperate
with another implementation which does not include the option with another implementation which does not include the option
(except, of course, for the feature the option provides.) (except, of course, for the feature the option provides.)
These key words are used in a manner consistent with their definition These key words are used in a manner consistent with their definition
in [RFC2119]. in [RFC2119]. The above interpretation of these key words applies
only when written in ALL CAPS. Lower- or mixed-case uses of these
key words are not to be interpreted as carrying special significance
in this memo.
We define the following terms:
RTP Media Stream: A sequence of RTP packets, and associated RTCP
packets, using a single synchronisation source (SSRC) that
together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session.
RTP Session: As defined by [RFC3550], the endpoints belonging to the
same RTP Session are those that share a single SSRC space. That
is, those endpoints can see an SSRC identifier transmitted by any
one of the other endpoints. An endpoint can see an SSRC either
directly in RTP and RTCP packets, or as a contributing source
(CSRC) in RTP packets from a mixer. The RTP Session scope is
hence decided by the endpoints' network interconnection topology,
in combination with RTP and RTCP forwarding strategies deployed by
endpoints and any interconnecting middle nodes.
WebRTC MediaStream: The MediaStream concept defined by the W3C in
the API.
Other terms are used according to their definitions from the RTP
Specification [RFC3550] and WebRTC overview
[I-D.ietf-rtcweb-overview] documents.
4. WebRTC Use of RTP: Core Protocols 4. WebRTC Use of RTP: Core Protocols
The following sections describe the core features of RTP and RTCP The following sections describe the core features of RTP and RTCP
that MUST be implemented, along with the mandated RTP profiles and that need to be implemented, along with the mandated RTP profiles and
payload formats. Also described are the core extensions providing payload formats. Also described are the core extensions providing
essential features that all WebRTC implementations MUST implement to essential features that all WebRTC implementations need to implement
function effectively on today's networks. to function effectively on today's networks.
4.1. RTP and RTCP 4.1. RTP and RTCP
The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be
implemented as the media transport protocol for WebRTC. RTP itself implemented as the media transport protocol for WebRTC. RTP itself
comprises two parts: the RTP data transfer protocol, and the RTP comprises two parts: the RTP data transfer protocol, and the RTP
control protocol (RTCP). RTCP is a fundamental and integral part of control protocol (RTCP). RTCP is a fundamental and integral part of
RTP, and MUST be implemented in all WebRTC applications. RTP, and MUST be implemented 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.
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. and resolution for SSRC values (but see also Section 4.8).
o Support reception of RTP data packets containing CSRC lists, as o Support for reception of RTP data packets containing CSRC lists,
generated by RTP mixers. 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, with RECOMMENDED support for the rapid RTP RTCP Sender Reports, to allow a receiver to implement lip-sync,
synchronisation extensions (see Section 5.2.1). with RECOMMENDED support for the rapid RTP synchronisation
extensions (see Section 5.2.1).
o Support for standard RTCP packet types, include SR, RR, SDES, and o Support for sending and receiving RTCP SR, RR, SDES, and BYE
BYE packets. packet types, with OPTIONAL support for other RTCP packet types;
implementations MUST ignore unknown RTCP packet types.
o Support for multiple end-points in a single RTP session, and for o Support for multiple end-points in a single RTP session, and for
scaling the RTCP transmission interval according to the number of scaling the RTCP transmission interval according to the number of
participants in the session; support randomised RTCP transmission participants in the session; support for randomised RTCP
intervals to avoid synchronisation of RTCP reports. transmission intervals to avoid synchronisation of RTCP reports;
support for RTCP timer reconsideration.
o Support for configuring the RTCP bandwidth as a fraction of the
media bandwidth, and for configuring the fraction of the RTCP
bandwidth allocated to senders, e.g., using the SDP "b=" line.
It is known that a significant number of legacy RTP implementations, It is known that a significant number of legacy RTP implementations,
especially those targetted for purely VoIP systems, do not support especially those targeted at VoIP-only systems, do not support all of
all of the above features. the above features, and in some cases do not support RTCP at all.
Implementers are advised to consider the requirements for graceful
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 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 Real-time Transport Control Protocol (RTCP)-
Based Feedback (RTP/SAVPF)" [RFC5124] is REQUIRED to be implemented. Based Feedback (RTP/SAVPF)" [RFC5124] is REQUIRED to be implemented.
This builds on the basic RTP/AVP profile [RFC3551], the RTP profile This builds on the basic RTP/AVP profile [RFC3551], the RTP profile
for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP
profile (RTP/SAVP) [RFC3711]. profile (RTP/SAVP) [RFC3711].
The RTP/AVPF part of RTP/SAVPF is required to get the improved RTCP The RTCP-based feedback extensions are needed for the improved RTCP
timer model, that allows more flexible transmission of RTCP packets timer model, that allows more flexible transmission of RTCP packets
in response to events, rather than strictly according to bandwidth. in response to events, rather than strictly according to bandwidth.
This is vital for being able to report congestion events. The RTP/ This is vital for being able to report congestion events. These
AVPF profile also saves RTCP bandwidth, and will commonly only use extensions also save RTCP bandwidth, and will commonly only use the
the full RTCP bandwidth allocation when there are many events that full RTCP bandwidth allocation if there are many events that require
require feedback. The RTP/AVPF functionality is also needed to make feedback. They are also needed to make use of the RTP conferencing
use of the RTP conferencing extensions discussed in Section 5.1. extensions discussed in Section 5.1.
Note: The enhanced RTCP timer model defined in the RTP/AVPF Note: The enhanced RTCP timer model defined in the RTP/AVPF
profile is backwards compatible with legacy systems that implement profile is backwards compatible with legacy systems that implement
only the base RTP/AVP profile, given some constraints on parameter only the base RTP/AVP profile, given some constraints on parameter
configuration such as the RTCP bandwidth value and "trr-int" (the configuration such as the RTCP bandwidth value and "trr-int" (the
most important factor for interworking with RTP/AVP end-points via most important factor for interworking with RTP/AVP end-points via
a gateway is to set the trr-int parameter to a value representing a gateway is to set the trr-int parameter to a value representing
4 seconds). 4 seconds).
The RTP/SAVP part of the RTP/SAVPF profile is for support for Secure The secure RTP profile is needed to provide SRTP media encryption,
RTP (SRTP) [RFC3711]. This provides media encryption, integrity integrity protection, replay protection and a limited form of source
protection, replay protection and a limited form of source
authentication. authentication.
WebRTC implementation MUST NOT send packets using the RTP/AVP profile WebRTC implementations MUST NOT send packets using the basic RTP/AVP
or the RTP/AVPF profile; they MUST use the RTP/SAVPF profile. WebRTC profile or the RTP/AVPF profile; they MUST employ the full RTP/SAVPF
implementations MUST support DTLS-SRTP [RFC5764] for key-management. profile to protect all RTP and RTCP packets that are generated. The
default and mandatory-to-implement transforms listed in Section 5 of
[RFC3711] SHALL apply.
(tbd: There is ongoing discussion on what additional keying mechanism Implementations MUST support DTLS-SRTP [RFC5764] for key-management.
is to be required, what are the mandated cryptographic transforms. Other key management schemes MAY be supported.
This section needs to be updated based on the results of that
discussion.)
4.3. Choice of RTP Payload Formats 4.3. Choice of RTP Payload Formats
(tbd: say something about the choice of RTP Payload Format for The requirement from Section 6 of [RFC3551] that "Audio applications
WebRTC. If there is a mandatory to implement set of codecs, this operating under this profile SHOULD, at a minimum, be able to send
should reference them. In any case, it should reference a discussion and/or receive payload types 0 (PCMU) and 5 (DVI4)" applies, since
of signalling for the choice of codec, once that discussion reaches Section 4.2 of this memo mandates the use of the RTP/SAVPF profile,
closure.) which inherits this restriction from the RTP/AVP profile.
Endpoints may signal support for multiple media formats, or multiple
(tbd: there is ongoing discussion on whether support for other audio
and video codecs is to be mandated)
Endpoints MAY signal support for multiple media formats, or multiple
configurations of a single format, provided each uses a different RTP configurations of a single format, provided each uses a different RTP
payload type number. An endpoint that has signalled it's support for payload type number. An endpoint that has signalled its support for
multiple formats is REQUIRED to accept data in any of those formats multiple formats is REQUIRED to accept data in any of those formats
at any time, unless it has previously signalled limitations on it's at any time, unless it has previously signalled limitations on its
decoding capability. This is modified if several media types are decoding capability.
sent in the same RTP session, in that case a source (SSRC) is
restricted to switch between any RTP payload format established for This requirement is constrained if several media types are sent in
the media type that is being sent by that source; see Section 4.4. the same RTP session. In such a case, a source (SSRC) is restricted
To support rapid rate adaptation, RTP does not require signalling in to switching only between the RTP payload formats signalled for the
media type that is being sent by that source; see Section 4.4. To
support rapid rate adaptation, RTP does not require signalling in
advance for changes between payload formats that were signalled advance for changes between payload formats that were signalled
during session setup. during session setup.
An RTP sender that changes between two RTP payload types that use
different RTP clock rates MUST follow the recommendations in Section
4.1 of [I-D.ietf-avtext-multiple-clock-rates]. RTP receivers MUST
follow the recommendations in Section 4.3 of
[I-D.ietf-avtext-multiple-clock-rates], in order to support sources
that switch between clock rates in an RTP session (these
recommendations for receivers are backwards compatible with the case
where senders use only a single clock rate).
4.4. RTP Session Multiplexing 4.4. RTP Session Multiplexing
An association amongst a set of participants communicating with RTP An association amongst a set of participants communicating with RTP
is known as an RTP session. A participant may be involved in is known as an RTP session. A participant can be involved in
multiple RTP sessions at the same time. In a multimedia session, multiple RTP sessions at the same time. In a multimedia session,
each medium has typically been carried in a separate RTP session with 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 its own RTCP packets (i.e., one RTP session for the audio, with a
separate RTP session running on a different transport connection for separate RTP session using a different transport address for the
the video; if SDP is used, this corresponds to one RTP session for video; if SDP is used, this corresponds to one RTP session for each
each "m=" line in the SDP). WebRTC implementations of RTP are "m=" line in the SDP). WebRTC implementations of RTP are REQUIRED to
REQUIRED to implement support for multimedia sessions in this way, implement support for multimedia sessions in this way, for
for compatibility with legacy systems. compatibility with legacy systems.
In today's networks, however, with the widespread use of Network In today's networks, however, with the widespread use of Network
Address/Port Translators (NAT/NAPT) and Firewalls (FW), it is Address/Port Translators (NAT/NAPT) and Firewalls (FW), it is
desirable to reduce the number of transport layer ports used by real- desirable to reduce the number of transport addresses used by real-
time media applications using RTP by combining multimedia traffic in time media applications using RTP by combining multimedia traffic in
a single RTP session. (Details of how this is to be done are tbd, a single RTP session. (Details of how this is to be done are tbd,
but see [I-D.lennox-rtcweb-rtp-media-type-mux], but see [I-D.lennox-rtcweb-rtp-media-type-mux],
[I-D.holmberg-mmusic-sdp-bundle-negotiation] and [I-D.holmberg-mmusic-sdp-bundle-negotiation] and
[I-D.westerlund-avtcore-multiplex-architecture].) Using a single RTP [I-D.westerlund-avtcore-multiplex-architecture].) Using a single RTP
session also effects the possibility for differentiated treament of session also effects the possibility for differentiated treatment of
media flows. This is further discussed in Section 12.9. media flows. This is further discussed in Section 12.9.
WebRTC implementations of RTP are REQUIRED to support multiplexing of WebRTC implementations of RTP are REQUIRED to support multiplexing of
a multimedia session onto a single RTP session according to (tbd). a multimedia session onto a single RTP session according to (tbd).
If such RTP session multiplexing is to be used, this MUST be If such RTP session multiplexing is to be used, this MUST be
negotiated during the signalling phase. Support for multiple RTP negotiated during the signalling phase. Support for multiple RTP
sessions over a single UDP flow as defined by sessions over a single UDP flow as defined by
[I-D.westerlund-avtcore-transport-multiplexing] is RECOMMENDED. [I-D.westerlund-avtcore-transport-multiplexing] is RECOMMENDED/
OPTIONAL.
4.5. RTP and RTCP Multiplexing (tbd: No consensus on the level of including support of Multiple RTP
sessions over a single UDP flow.)
Historically, RTP and RTCP have been run on separate transport-layer 4.5. RTP and RTCP Multiplexing
ports (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/Port
Translation (NAPT) this has become problematic, since maintaining
multiple NAT bindings can be costly. It also complicates firewall
administration, since multiple ports must be opened to allow RTP
traffic. To reduce these costs and session setup times, support for
multiplexing RTP data packets and RTCP control packets on a single
port [RFC5761] for each RTP session is REQUIRED.
(tbd: Are WebRTC implementations required to support the case where Historically, RTP and RTCP have been run on separate transport layer
the RTP and RTCP are run on separate UDP ports, for interoperability addresses (e.g., two UDP ports for each RTP session, one port for RTP
with legacy systems?) and one port for RTCP). With the increased use of Network Address/
Port Translation (NAPT) this has become problematic, since
maintaining multiple NAT bindings can be costly. It also complicates
firewall administration, since multiple ports need to be opened to
allow RTP traffic. To reduce these costs and session setup times,
support for multiplexing RTP data packets and RTCP control packets on
a single port for each RTP session is REQUIRED, as specified in
[RFC5761]. For backwards compatibility, implementations are also
REQUIRED to support sending of RTP and RTCP to separate destination
ports.
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 may be useful to keep- if there is no active media traffic. This can be useful to keep NAT
alive NAT bindings, 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]
requires that those compound packets start with an Sender Report (SR) requires that those compound packets start with an Sender Report (SR)
or Receiver Report (RR) packet. When using frequent RTCP feedback or Receiver Report (RR) packet. When using frequent RTCP feedback
messages, these general statistics are not needed in every packet and messages, these general statistics are not needed in every packet and
unnecessarily increase the mean RTCP packet size. This can limit the unnecessarily increase the mean RTCP packet size. This can limit the
frequency at which RTCP packets can be sent within the RTCP bandwidth frequency at which RTCP packets can be sent within the RTCP bandwidth
share. 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 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 application quickly feedback, in turn, is essential to make real-time applications
aware of changing network conditions and allow them to adapt their quickly aware of changing network conditions, and to allow them to
transmission and encoding behaviour. Support for RFC5506 is adapt their transmission and encoding behaviour. Support for sending
REQUIRED. RTCP feedback packets as [RFC5506] non-compound packets is REQUIRED
when signalled. For backwards compatibility, implementations are
also REQUIRED to support the use of compound RTCP feedback packets.
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 Symmetric RTP [RFC4961]. This requires that REQUIRED to implement and use Symmetric RTP [RFC4961]. This requires
the IP address and port used for sending and receiving RTP and RTCP that the IP address and port used for sending and receiving RTP and
packets are identical. The reasons for using symmetric RTP is RTCP packets are identical. The reasons for using symmetric RTP is
primarily to avoid issues with NAT and Firewalls by ensuring that the primarily to avoid issues with NAT and Firewalls by ensuring that the
flow is actually bi-directional and thus kept alive and registered as flow is actually bi-directional and thus kept alive and registered as
flow the intended recipient actually wants. In addition it saves flow the intended recipient actually wants. In addition, it saves
resources in the form of ports at the end-points, but also in the resources, specifically ports at the end-points, but also in the
network as NAT mappings or firewall state is not unnecessary bloated. network as NAT mappings or firewall state is not unnecessary bloated.
Also the amount of QoS state is reduced. Also the amount of QoS state is reduced.
4.8. Generation of the RTCP Canonical Name (CNAME) 4.8. Choice of RTP Synchronisation Source (SSRC)
Implementations are REQUIRED to support signalled RTP SSRC values,
using the "a=ssrc:" SDP attribute defined in Sections 4.1 and 5 of
[RFC5576], and MUST also support the "previous-ssrc" source attribute
defined in Section 6.2 of [RFC5576]. Other attributes defined in
[RFC5576] MAY be supported.
Use of the "a=ssrc:" attribute is OPTIONAL. Implementations MUST
support random SSRC assignment, and MUST support SSRC collision
detection and resolution, both according to [RFC3550].
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 may 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, it's RTCP CNAME detected, or when the RTP application is restarted, its RTCP CNAME is
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. For proper identified and associated with their RTP media streams within a set
functionality, each RTP endpoint needs to have a unique RTCP CNAME of related RTP sessions. For proper functionality, each RTP endpoint
value. needs to have a unique RTCP CNAME value.
The RTP specification [RFC3550] includes guidelines for choosing a The RTP specification [RFC3550] includes guidelines for choosing a
unique RTP CNAME, but these are not sufficient in the presence of NAT unique RTP CNAME, but these are not sufficient in the presence of NAT
devices. In addition, some may find long-term persistent identifiers devices. In addition, long-term persistent identifiers can be
problematic from a privacy viewpoint. Accordingly, support for problematic from a privacy viewpoint. Accordingly, support for
generating a short-term persistent RTCP CNAMEs following method (b) generating a short-term persistent RTCP CNAMEs following method (b)
specified in Section 4.2 of "Guidelines for Choosing RTP Control specified in Section 4.2 of "Guidelines for Choosing RTP Control
Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is REQUIRED, Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is RECOMMENDED.
since this addresses both concerns. Note, however, that this does not resolve the privacy concern as
there is not sufficient randomness to avoid tracking of an end-point.
An WebRTC end-point MUST support reception of any CNAME that matches
the syntax limitations specified by the RTP specification [RFC3550]
and cannot assume that any CNAME will be according to the recommended
form above.
(tbd: there seems to be a growing consensus that the working group
wants randomly-chosen CNAME values; need to reference a draft that
describes how this is to be done)
5. WebRTC Use of RTP: Extensions 5. WebRTC Use of RTP: Extensions
There are a number of RTP extensions that are either required to There are a number of RTP extensions that are either needed to obtain
obtain full functionality, or extremely useful to improve on the full functionality, or extremely useful to improve on the baseline
baseline performance, in the WebRTC application context. One set of performance, in the WebRTC application context. One set of these
these extensions is related to conferencing, while others are more extensions is related to conferencing, while others are more generic
generic in nature. The following subsections describe the various in nature. The following subsections describe the various RTP
RTP extensions mandated or strongly recommended within WebRTC. 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 was popular in early deployments, in
today's practice, overlay-based conferencing dominates, typically today's practice, overlay-based conferencing dominates, typically
using one or more central servers to connect endpoints in a star or using one or more central servers to connect endpoints in a star or
flat tree topology. These central servers can be implemented in a flat tree topology. These central servers can be implemented in a
number of ways as discussed in Appendix A, and in the memo on RTP number of ways as discussed in Appendix A, and in the memo on RTP
Topologies [RFC5117]. Topologies [RFC5117].
As discussed in Section 3.5 of [RFC5117], the use of a video As discussed in Section 3.5 of [RFC5117], the use of a video
switching MCU makes the use of RTCP for congestion control, or any switching MCU makes the use of RTCP for congestion control, or any
type of quality reports, very problematic. Also, as discussed in type of quality reports, very problematic. Also, as discussed in
section 3.6 of [RFC5117], the use of a content modifying MCU with section 3.6 of [RFC5117], the use of a content modifying MCU with
RTCP termination breaks RTP loop detection and removes the ability RTCP termination breaks RTP loop detection and removes the ability
for receivers to identify active senders. Accordingly, only RTP for receivers to identify active senders. RTP Transport Translators
Transport Translators (relays), RTP Mixers, and end-point based (Topo-Translator) are not of immediate interest to WebRTC, although
forwarding topologies are supported in WebRTC. These RECOMMENDED the main difference compared to point to point is the possibility of
topologies are expected to be supported by all WebRTC end-points seeing multiple different transport paths in any RTCP feedback.
(these three topologies require no special support in the end-point, Accordingly, only Point to Point (Topo-Point-to-Point), Multiple
if the RTP features mandated in this memo are implemented). 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 protocol extensions to be used with conferencing, described The RTP extensions described below to be used with centralised
below, are not required for correctness; an RTP endpoint that does conferencing -- where one RTP Mixer (e.g., a conference bridge)
not implement these extensions will work correctly, but offer poor receives a participant's RTP media streams and distributes them to
performance. Support for the listed extensions will greatly improve the other participants -- are not necessary for interoperability; an
the quality of experience, however, in the context of centralised RTP endpoint that does not implement these extensions will work
conferencing, where one RTP Mixer (Conference Focus) receives a correctly, but may offer poor performance. Support for the listed
participants media streams and distribute them to the other extensions will greatly improve the quality of experience and, to
participants. These messages are defined in the Extended RTP Profile provide a reasonable baseline quality, some these extensions are
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/ mandatory to be supported by WebRTC end-points.
AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
5.1.1. Full Intra Request The RTCP packets assisting in such operation are defined in the
Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-
Based Feedback (RTP/AVPF) [RFC4585] and the "Codec Control Messages
in the RTP Audio-Visual Profile with Feedback (AVPF)" (CCM) [RFC5104]
and are fully usable by the Secure variant of this profile (RTP/
SAVPF) [RFC5124].
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 have 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 predicted media encoding with receivers can decode the video or other predictive media encoding
long prediction chains. It is REQUIRED that this feedback message is with long prediction chains. It is REQUIRED that this feedback
supported by RTP senders in WebRTC, since it greatly improves the message is supported by RTP senders in WebRTC, since it greatly
user experience when using centralised mixers-based conferencing. improves the user experience when using centralised mixers-based
conferencing.
5.1.2. Picture Loss Indication 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 can exist multiple methods to fulfil the Request above as there there may be multiple methods to fulfill the
request. It is RECOMMENDED that this feedback message is supported request. It is REQUIRED that senders understand and react to this
as a loss tolerance mechanism. feedback message as a loss tolerance mechanism; receivers MAY send
PLI messages.
5.1.3. Slice Loss Indication 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
macroblocks, and would like to have these repaired somehow. The use macroblocks, and would like to have these repaired somehow. The use
of this feedback message is OPTIONAL as a loss tolerance mechanism. of this feedback message is OPTIONAL as a loss tolerance mechanism.
5.1.4. Reference Picture Selection Indication 5.1.4. Reference Picture Selection Indication (RPSI)
Reference Picture Selection Indication (RPSI) is defined in Section Reference Picture Selection Indication (RPSI) is defined in Section
6.3.3 of the RTP/AVPF profile [RFC4585]. Some video coding standards 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video coding standards
allow the use of older reference pictures than the most recent one allow the use of older reference pictures than the most recent one
for predictive coding. If such a codec is in used, and if the for predictive coding. If such a codec is in used, and if the
encoder has learned about a loss of encoder-decoder synchronicity, a encoder has learned about a loss of encoder-decoder synchronisation,
known-as-correct reference picture can be used for future coding. a known-as-correct reference picture can be used for future coding.
The RPSI message allows this to be signalled. The use of this RTCP The RPSI message allows this to be signalled.
feedback message is OPTIONAL as a loss tolerance mechanism.
5.1.5. Temporary Maximum Media Stream Bit Rate Request Support for RPSI messages is OPTIONAL.
5.1.5. Temporal-Spatial Trade-off Request (TSTR)
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
to ask the video encoder to change the trade-off it makes between
temporal and spatial resolution, for example to prefer high spatial
image quality but low frame rate.
Support for TSTR requests and notifications is OPTIONAL.
5.1.6. Temporary Maximum Media Stream Bit Rate Request
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 is 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 for various reasons, and can for example this receiver. This may have various reasons; for example, an RTP
be used by an RTP mixer to limit the media sender being forwarded by mixer may use this message to limit the media rate of the sender
the mixer (without doing media transcoding) to fit the bottlenecks being forwarded by the mixer (without doing media transcoding) to fit
existing towards the other session participants. It is REQUIRED that the bottlenecks existing towards the other session participants. It
this feedback message is supported. is REQUIRED that this feedback message is supported. A RTP media
stream sender receiving a TMMBR for its SSRC MUST follow the
limitations set by the message; 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.
skipping to change at page 13, line 11 skipping to change at page 15, line 25
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. The rapid synchronisation
extensions use the general RTP header extension mechanism [RFC5285], extensions use the general RTP header extension mechanism [RFC5285],
which requires signalling, but are otherwise backwards compatible. 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 [RFC6464] is an RTP header extension The Client to Mixer Audio Level extension [RFC6464] is an RTP header
used by a client to inform a mixer about the level of audio activity extension used by a client to inform a mixer about the level of audio
in the packet the header is attached to. This enables a central node activity in the packet to which the header is attached. This enables
to make mixing or selection decisions without decoding or detailed a central node to make mixing or selection decisions without decoding
inspection of the payload. Thus reducing the needed complexity in or detailed inspection of the payload, reducing the complexity in
some types of central RTP nodes. It can also be used to save some types of central RTP nodes. It can also save decoding resources
decoding resources in a WebRTC receiver in a mesh topology, which if in receivers, which can choose to decode only the most relevant RTP
it has limited decoding resources, may select to decode only the most media streams based on audio activity levels.
relevant media streams based on audio activity levels.
The Client-to-Mixer Audio Level [RFC6464] extension is RECOMMENDED to The Client-to-Mixer Audio Level [RFC6464] extension is RECOMMENDED to
be implemented. If it is implemented, it is REQUIRED that the header be implemented. 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.
5.2.3. Mixer to Client Audio Level 5.2.3. Mixer-to-Client Audio Level
The Mixer to Client Audio Level header extension [RFC6465] provides The Mixer to Client Audio Level header extension [RFC6465] provides
the client with the audio level of the different sources mixed into a the client with the audio level of the different sources mixed into a
common mix by a RTP mixer. This enables a user interface to indicate common mix by a RTP mixer. This enables a user interface to indicate
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 some tools that can make RTP flows robust against Packet
loss and reduce the impact on media quality. However they all add loss and reduce the impact on media quality. However, they all add
extra bits compared to a non-robust stream. These extra bits need to extra bits compared to a non-robust stream. These extra bits need to
be considered, and the aggregate bit-rate must be rate controlled. be considered, and the aggregate bit-rate must be rate-controlled.
Thus improving robustness might require a lower base encoding Thus, improving robustness might require a lower base encoding
quality, but has the potential to give that quality with fewer quality, but has the potential to deliver that quality with fewer
errors. The mechanisms described in the following sub-sections can errors. The mechanisms described in the following sub-sections can
be used to improve tolerance to packet loss. be used to improve tolerance to packet loss.
6.1. Retransmission 6.1. Negative Acknowledgements and RTP Retransmission
Support for RTP retransmission as defined by "RTP Retransmission
Payload Format" [RFC4588] is RECOMMENDED.
The retransmission scheme in RTP allows flexible application of
retransmissions. Only selected missing packets can be requested by
the receiver. It also allows for the sender to prioritise between
missing packets based on senders knowledge about their content.
Compared to TCP, RTP retransmission also allows one to give up on a
packet that despite retransmission(s) still has not been received
within a time window.
"WebRTC Media Transport Requirements" [I-D.cbran-rtcweb-data] raises
two issues that they think makes RTP Retransmission unsuitable for
WebRTC. We here consider these issues and explain why they are in
fact not a reason to exclude RTP retransmission from the tool box
available to WebRTC media sessions.
The additional latency added by [RFC4588] will exceed the latency As a consequence of supporting the RTP/SAVPF profile, implementations
threshold for interactive voice and video: RTP Retransmission will will support negative acknowlegdements (NACKs) for RTP data packets
require at least one round trip time for a retransmission request [RFC4585]. This feedback can be used to inform a sender of the loss
and repair packet to arrive. Thus the general suitability of of particular RTP packets, subject to the capacity limitations of the
using retransmissions will depend on the actual network path RTCP feedback channel. A sender can use this information to optimise
latency between the end-points. In many of the actual usages the the user experience by adapting the media encoding to compensate for
latency between two end-points will be low enough for RTP known lost packets, for example.
retransmission to be effective. Interactive communication with
end-to-end delays of 400 ms still provide a fair quality. Even
removing half of that in end-point delays allows functional
retransmission between end-points on the same continent. In
addition, some applications may accept temporary delay spikes to
allow for retransmission of crucial codec information such an
parameter sets, intra picture etc, rather than getting no media at
all.
The undesirable increase in packet transmission at the point when Senders are REQUIRED to understand the Generic NACK message defined
congestion occurs: Congestion loss will impact the rate controls in Section 6.2.1 of [RFC4585], but MAY choose to ignore this feedback
view of available bit-rate for transmission. When using (following Section 4.2 of [RFC4585]). Receivers MAY send NACKs for
retransmission one will have to prioritise between performing missing RTP packets; [RFC4585] provides some guidelines on when to
retransmissions and the quality one can achieve with ones send NACKs. It is not expected that a receiver will send a NACK for
adaptable codecs. In many use cases one prefer error free or low every lost RTP packet, rather it should consider the cost of sending
rates of error with reduced base quality over high degrees of NACK feedback, and the importance of the lost packet, to make an
error at a higher base quality. informed decision on whether it is worth telling the sender about a
packet loss event.
The WebRTC end-point implementations will need to both select when to The RTP Retransmission Payload Format [RFC4588] offers the ability to
enable RTP retransmissions based on API settings and measurements of retransmit lost packets based on NACK feedback. Retransmission needs
the actual round trip time. In addition for each NACK request that a to be used with care in interactive real-time applications to ensure
media sender receives it will need to make a prioritisation based on that the retransmitted packet arrives in time to be useful, but can
the importance of the requested media, the probability that the be effective in environments with relatively low network RTT (an RTP
packet will reach the receiver in time for being usable, the sender can estimate the RTT to the receivers using the information in
consumption of available bit-rate and the impact of the media quality RTCP SR and RR packets). The use of retransmissions can also
for new encodings. increase the forward RTP bandwidth, and can potentially worsen the
problem if the packet loss was caused by network congestion. We
note, however, that retransmission of an important lost packet to
repair decoder state may be lower cost than sending a full intra
frame. It is not appropriate to blindly retransmit RTP packets in
response to a NACK. The importance of lost packets and the
likelihood of them arriving in time to be useful needs to be
considered before RTP retransmission is used.
To conclude, the issues raised are implementation concerns that an Receivers are REQUIRED to implement support for RTP retransmission
implementation needs to take into consideration, they are not packets [RFC4588]. Senders MAY send RTP retransmission packets in
arguments against including a highly versatile and efficient packet response to NACKs if the RTP retransmission payload format has been
loss repair mechanism. 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
RTP sender is not expected to retransmit every NACKed packet.
6.2. Forward Error Correction (FEC) 6.2. Forward Error Correction (FEC)
Support of some type of FEC to combat the effects of packet loss is The use of Forward Error Correction (FEC) can provide an effective
beneficial, but is heavily application dependent. However, some FEC protection against some degree of packet loss, at the cost of steady
mechanisms are encumbered. bandwidth overhead. There are several FEC schemes that are defined
for use with RTP. Some of these schemes are specific to a particular
The main benefit from FEC is the relatively low additional delay RTP payload format, others operate across RTP packets and can be used
needed to protect against packet losses. The transmission of any with any payload format. It should be noted that using redundancy
repair packets should preferably be done with a time delay that is encoding or FEC will lead to increased playout delay, which should be
just larger than any loss events normally encountered. That way the considered when choosing the redundancy or FEC formats and their
repair packet isn't also lost in the same event as the source data. respective parameters.
The amount of repair packets needed varies depending on the amount
and pattern of packet loss to be recovered, and on the mechanism used
to derive repair data. The later choice also effects the the
additional delay required to both encode the repair packets and in
the receiver to be able to recover the lost packet(s).
6.2.1. Basic Redundancy
The method for providing basic redundancy is to simply retransmit a
some time earlier sent packet. This is relatively simple in theory,
i.e. one saves any outgoing source (original) packet in a buffer
marked with a timestamp of actual transmission, some X ms later one
transmit this packet again. Where X is selected to be longer than
the common loss events. Thus any loss events shorter than X can be
recovered assuming that one doesn't get an another loss event before
all the packets lost in the first event has been received.
The downside of basic redundancy is the overhead. To provide each
packet with once chance of recovery, then the transmission rate
increases with 100% as one needs to send each packet twice. It is
possible to only redundantly send really important packets thus
reducing the overhead below 100% for some other trade-off is
overhead.
In addition the basic retransmission of the same packet using the
same SSRC in the same RTP session is not possible in RTP context.
The reason is that one would then destroy the RTCP reporting if one
sends the same packet twice with the same sequence number. Thus one
needs more elaborate mechanisms.
RTP Payload Format Support: Some RTP payload format do support basic
redundancy within the RTP paylaod format itself. Examples are
AMR-WB [RFC4867] and G.719 [RFC5404].
RTP Payload for Redundant Audio Data: This audio and text redundancy
format defined in [RFC2198] allows for multiple levels of
redundancy with different delay in their transmissions, as long as
the source plus payload parts to be redundantly transmitted
together fits into one MTU. This should work fine for most
interactive audio and text use cases as both the codec bit-rates
and the framing intervals normally allow for this requirement to
hold. This payload format also don't increase the packet rate, as
original data and redundant data are sent together. This format
does not allow perfect recovery, only recovery of information
deemed necessary for audio, for example the sequence number of the
original data is lost.
RTP Retransmission Format: The RTP Retransmission Payload format
[RFC4588] can be used to pro-actively send redundant packets using
either SSRC or session multiplexing. By using different SSRCs or
a different session for the redundant packets the RTCP receiver
reports will be correct. The retransmission payload format is
used to recover the packets original data thus enabling a perfect
recovery.
Duplication Grouping Semantics in the Session Description Protocol:
This [I-D.begen-mmusic-redundancy-grouping] is proposal for new
SDP signalling to indicate media stream duplication using
different RTP sessions, or different SSRCs to separate the source
and the redundant copy of the stream.
6.2.2. Block Based FEC
Block based redundancy collects a number of source packets into a
data block for processing. The processing results in some number of
repair packets that is then transmitted to the other end allowing the
receiver to attempt to recover some number of lost packets in the
block. The benefit of block based approaches is the overhead which
can be lower than 100% and still recover one or more lost source
packet from the block. The optimal block codes allows for each
received repair packet to repair a single loss within the block.
Thus 3 repair packets that are received should allow for any set of 3
packets within the block to be recovered. In reality one commonly
don't reach this level of performance for any block sizes and number
of repair packets, and taking the computational complexity into
account there are even more trade-offs to make among the codes.
One result of the block based approach is the extra delay, as one
needs to collect enough data together before being able to calculate
the repair packets. In addition sufficient amount of the block needs
to be received prior to recovery. Thus additional delay are added on
both sending and receiving side to ensure possibility to recover any
packet within the block.
The redundancy overhead and the transmission pattern of source and
repair data can be altered from block to block, thus allowing a
adaptive process adjusting to meet the actual amount of loss seen on
the network path and reported in RTCP.
The alternatives that exist for block based FEC with RTP are the
following:
RTP Payload Format for Generic Forward Error Correction: This RTP
payload format [RFC5109] defines an XOR based recovery packet.
This is the simplest processing wise that an block based FEC
scheme can be. It also results in some limited properties, as
each repair packet can only repair a single loss. To handle
multiple close losses a scheme of hierarchical encodings are need.
Thus increasing the overhead significantly.
Forward Error Correction (FEC) Framework: This framework
[I-D.ietf-fecframe-framework] defines how not only RTP packets but
how arbitrary packet flows can be protected. Some solutions
produced or under development in FECFRAME WG are RTP specific.
There exist alternatives supporting block codes such as Reed-
Salomon and Raptor.
6.2.3. Recommendations for FEC If an RTP payload format negotiated for use in a WebRTC session
supports redundant transmission or FEC as a standard feature of that
payload format, then that support MAY be used in the WebRTC session,
subject to any appropriate signalling.
Open Issue: Decision of need for FEC and if to be included in There are several block-based FEC schemes that are designed for use
recommendation which FEC scheme to be supported needs to be with RTP independent of the chosen RTP payload format. At the time
documented. of this writing there is no consensus on which, if any, of these FEC
schemes is appropriate for use in the WebRTC context. Accordingly,
this memo makes no recommendation on the choice of block-based FEC
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 very varied network environment with a WebRTC will be used in very varied network environment with a
hetrogenous set of link technologies, including wired and wireless, heterogeneous set of link technologies, including wired and wireless,
interconnecting peers at different topological locations resulting in interconnecting peers at different topological locations resulting in
network paths with widely varying one way delays, bit-rate capacity, network paths with widely varying one way delays, bit-rate capacity,
load levels and traffic mixes. In addition individual end-points load levels and traffic mixes. In addition, individual end-points
will open one or more WebRTC sessions between one or more peers. will open one or more WebRTC sessions between one or more peers.
Each of these session may contain different mixes of media and data Each of these session may contain different mixes of media and data
flows. Assymetric usage of media bit-rates and number of media flows. Asymmetric usage of media bit-rates and number of RTP media
streams is also to be expected. A single end-point may receive zero streams is also to be expected. A single end-point may receive zero
to many simultanous media streams while itself transmitting one or to many simultaneous RTP media streams while itself transmitting one
more streams. or more streams.
The WebRTC application is very dependent from a quality perspective The WebRTC application is very dependent from a quality perspective
on the media adapation working well so that an end-point doesn't on the media adaptation working well so that an end-point doesn't
transmit significantly more than the path is capable of handling. If transmit significantly more than the path is capable of handling. If
it would, the result would be high levels of packet loss or delay it would, the result would be high levels of packet loss or delay
spikes causing media degradations. spikes causing media quality degradation.
WebRTC applications using more than a single media stream of any WebRTC applications using more than a single RTP media stream of any
media type or data flows has an additional concern. In this case the media type or data flows have an additional concern. In this case,
different flows should try to avoid affecting each other negatively. the different flows should try to avoid affecting each other
In addition in case there is a resource limiation, the available negatively. In addition, in case there is a resource limitation, the
resources needs to be shared. How to share them is something the available resources need to be shared. How to share them is
application should prioritize so that the limiation in quality or something the application should prioritize so that the limitations
capabilities are the ones that provide the least affect on the in quality or capabilities are those that have the least impact on
application. the application.
This hetrogenous situation results in a requirement to have Overall, the diversity of operating environments lead to the need for
functionality that adapts to the available capacity and that competes functionality that adapts to the available capacity and that competes
fairly with other network flows. If it would not compete fairly fairly with other network flows. If it would not compete fairly
enough WebRTC could be used as an attack method for starving out enough WebRTC could be used as an attack method for starving out
other traffic on specific links as long as the attacker is able to other traffic on specific links as long as the attacker is able to
create traffic across a specific link. This is not far-fetched for a create traffic across the links in question. A possible attack
web-service capable of attracting large number of end-points and use scenario is to use a web-service capable of attracting large numbers
the service, combined with BGP routing state a server could pick of end-points, combined with BGP routing state to have the server
client pairs to drive traffic to specific paths. pick client pairs to drive traffic to specific paths.
The above estalish a clear need based on several reasons why there The above clearly motivates the need for a well working media
need to be a well working media adaptation mechanism. This mechanism adaptation mechanism. This mechanism also have a number of
also have a number of requirements on what services it should provide requirements on what services it should provide and what performance
and what performance it needs to provide. it needs to provide.
The biggest issue is that there are no standardised and ready to use The biggest issue is that there are no standardised and ready to use
mechanism that can simply be included in WebRTC. Thus there will be mechanism that can simply be included in WebRTC. Thus, there will be
need for the IETF to produce such a specification. Therefore the a need for the IETF to produce such a specification. Therefore, the
suggested way forward is to specify requirements on any solution for suggested way forward is to specify requirements on any solution for
the media adaptation. These requirements is for now proposed to be the media adaptation. For now, we propose that these requirements be
documented in this specification. In addition a proposed detailed documented in this specification. In addition, a proposed detailed
solution will be developed, but is expected to take longer time to solution will be developed, but is expected to take longer time to
finalize than this document. finalize than this document.
7.1. Congestion Control Requirements 7.1. Congestion Control Requirements
Requirements for congestion control of WebRTC sessions are discussed Requirements for congestion control of WebRTC sessions are discussed
in [I-D.jesup-rtp-congestion-reqs]. in [I-D.jesup-rtp-congestion-reqs].
Implementations are REQUIRED to implement the RTP circuit breakers Implementations are REQUIRED to implement the RTP circuit breakers
described in [I-D.perkins-avtcore-rtp-circuit-breakers]. described in [I-D.perkins-avtcore-rtp-circuit-breakers].
(tbd: Should add the RTP/RTCP Mechanisms that an WebRTC
implementation is required to support. Potential candidates include
Transmission Timestamps (RFC 5450).)
7.2. Rate Control Boundary Conditions 7.2. Rate Control Boundary Conditions
The session establishment signalling will establish certain boundary The session establishment signalling will establish certain boundary
that the media bit-rate adaptation can act within. First of all the that the media bit-rate adaptation can act within. First of all the
set of media codecs provide practical limitations in the supported set of media codecs provide practical limitations in the supported
bit-rate span where it can provide useful quality, which bit-rate span where it can provide useful quality, which
packetization choices that exist. Next the signalling can establish packetization choices that exist. Next the signalling can establish
maximum media bit-rate boundaries using SDP b=AS or b=CT. maximum media bit-rate boundaries using SDP b=AS or b=CT.
7.3. RTCP Limiations (tbd: This section needs expanding on how to use these limits)
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 capabilities of real-time media round trip time. We note that the real-time media traffic may not
traffic to adapt to changing path conditions may be less rapid than have to adapt to changing path conditions as rapidly as needed for
for the elastic applications TCP was designed for, but frequent the elastic applications TCP was designed for, but frequent feedback
feedback is still required to allow the congestion control algorithm is still required to allow the congestion control algorithm to track
to track the path dynamics. the path dynamics.
The total RTCP bandwidth is limited in its transmission rate to a The total RTCP bandwidth is limited in its transmission rate to a
fraction of the RTP traffic (by default 5%). RTCP packets are larger fraction of the RTP traffic (by default 5%). RTCP packets are larger
than, e.g., TCP ACKs (even when non-compound RTCP packets are used). than, e.g., TCP ACKs (even when non-compound RTCP packets are used).
The media stream bit rate thus limits the maximum feedback rate as a The RTP media stream bit rate thus limits the maximum feedback rate
function of the mean RTCP packet size. as a function of the mean RTCP packet size.
Interactive communication may not be able to afford waiting for Interactive communication may not be able to afford waiting for
packet losses to occur to indicate congestion, because an increase in packet losses to occur to indicate congestion, because an increase in
playout delay due to queuing (most prominent in wireless networks) playout delay due to queuing (most prominent in wireless networks)
may easily lead to packets being dropped due to late arrival at the may easily lead to packets being dropped due to late arrival at the
receiver. Therefore, more sophisticated cues may need to be reported receiver. Therefore, more sophisticated cues may need to be reported
-- to be defined in a suitable congestion control framework as noted -- to be defined in a suitable congestion control framework as noted
above -- which, in turn, increase the report size again. For above -- which, in turn, increase the report size again. For
example, different RTCP XR report blocks (jointly) provide the example, different RTCP XR report blocks (jointly) provide the
necessary details to implement a variety of congestion control necessary details to implement a variety of congestion control
skipping to change at page 20, line 10 skipping to change at page 20, line 7
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. Legacy Interop Limitations 7.4. Congestion Control Interoperability With Legacy Systems
Congestion control interoperability with most type of legacy devices,
even using an translator could be difficult. There are numerous
reasons for this:
No RTCP Support: There exist legacy implementations that does not
even implement RTCP at all. Thus no feedback at all is provided.
RTP/AVP Minimal RTCP Interval of 5s: RTP [RFC3550] under the RTP/AVP
profile specifies a recommended minimal fixed interval of 5
seconds. Sending RTCP report blocks as seldom as 5 seconds makes
it very difficult for a sender to use these reports and react to
any congestion event.
RTP/AVP Scaled Minimal Interval: If a legacy device uses the scaled There are legacy implementations that do not implement RTCP, and
minimal RTCP compound interval, the "RECOMMENDED value for the hence do not provide any congestion feedback. Congestion control
reduced minimum in seconds is 360 divided by the session bandwidth cannot be performed with these end-points. WebRTC implementations
in kilobits/second" ([RFC3550], section 6.2). The minimal that must interwork with such end-points MUST limit their
interval drops below a second, still several times the RTT in transmission to a low rate, equivalent to a VoIP call using a low
almost all paths in the Internet, when the session bandwidht bandwidth codec, that is unlikely to cause any significant
becomes 360 kbps. A session bandwidth of 1 Mbps still has a congestion.
minimal interval of 360 ms. Thus, with the exception for rather
high bandwidth sessions, getting frequent enough RTCP Report
Blocks to report on the order of the RTT is very difficult as long
as the legacy device uses the RTP/AVP profile.
RTP/AVPF Supporting Legacy Device: If a legacy device supports RTP/ When interworking with legacy implementations that support RTCP using
AVPF, then that enables negotation of important parameters for the RTP/AVP profile [RFC3551], congestion feedback is provided in
frequent reporting, such as the "trr-int" parameter, and the RTCP RR packets every few seconds. Implementations that are required
possibility that the end-point supports some useful feedback to interwork with such end-points MUST ensure that they keep within
format for congestion control purpose such as TMMBR [RFC5104]. the RTP circuit breaker [I-D.perkins-avtcore-rtp-circuit-breakers]
constraints to limit the congestion they can cause.
It has been suggested on the WebRTC mailing list that if If a legacy end-point supports RTP/AVPF, this enables negotiation of
interoperating with really limited legacy devices an WebRTC end-point important parameters for frequent reporting, such as the "trr-int"
may not send more than 64 kbps of media streams, to avoid it causing parameter, and the possibility that the end-point supports some
massive congestion on most paths in the Internet when communicating useful feedback format for congestion control purpose such as TMMBR
with a legacy node not providing sufficient feedback for effective [RFC5104]. Implementations that are required to interwork with such
congestion control. This warrants further discussion as there is end-points MUST ensure that they stay within the RTP circuit breaker
clearly a number of link layers that don't even provide that amount [I-D.perkins-avtcore-rtp-circuit-breakers] constraints to limit the
of bit-rate consistently, and that assumes no competing traffic. congestion they can cause, but may find that they can achieve better
congestion response depending on the amount of feedback that is
available.
8. WebRTC Use of RTP: Performance Monitoring 8. WebRTC Use of RTP: Performance Monitoring
RTCP does contains a basic set of RTP flow monitoring points like RTCP does contains a basic set of RTP flow monitoring metrics like
packet loss and jitter. There exist a number of extensions that packet loss and jitter. There are a number of extensions that could
could be included in the set to be supported. However, in most cases be included in the set to be supported. However, in most cases which
which RTP monitoring that is needed depends on the application, which RTP monitoring that is needed depends on the application, which makes
makes it difficult to select which to include when the set of it difficult to select which to include when the set of applications
applications is very large. is very large.
Exposing some metrics in the WebRTC API should be considered allowing Exposing some metrics in the WebRTC API should be considered allowing
the application to gather the measurements of interest. However, the application to gather the measurements of interest. However,
security implications for the different data sets exposed will need security implications for the different data sets exposed will need
to be considered in this. to be considered in this.
(tbd: If any RTCP XR metrics should be added is still an open
question, but possible to extend at a later stage)
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
developed. developed.
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 should adopt RTP in others. Where possible, the WebRTC framework should adopt RTP
extensions that are of general utility, to enable easy gatewaying to extensions that are of general utility, to enable easy gatewaying to
other applications using RTP, rather than adopt mechanisms that are other applications using RTP, rather than adopt mechanisms that are
narrowly targetted at specific WebRTC use cases. 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 of an external signalling channel
that can be used to configure the RTP sessions and their features. that can be used to configure the 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 in addition to any
cryptographic transformation of the packet content. As WebRTC cryptographic transformation of the packet content. As WebRTC
requires the usage of the SAVPF profile only a single profile will requires the usage of the RTP/SAVPF profile this can be inferred
need to be signalled. Interworking functions may transform this as there is only a single profile, but in SDP this is still
into SAVP for a legacy use case by indicating to the WebRTC end- required information to be signalled. Interworking functions may
point a SAVPF end-point and limiting the usage of the a=rtcp transform this into RTP/SAVP for a legacy use case by indicating
attribute to indicate a trr-int value of 4 seconds. to the WebRTC end-point a RTP/SAVPF end-point and limiting the
usage of the a=rtcp attribute to indicate a trr-int value of 4
seconds.
Transport Information: Source and destination address(s) and ports Transport Information: Source and destination IP address(s) and
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 end-points will be provided by ICE that signalls WebRTC these transport addresses will be provided by ICE that
candidates and arrive at nominated candidate pairs. If RTP and signals candidates and arrives at nominated candidate address
RTCP multiplexing [RFC5761] is to be used, such that a single port pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such
is used for RTP and RTCP flows, this MUST be signalled (see that a single port is used for RTP and RTCP flows, this MUST be
Section 4.5). If several RTP sessions are to be multiplexed onto signalled (see Section 4.5). If several RTP sessions are to be
a single transport layer flow, this MUST also be signalled (see multiplexed onto a single transport layer flow, this MUST also be
Section 4.4). signalled (see Section 4.4).
RTP Payload Types, media formats, and media format RTP Payload Types, media formats, and media format
parameters: The mapping between media type names (and hence the RTP parameters: The mapping between media type names (and hence the RTP
payload formats to be used) and the RTP payload type numbers must payload formats to be used) and the RTP payload type numbers MUST
be signalled. Each media type may also have a number of media be signalled. Each media type MAY also have a number of media
type parameters that must also be signalled to configure the codec type parameters that MUST also be signalled to configure the codec
and RTP payload format (the "a=fmtp:" line from SDP). and RTP payload format (the "a=fmtp:" line from SDP).
RTP Extensions: The RTP extensions one intends to use need to be RTP Extensions: The RTP extensions to be used SHOULD be agreed upon,
agreed upon, including any parameters for each respective including any parameters for each respective extension. At the
extension. At the very least, this will help avoiding using very least, this will help avoiding using bandwidth for features
bandwidth for features that the other end-point will ignore. But that the other end-point will ignore. But for certain mechanisms
for certain mechanisms there is requirement for this to happen as there is requirement for this to happen as interoperability
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, as described in "Session Description end-points will be necessary. This SHALL be done as described in
Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) "Session Description Protocol (SDP) Bandwidth Modifiers for RTP
Bandwidth" [RFC3556], or something semantically equivalent. This Control Protocol (RTCP) Bandwidth" [RFC3556], or something
also ensures that the end-points have a common view of the RTCP semantically equivalent. This also ensures that the end-points
bandwidth, this is important as too different view of the have a common view of the RTCP bandwidth, this is important as too
bandwidths may lead to failure to interoperate. different view of the bandwidths may lead to failure to
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/answer model, but does require all the necessary parameters to offer/answer model, but does require all the necessary parameters to
be agreed somehow, and provided to the RTP implementation. We note be agreed upon, and provided to the RTP implementation. We note that
that in the WebRTC context it will depend on the signalling model and in the WebRTC context it will depend on the signalling model and API
API how these parameters need to be configured but they will be need how these parameters need to be configured but they will be need to
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 following sections describe how the WebRTC API features map onto The following sections describe how the WebRTC API features map onto
the RTP mechanisms described in this memo. the RTP mechanisms described in this memo.
11.1. API MediaStream to RTP Mapping 11.1. API MediaStream to RTP Mapping
The WebRTC API and its media function have the concept of a The WebRTC API and its media function have the concept of a WebRTC
MediaStream that consists of zero or more tracks. Where a track is MediaStream that consists of zero or more tracks. A track is an
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 coneptual sources, like a audio mix microphone or a camera, but also conceptual sources, like a audio mix
or a video composition. The tracks within a MediaStream are expected or a video composition, are possible. The tracks within a WebRTC
to be synchronized. MediaStream are expected to be synchronized.
A track correspondes to the media received with one particular SSRC. A track correspond to the media received with one particular SSRC.
There might be additional SSRCs associated with that SSRC, like for There might be additional SSRCs associated with that SSRC, like for
RTP retransmission or Forward Error Correction. However, one SSRC RTP retransmission or Forward Error Correction. However, one SSRC
will identify a media stream and its timing. will identify an RTP media stream and its timing.
Thus a MediaStream is a collection of SSRCs carrying the different As a result, a WebRTC MediaStream is a collection of SSRCs carrying
media included in the synchornized aggregate. Thus also the the different media included in the synchronised aggregate.
synchronization state associated with the included SSRCs are part of Therefore, also the synchronization state associated with the
concept. One important thing to consider is that there can be included SSRCs are part of concept. It is important to consider that
multiple different MediaStreams containing a given Track (SSRC). there can be multiple different WebRTC MediaStreams containing a
Thus to avoid unnecessary duplication of media at transport level one given Track (SSRC). To avoid unnecessary duplication of media at the
need to do the binding of which MediaStreams a given SSRC is transport level in such cases, a need arises for a binding defining
associated with at signalling level. which WebRTC MediaStreams a given SSRC is associated with at the
signalling level.
A proposal for how the binding between MediaStreams and SSRC can be A proposal for how the binding between WebRTC MediaStreams and SSRC
done exist in "Cross Session Stream Identification in the Session can be done is specified in "Cross Session Stream Identification in
Description Protocol" [I-D.alvestrand-rtcweb-msid]. the Session Description Protocol" [I-D.alvestrand-rtcweb-msid].
(tbd: This text must be improved and achieved consensus on. Interim
meeting in June 2012 shows large differences in opinions.)
12. RTP Implementation Considerations 12. RTP Implementation Considerations
The following provide some guidance on the implementation of the RTP The following provide some guidance on the implementation of the RTP
features described in this memo. features described in this memo.
This section discusses RTP functionality that is part of the RTP This section discusses RTP functionality that is part of the RTP
standard, required by decisions made, or to enable use cases raised standard, required by decisions made, or to enable use cases raised
and their motivations. This discussion is done from an WebRTC end- and their motivations. This discussion is from an WebRTC end-point
point perspective. It will occassional go into central nodes, but as perspective. It will occasionally talk about central nodes, but as
the specification is for an end-point that is where the focus lies. this specification is for an end-point, this is where the focus lies.
For more discussion on the central nodes and details about RTP For more discussion on the central nodes and details about RTP
topologies please reveiw Appendix A. topologies please see Appendix A.
The section will touch on the relation with certain RTP/RTCP The section will touch on the relation with certain RTP/RTCP
extensions, but will focus on the RTP core functionality. The extensions, but will focus on the RTP core functionality. The
definition of what functionalities and the level of requirement on definition of what functionalities and the level of requirement on
implementing it is defined in Section 2. implementing it is defined in Section 2.
12.1. RTP Sessions and PeerConnection 12.1. RTP Sessions and PeerConnection
An RTP session is an association among RTP nodes, which have one An RTP session is an association among RTP nodes, which have one
common SSRC space. An RTP session can include any number of end- common SSRC space. An RTP session can include any number of end-
points and nodes sourcing, sinking, manipulating or reporting on the points and nodes sourcing, sinking, manipulating or reporting on the
media streams being sent within the RTP session. A PeerConnection RTP media streams being sent within the RTP session. A
being a point to point association between an end-point and another PeerConnection being a point-to-point association between an end-
node. That peer node may be both an end-point or centralized point and another node. That peer node may be both an end-point or
processing node of some type, thus the RTP session may terminate centralized processing node of some type; thus, the RTP session may
immediately on the far end of the PeerConnection, but it may also terminate immediately on the far end of the PeerConnection, but it
continue as further discused below in Multiparty (Section 12.3) and may also continue as further discussed below in Multiparty
Multiple RTP End-points (Section 12.7). (Section 12.3) and Multiple RTP End-points (Section 12.7).
A PeerConnection can contain one or more RTP session depending on how A PeerConnection can contain one or more RTP session depending on how
it is setup and how many UDP flows it uses. A common usage has been it is setup and how many UDP flows it uses. A common usage has been
to have one RTP session per media type, e.g. one for audio and one to have one RTP session per media type, e.g. one for audio and one
for Video, each sent over different UDP flows. However, the default for video, each sent over different UDP flows. However, the default
usage in WebRTC will be to use one RTP session for all media types. usage in WebRTC will be to use one RTP session for all media types.
This usage then uses only one UDP flow, as also RTP and RTCP This usage then uses only one UDP flow, as also RTP and RTCP
multiplexing is mandated (Section 4.5). However, for legacy multiplexing is mandated (Section 4.5). However, for legacy
interworking and network prioritization (Section 12.9) based on flows interworking and network prioritization (Section 12.9) based on
a WebRTC end-point needs to support a mode of operation where one RTP flows, a WebRTC end-point needs to support a mode of operation where
session per media type is used. Currently each RTP session must use one RTP session per media type is used. Currently, each RTP session
its own UDP flow. Discussion are ongoing if a solution enabling must use its own UDP flow. Discussions are ongoing if a solution
multiple RTP sessions over a single UDP flow, see Section 4.4. enabling multiple RTP sessions over a single UDP flow, see
Section 4.4.
The multi-unicast or mesh based multi-party topology (Figure 1) is The multi-unicast- or mesh-based multi-party topology (Figure 1) is a
best to raise in this section as it concers the relation between RTP good example for this section as it concerns the relation between RTP
sessions and PeerConnections. In this topology, each participant sessions and PeerConnections. In this topology, each participant
sends individual unicast RTP/UDP/IP flows to each of the other sends individual unicast RTP/UDP/IP flows to each of the other
participants using independent PeerConnections in a full mesh. This participants using independent PeerConnections in a full mesh. This
topology has the benefit of not requiring central nodes. The topology has the benefit of not requiring central nodes. The
downside is that it increases the used bandwidth at each sender by downside is that it increases the used bandwidth at each sender by
requiring one copy of the media streams for each participant that are requiring one copy of the RTP media streams for each participant that
part of the same session beyond the sender itself. Hence, this are part of the same session beyond the sender itself. Hence, this
topology is limited to scenarios with few participants unless the topology is limited to scenarios with few participants unless the
media is very low bandwidth. media is very low bandwidth.
+---+ +---+ +---+ +---+
| A |<---->| B | | A |<---->| B |
+---+ +---+ +---+ +---+
^ ^ ^ ^
\ / \ /
\ / \ /
v v v v
skipping to change at page 25, line 29 skipping to change at page 25, line 14
session, spanning multiple peer-to-peer transport layer connections, session, spanning multiple peer-to-peer transport layer connections,
or as several pairwise RTP sessions, one between each pair of peers. or as several pairwise RTP sessions, one between each pair of peers.
To maintain a coherent mapping between the relation between RTP To maintain a coherent mapping between the relation between RTP
sessions and PeerConnections we recommend that one implements this as sessions and PeerConnections we recommend that one implements this as
individual RTP sessions. The only downside is that end-point A will individual RTP sessions. The only downside is that end-point A will
not learn of the quality of any transmission happening between B and not learn of the quality of any transmission happening between B and
C based on RTCP. This has not been seen as a significant downside as C based on RTCP. This has not been seen as a significant downside as
no one has yet seen a clear need for why A would need to know about no one has yet seen a clear need for why A would need to know about
the B's and C's communication. An advantage of using separate RTP the B's and C's communication. An advantage of using separate RTP
sessions is that it enables using different media bit-rates to the sessions is that it enables using different media bit-rates to the
differnt peers, thus not forcing B to endure the same quality different peers, thus not forcing B to endure the same quality
reductions if there are limiations in the transport from A to C as C reductions if there are limitations in the transport from A to C as C
will. will.
12.2. Multiple Sources 12.2. Multiple Sources
A WebRTC end-point may have multiple cameras, microphones or audio A WebRTC end-point may have multiple cameras, microphones or audio
inputs thus a single end-point can source multiple media streams inputs and thus a single end-point can source multiple RTP media
concurrently of the same media type. In addition the above discussed streams of the same media type concurrently. Even if an end-point
criteria to support multiple media types in one single RTP session does not have multiple media sources of the same media type it will
results that also an end-point that has one audio and one video be required to support transmission using multiple SSRCs concurrently
source still need two transmit using two SSRCs concurrently. As in the same RTP session. This is due to the requirement on an WebRTC
multi-party conferences are supported, as discussed below in end-point to support multiple media types in one RTP session. For
Section 12.3, a WebRTC end-point will need to be capable of example, one audio and one video source can result in the end-point
receiving, decoding and playout multiple media streams of the same sending with two different SSRCs in the same RTP session. As multi-
type concurrently. party conferences are supported, as discussed below in Section 12.3,
a WebRTC end-point will need to be capable of receiving, decoding and
playout multiple RTP media streams of the same type concurrently.
Open Issue:Are any mechanism needed to signal limiations in the tbd: Are any mechanism needed to signal limitations in the number of
number of SSRC that an end-point can handle? SSRC that an end-point can handle?
12.3. Multiparty 12.3. Multiparty
There exist numerous situations and clear use cases for WebRTC There are numerous situations and clear use cases for WebRTC
supporting sessions supoprting multi-party. This can be realized in supporting RTP sessions supporting multi-party. This can be realized
a number of ways using a number of different implementations in a number of ways using a number of different implementation
strategies. This focus on the different set of WebRTC end-point strategies. In the following, the focus is on the different set of
requirements that arise from different sets of multi-party WebRTC end-point requirements that arise from different sets of
topologies. multi-party topologies.
The multi-unicast mesh (Figure 1) based multi-party topoology The multi-unicast mesh (Figure 1)-based multi-party topology
discussed above provides a non-centralized solution but can easily discussed above provides a non-centralized solution but may incur a
tax the end-points outgoing paths. It may also consume large amount heavy tax on the end-points' outgoing paths. It may also consume
of encoding resources if each outgoing stream is specifically large amount of encoding resources if each outgoing stream is
encoded. If an encoding is transmitted to multiple parties, either specifically encoded. If an encoding is transmitted to multiple
as in the mesh case or when using relaying central nodes (see below) parties, as in some implementations of the mesh case, a requirement
a requirement on the end-point becomes to be able to create media on the end-point becomes to be able to create RTP media streams
streams suitable to multiple destinations requirements. These suitable for multiple destinations requirements. These requirements
requirements may both be dependent on transport path and the may both be dependent on transport path and the different end-points
different end-points preferences related to playout of the media. preferences related to playout of the media.
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Mixer | | Mixer |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 2: RTP Mixer with Only Unicast Paths Figure 2: RTP Mixer with Only Unicast Paths
A Mixer (Figure 2) is an RTP end-point that optimizes the A Mixer (Figure 2) is an RTP end-point that optimizes the
transmission of media streams from certain perspectives, either by transmission of RTP media streams from certain perspectives, either
only sending some of the received media stream to any given receiver by only sending some of the received RTP media stream to any given
or by providing a combined media stream out of a set of contributing receiver or by providing a combined RTP media stream out of a set of
streams. There exist various methods of implementation as discussed contributing streams. There are various methods of implementation as
in Appendix A.3. A common aspect is that these central nodes a discussed in Appendix A.3. A common aspect is that these central
number of tools to control the media encoding provided by a WebRTC nodes may use a number of tools to control the media encoding
end-point. This includes functions like requesting breaking the provided by a WebRTC end-point. This includes functions like
encoding chain and have the encoder produce a so called Intra frame. requesting breaking the encoding chain and have the encoder produce a
Another is limiting the bit-rate of a given stream to better suit the so called Intra frame. Another is limiting the bit-rate of a given
mixer view of the multiple down-streams. Others are controling the stream to better suit the mixer view of the multiple down-streams.
most suitable frame-rate, picture resultion, the trade-off between Others are controlling the most suitable frame-rate, picture
frame-rate and spatial quality. resolution, the trade-off between frame-rate and spatial quality.
A mixer gets a significant responsibility to correctly perform A mixer gets a significant responsibility to correctly perform
congestion control, identity management, manage synchronization while congestion control, source identification, manage synchronization
providing a for the application suitable media optimization. while providing the application with suitable media optimizations.
Mixers also need to be a trusted node when it comes to security as it Mixers also need to be trusted nodes when it comes to security as it
manipulates either RTP or the media itself before sending it on manipulates either RTP or the media itself before sending it on
towards the end-point(s) thus must be able to decrypt and then towards the end-point(s), thus they must be able to decrypt and then
encrypt it before sending it out. There exist one type of central encrypt it before sending it out.
node, the relay that one doesn't need to trust with the keys to the
media. The relay operates only on the IP/UDP level of the transport.
It is configured so that it would forward any RTP/RTCP packets from A
to the other participants B-D.
+---+ +---+
| | +-----------+ | |
| A |<------->| DTLS-SRTP |<------->| C |
| |<-- -->| HOST |<-- -->| |
+---+ \ / +-----------+ \ / +---+
X X
+---+ / \ +-----------+ / \ +---+
| |<-- -->| RTP |<-- -->| |
| B |<------->| RELAY |<------->| D |
| | +-----------+ | |
+---+ +---+
Figure 3: DTLS-SRTP host and RTP Relay Separated
To accomplish the security properties discussed above using a relay
one need to have a separate key handling server and also support for
distribute the different keys such as Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt]. The relay also creates a situation where
there is multiple end-points visible in the RTCP reporting and any
feedback events. Thus becoming yet another situation in addition to
Mesh where the end-point will have to have logic for merging
different requirements and preferences. This is more detail
discussed in Section 12.7.
+---+ +---+ +---+
| A |--->| B |--->| C |
+---+ +---+ +---+
Figure 4: MediaStream Forwarding
The above Figure 4 depicts a possible scenario where an WebRTC end-
point (A) sends a media stream to B. B decides to forward the media
stream to C. This can either be realized in B (WebRTC end-point)
using a simple relay functionality creating similar consideration and
implementation requirements. Another implmentation strategy in B
could be to select to transcode the media from A to C, thus breaking
most of the dependecies between A and C. In that case A is not
required to be aware of B forwarding the media to C.
12.4. SSRC Collision Detection 12.4. SSRC Collision Detection
The RTP standard [RFC3550] requires any RTP implementation to have The RTP standard [RFC3550] requires any RTP implementation to have
support for detecting and handling SSRC collisions, i.e. when two support for detecting and handling SSRC collisions, i.e., resolve the
different end-points uses the same SSRC value. This requirement conflict when two different end-points use the same SSRC value. This
applies also to WebRTC end-points. There exist several scenarios requirement also applies to WebRTC end-points. There are several
where SSRC collisions may occur. scenarios where SSRC collisions may occur.
In a point to point session where each SSRC are associated with In a point-to-point session where each SSRC is associated with either
either of the two end-points and where the main media carrying SSRC of the two end-points and where the main media carrying SSRC
identifier will be announced in the signalling there is less likely identifier will be announced in the signalling channel, a collision
to occur due to the information about used SSRCs provided by Source- is less likely to occur due to the information about used SSRCs
Specific SDP Attributes [RFC5576]. Still if both end-points starts provided by Source-Specific SDP Attributes [RFC5576]. Still if both
uses an new SSRC identifier prior to having signalled it to the peer end-points start uses an new SSRC identifier prior to having
and received acknowledgement on the signalling message there can be signalled it to the peer and received acknowledgement on the
collisions. The Source-Specific SDP Attributes [RFC5576] contains no signalling message, there can be collisions. The Source-Specific SDP
mechanism to resolve SSRC collisions or reject a end-points usage of Attributes [RFC5576] contains no mechanism to resolve SSRC collisions
an SSRC. or reject a end-points usage of an SSRC.
There could also appear unsignalled SSRCs, this may be considered a There could also appear unsignalled SSRCs. This is more likely than
bug. This is more likely than it appears as certain RTP it appears as certain RTP functions need extra SSRCs to provide
functionalities need extra SSRCs to provide functionality related to functionality related to another (the "main") SSRC, for example, SSRC
another SSRC, for example SSRC multiplexed RTP retransmission multiplexed RTP retransmission [RFC4588]. In those cases, an end-
[RFC4588]. In those cases an end-point can create a new SSRC which point can create a new SSRC that strictly doesn't need to be
strictly don't need to be announced over the signalling channel to announced over the signalling channel to function correctly on both
function correctly on both RTP and PeerConnection level. RTP and PeerConnection level.
The more likely cases for SSRC collision is that multiple end-points The more likely case for SSRC collision is that multiple end-points
in an multiparty creates new soruces and signalls those towards the in a multiparty conference create new sources and signals those
central server. In cases where the SSRC/CSRC are propogated between towards the central server. In cases where the SSRC/CSRC are
the different end-points from the central node collisions can occur. propagated between the different end-points from the central node
collisions can occur.
Another scenario is when the central node manage to connect an end- Another scenario is when the central node manages to connect an end-
points PeerConnection to another PeerConnectio the end-point it has. point's PeerConnection to another PeerConnection the end-point
Thus forming a loop where the end-point will receive its own traffic. already has, thus forming a loop where the end-point will receive its
This must be considered a bug, but still if it occurs it is important own traffic. While is is clearly considered a bug, it is important
that the end-point can handle the situation. that the end-point is able to recognise and handle the case when it
occurs.
12.5. Contributing Sources 12.5. Contributing Sources
Contributing Sources (CSRC) is a functionality in RTP header that Contributing Sources (CSRC) is a functionality in the RTP header that
enables a RTP node combing multiple sources into one to identify the allows an RTP node to combine media packets from multiple sources
sources that has gone into the combination. For WebRTC end-point the into one and to identify which sources yielded the result. For
support of contributing sources are trivial. The set of CSRC are WebRTC end-points, supporting contributing sources is trivial. The
provided for a given RTP packet. This information can then be set of CSRCs is provided in a given RTP packet. This information can
exposed towards the applications using some form of API, most likely then be exposed to the applications using some form of API, possibly
a mapping back into MediaStream identities to avoid having to expose a mapping back into WebRTC MediaStream identities to avoid having to
two namespaces and the handling of SSRC collision handling to the expose two namespaces and the handling of SSRC collision handling to
JavaScript. the JavaScript.
There are also at least one extension that is dependent on the CRSRC (tbd: should the API provide the ability to add a CSRC list to an
list being used, that is the Mixer to client audio level [RFC6465], outgoing packet? this is only useful if the sender is mixing content)
that enhances the information provided by the CSRC to actual energy
levels for audio for each contributing source. There are also at least one extension that depends on the CRSRC list
being used: the Mixer-to-client audio level [RFC6465], which enhances
the information provided by the CSRC to actual energy levels for
audio for each contributing source.
12.6. Media Synchronization 12.6. Media Synchronization
When an end-point has more than one media source being sent one need When an end-point sends media from more than one media source, it
to consider if these media source are to be synchronized. In RTP/ needs to consider if (and which of) these media sources are to be
RTCP synchronziation is provided by having a set of media streams be synchronized. In RTP/RTCP, synchronisation is provided by having a
indicated as comming from the same synchroniztion context and logical set of RTP media streams be indicated as coming from the same
end-point by using the same CNAME identifier. synchronisation context and logical end-point by using the same CNAME
identifier.
The next provision is that all media sources internal clock, i.e. The next provision is that the internal clocks of all media sources,
what drives the RTP timestamp can be correlated with a system clock i.e., what drives the RTP timestamp, can be correlated to a system
that is provided in RTCP Sender Reports encoded in an NTP format. By clock that is provided in RTCP Sender Reports encoded in an NTP
having the RTP timestamp to system clock being provided for all format. By correlating all RTP timestamps to a common system clock
sources the relation of the different media stream, also across for all sources, the timing relation of the different RTP media
multiple RTP sessions can if chosen to be synchronized. The streams, also across multiple RTP sessions can be derived at the
requirement is for the media sender to provide the information, the receiver and, if desired, the streams can be synchronized. The
receiver can chose to use it or not. requirement is for the media sender to provide the correlation
information; it is up to the receiver to use it or not.
12.7. Multiple RTP End-points 12.7. Multiple RTP End-points
A number of usages of RTP discussed here results in that an WebRTC Some usages of RTP beyond the recommend topologies result in that an
end-point sending media in an RTP session out over an PeerConnection WebRTC end-point sending media in an RTP session out over a single
will receive receiver reports from multiple RTP receiving nodes. PeerConnection will receive receiver reports from multiple RTP
Note that receiving multiple receiver reports are expected due to receivers. Note that receiving multiple receiver reports is expected
that any RTP node that has multiple SSRCs are required to report on because any RTP node that has multiple SSRCs is required to report to
the media sender. The difference here is that they are multiple the media sender. The difference here is that they are multiple
nodes, and thus will have different path characteristics. nodes, and thus will likely have different path characteristics.
The topologies relevant to WebRTC when this can occur are centralized RTP Mixers may create a situation where an end-point experiences a
relay and a end-point forwarding a media stream. Mixers are expected situation in-between a session with only two end-points and multiple
to not forward media stream reports across itself due to the end-points. Mixers are expected to not forward RTCP reports
difference in the media stream provided to different end-points which regarding RTP media streams across themselves. This is due to the
the original media source lacks information about the mixers difference in the RTP media streams provided to the different end-
manipulation. points. The original media source lacks information about a mixer's
manipulations prior to sending it the different receivers. This
setup also results in that an end-point's feedback or requests goes
to the mixer. When the mixer can't act on this by itself, it is
forced to go to the original media source to fulfill the receivers
request. This will not necessarily be explicitly visible any RTP and
RTCP traffic, but the interactions and the time to complete them will
indicate such dependencies.
Having multiple RTP nodes receive ones RTP flow and send reports and The topologies in which an end-point receives receiver reports from
feedback about it has several impacts. As previously discussed multiple other end-points are the centralized relay, multicast and an
(Section 12.3) any codec control and rate control needs to be capable end-point forwarding an RTP media stream. Having multiple RTP nodes
of merging the requirements and preferences to provide a single best receive an RTP flow and send reports and feedback about it has
according to the situation media stream. Specifically when it comes several impacts. As previously discussed (Section 12.3) any codec
to congestion control it needs to be capable of identifying the control and rate control needs to be capable of merging the
requirements and preferences to provide a single best encoding
according to the situation RTP media stream. Specifically, when it
comes to congestion control it needs to be capable of identifying the
different end-points to form independent congestion state information different end-points to form independent congestion state information
for each different path. for each different path.
Providing source authentication in multi-party is a challange. In Providing source authentication in multi-party scenarios is a
the mixer based topologies an end-points source authentication is challenge. In the mixer-based topologies, end-points source
based on verifying that media comes from the mixer by cryptographic authentication is based on, firstly, verifying that media comes from
verification and secondly trust the mixer to correctly identify any the mixer by cryptographic verification and, secondly, trust in the
source towards the end-point. In RTP sessions where multiple end- mixer to correctly identify any source towards the end-point. In RTP
points are directly visible to an end-point all end-points have sessions where multiple end-points are directly visible to an end-
knowledge about each others master keys, and can thus inject packets point, all end-points will have knowledge about each others' master
claimed to come from another end-point in the session. Any node keys, and can thus inject packets claimed to come from another end-
performing relay can perform non-cryptographic mitigation by point in the session. Any node performing relay can perform non-
preventing forwarding of packets that has SSRC fields that has cryptographic mitigation by preventing forwarding of packets that
previously come from other end-points. For cryptographic have SSRC fields that came from other end-points before. For
verification of the source SRTP will require additional security cryptographic verification of the source SRTP would require
mechanisms, like TESLA for SRTP [RFC4383]. additional security mechanisms, like TESLA for SRTP [RFC4383].
12.8. Simulcast 12.8. Simulcast
This section discusses simulcast in the meaning of providing a node, This section discusses simulcast in the meaning of providing a node,
for example a Mixer, with multiple different encoded version of the for example a Mixer, with multiple different encoded versions of the
same media source. In the WebRTC context that appears to be most same media source. In the WebRTC context, this could be accomplished
easily accomplished by establishing mutliple PeerConnection all being in two ways. One is to establish multiple PeerConnection all being
feed the same set of MediaStreams. Each PeerConnection is then feed the same set of WebRTC MediaStreams. Another method is to use
configured to deliver a particular media quality and thus media bit- multiple WebRTC MediaStreams that are differently configured when it
rate. This will work well as long as the end-point implements comes to the media parameters. This would result in that multiple
indepdentent media encoding for each PeerConnection and not share the different RTP Media Streams (SSRCs) being in used with different
encoder. Simulcast will fail if the end-point uses a common encoder encoding based on the same media source (camera, microphone).
instance to multiple PeerConnections.
Thus it should be considered to explicitly signal which of the two When intending to use simulcast it is important that this is made
implementation strategies that are desired and which will be done. explicit so that the end-points don't automatically try to optimize
At least making the application and possible the central node away the different encodings and provide a single common version.
interested in receiving simulcast of an end-points media streams to Thus, some explicit indications that the intent really is to have
be aware if it will function or not. different media encodings is likely required. It should be noted
that it might be a central node, rather than an WebRTC end-point that
would benefit from receiving simulcasted media sources.
tbd: How to perform simulcast needs to be determined and the
appropriate API or signalling for its usage needs to be defined.
12.9. Differentiated Treatment of Flows 12.9. Differentiated Treatment of Flows
There exist use cases for differentiated treatment of media streams. There are use cases for differentiated treatment of RTP media
Such differentiation can happen at several places in the system. streams. Such differentiation can happen at several places in the
First of all is the prioritization within the end-point for which system. First of all is the prioritization within the end-point
media streams that should be sent, there allocation of bit-rate out sending the media, which controls, both which RTP media streams that
of the current available aggregate as determined by the congestion will be sent, and their allocation of bit-rate out of the current
control. available aggregate as determined by the congestion control.
Secondly, the transport can prioritize a media streams. This is done Secondly, the network can prioritize packet flows, including RTP
according to three methods; media streams. Typically, differential treatment includes two steps,
the first being identifying whether an IP packet belongs to a class
which should be treated differently, the second the actual mechanism
to prioritize packets. This is done according to three methods;
Diffserv: The end-point could mark the packet with a diffserv code Diffserv: The end-point marks a packet with a diffserv code point to
point to indicate to the network how the WebRTC application and indicate to the network that the packet belongs to a particular
browser would like this particular packet treated. class.
Flow based: Prioritization of all packets belonging to a particular Flow based: Packets that shall be given a particular treatment are
media flow or RTP session by keeping them in separated UDP flows. identified using a combination of IP and port address.
Thus enabling either end-point initiated or network initiated
prioritization of the flow.
Deep Packet Inspection: A network classifier (DPI) inspects the Deep Packet Inspection: A network classifier (DPI) inspects the
packet and tries to determine if the packet represents a packet and tries to determine if the packet represents a
particular application and type that is to be prioritized. particular application and type that is to be prioritized.
With the exception of diffserv both flow based and DPI have issues With the exception of diffserv both flow based and DPI have issues
with running multiple media types and flows on a single UDP flow, with running multiple media types and flows on a single UDP flow,
especially when combined with data transport (SCTP/DTLS). DPI has especially when combined with data transport (SCTP/DTLS). DPI has
issues due to that multiple different type of flows are aggregated issues because multiple types of flows are aggregated and thus it
and thus becomes more difficult to apply analysis on. The flow based becomes more difficult to analyse them. The flow-based
differentiation will provide the same treatment to all packets within differentiation will provide the same treatment to all packets within
the flow. Thus relative prioritization is not possible. In addition the flow, i.e., relative prioritization is not possible. Moreover,
if the resources are limited it may not be possible to provide if the resources are limited it may not be possible to provide
differential treatment compared to best-effort for all the flows in a differential treatment compared to best-effort for all the flows in a
WebRTC application. WebRTC application.
When flow based differentiation is available the WebRTC application When flow-based differentiation is available the WebRTC application
needs to know about so that it can provide the separation of the needs to know about it so that it can provide the separation of the
media streams onto different UDP flows to enable a more granular RTP media streams onto different UDP flows to enable a more granular
usage of flow based differentiation. usage of flow based differentiation.
Diffserv is based on that either the end-point or a classifier can Diffserv assumes that either the end-point or a classifier can mark
mark the packets with an appropriate DSCP so the packets is treated the packets with an appropriate DSCP so that the packets are treated
according to that marking. If the end-point is to mark the traffic according to that marking. If the end-point is to mark the traffic
there exist two requirements in the WebRTC context. The first is two requirements arise in the WebRTC context: 1) The WebRTC
that the WebRTC application or browser knows which DSCP to use and application or browser has to know which DSCP to use and that it can
that it can use them on some set of media streams. Secondly the use them on some set of RTP media streams. 2) The information needs
information needs to be propagated to the operating system when to be propagated to the operating system when transmitting the
transmitting the packet. packet.
Open Issue: How will the WebRTC application and/or browser know that tbd: The model for providing differentiated treatment needs to be
differentiated treatment is desired and available and ensure that it evolved. This includes:
gets the information required to correctly configure the WebRTC
multimedia conference. 1. How the application can prioritize MediaStreamTracks differently
in the API
2. How the browser or application determine availability of
transport differentiation
3. How to learn about any configuration information for transport
differentiation, such as DSCPs.
13. IANA Considerations 13. IANA Considerations
This memo makes no request of IANA. This memo makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. RFC.
14. Security Considerations 14. Security Considerations
RTP and its various extensions each have their own security RTP and its various extensions each have their own security
considerations. These should be taken into account when considering considerations. These should be taken into account when considering
the security properties of the complete suite. We currently don't the security properties of the complete suite. We currently don't
think this suite creates any additional security issues or think this suite creates any additional security issues or
properties. The use of SRTP [RFC3711] will provide protection or properties. The use of SRTP [RFC3711] will provide protection or
mitigation against all the fundamental issues by offering mitigation against most of the fundamental issues by offering
confidentiality, integrity and partial source authentication. A confidentiality, integrity and partial source authentication. A
mandatory to implement media security solution will be required to be mandatory to implement media security solution will be required to be
picked. We currently don't discuss the key-management aspect of SRTP picked. We currently don't discuss the key-management aspect of SRTP
in this memo, that needs to be done taking the WebRTC communication in this memo, that needs to be done taking the WebRTC communication
model into account. model into account.
The guidelines in [I-D.ietf-avtcore-srtp-vbr-audio] apply when using Privacy concerns are under discussion and the generation of non-
variable bit rate (VBR) audio codecs, for example Opus or the Mixer trackable CNAMEs are under discussion.
audio level header extensions.
The guidelines in [RFC6562] apply when using variable bit rate (VBR)
audio codecs, for example Opus or the Mixer audio level header
extensions.
Security considerations for the WebRTC work are discussed in Security considerations for the WebRTC work are discussed in
[I-D.ietf-rtcweb-security]. [I-D.ietf-rtcweb-security].
15. Acknowledgements 15. 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 and Cullen Jennings for valuable feedback.
16. References 16. References
skipping to change at page 32, line 47 skipping to change at page 32, line 22
Using Session Description Protocol (SDP) Port Numbers", Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011. progress), October 2011.
[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-srtp-encrypted-header-ext-01 (work in draft-ietf-avtcore-srtp-encrypted-header-ext-01 (work in
progress), October 2011. progress), October 2011.
[I-D.ietf-avtcore-srtp-vbr-audio] [I-D.ietf-avtext-multiple-clock-rates]
Perkins, C. and J. Valin, "Guidelines for the use of Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Variable Bit Rate Audio with Secure RTP", Clock Rates in an RTP Session",
draft-ietf-avtcore-srtp-vbr-audio-04 (work in progress), draft-ietf-avtext-multiple-clock-rates-05 (work in
December 2011. progress), May 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-03 (work based Applications", draft-ietf-rtcweb-overview-04 (work
in progress), March 2012. in progress), June 2012.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web", Rescorla, E., "Security Considerations for RTC-Web",
draft-ietf-rtcweb-security-02 (work in progress), draft-ietf-rtcweb-security-03 (work in progress),
March 2012. June 2012.
[I-D.jesup-rtp-congestion-reqs]
Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media",
draft-jesup-rtp-congestion-reqs-00 (work in progress),
March 2012.
[I-D.lennox-rtcweb-rtp-media-type-mux] [I-D.lennox-rtcweb-rtp-media-type-mux]
Rosenberg, J. and J. Lennox, "Multiplexing Multiple Media Rosenberg, J. and J. Lennox, "Multiplexing Multiple Media
Types In a Single Real-Time Transport Protocol (RTP) Types In a Single Real-Time Transport Protocol (RTP)
Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work
in progress), October 2011. in progress), October 2011.
[I-D.perkins-avtcore-rtp-circuit-breakers] [I-D.perkins-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "RTP Congestion Control: Circuit Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
Breakers for Unicast Sessions", Breakers for Unicast Sessions",
draft-perkins-avtcore-rtp-circuit-breakers-00 (work in draft-perkins-avtcore-rtp-circuit-breakers-00 (work in
progress), March 2012. progress), March 2012.
[I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., and C. Perkins, "RTP
Multiplexing Architecture",
draft-westerlund-avtcore-multiplex-architecture-01 (work
in progress), March 2012.
[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-transport-multiplexing-02 (work draft-westerlund-avtcore-transport-multiplexing-02 (work
in progress), March 2012. in progress), March 2012.
[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
skipping to change at page 34, line 37 skipping to change at page 33, line 48
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
with Feedback (AVPF)", RFC 5104, February 2008. with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008. (RTP/SAVPF)", RFC 5124, February 2008.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008. Header Extensions", RFC 5285, July 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, April 2009.
skipping to change at page 35, line 24 skipping to change at page 34, line 34
(CNAMEs)", RFC 6222, April 2011. (CNAMEs)", RFC 6222, April 2011.
[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
Variable Bit Rate Audio with Secure RTP", RFC 6562,
March 2012.
16.2. Informative References 16.2. 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-rtcweb-msid-02 (work in progress), draft-alvestrand-rtcweb-msid-02 (work in progress),
May 2012. May 2012.
[I-D.begen-mmusic-redundancy-grouping]
Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
Semantics in the Session Description Protocol",
draft-begen-mmusic-redundancy-grouping-03 (work in
progress), March 2012.
[I-D.cbran-rtcweb-data]
Bran, C. and C. Jennings, "RTC-Web Non-Media Data
Transport Requirements", draft-cbran-rtcweb-data-00 (work
in progress), July 2011.
[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-fecframe-framework] [I-D.ietf-rtcweb-use-cases-and-requirements]
Watson, M., Begen, A., and V. Roca, "Forward Error Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Correction (FEC) Framework", Time Communication Use-cases and Requirements",
draft-ietf-fecframe-framework-15 (work in progress), draft-ietf-rtcweb-use-cases-and-requirements-09 (work in
June 2011. progress), June 2012.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., [I-D.jesup-rtp-congestion-reqs]
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- Jesup, R. and H. Alvestrand, "Congestion Control
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, Requirements For Real Time Media",
September 1997. draft-jesup-rtp-congestion-reqs-00 (work in progress),
March 2012.
[I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., and C. Perkins, "RTP
Multiplexing Architecture",
draft-westerlund-avtcore-multiplex-architecture-01 (work
in progress), March 2012.
[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 2006. February 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 2007. April 2007.
[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"RTP Payload Format and File Storage Format for the
Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs", RFC 4867, April 2007.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008. January 2008.
[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 5348, September 2008. RFC 5348, September 2008.
[RFC5404] Westerlund, M. and I. Johansson, "RTP Payload Format for
G.719", RFC 5404, January 2009.
[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 37, line 20 skipping to change at page 36, line 23
RTP supports both unicast and group communication, with participants RTP supports both unicast and group communication, with participants
being connected using wide range of transport-layer topologies. Some being connected using wide range of transport-layer topologies. Some
of these topologies involve only the end-points, while others use RTP of these topologies involve only the end-points, while others use RTP
translators and mixers to provide in-network processing. Properties translators and mixers to provide in-network processing. Properties
of some RTP topologies are discussed in [RFC5117], and we further of some RTP topologies are discussed in [RFC5117], and we further
describe those expected to be useful for WebRTC in the following. We describe those expected to be useful for WebRTC in the following. We
also goes into important RTP session aspects that the topology or also goes into important RTP session aspects that the topology or
implementation variant can place on a WebRTC end-point. implementation variant can place on a WebRTC end-point.
This section includes RTP topologies beyond the recommended ones.
This in an attempt to highlight the differencies and the in many case
small differences in implementation to support a larger set of
possible topologies.
A.1. Point to Point A.1. Point to Point
The point-to-point RTP topology (Figure 5) is the simplest scenario The point-to-point RTP topology (Figure 3) is the simplest scenario
for WebRTC applications. This is going to be very common for user to for WebRTC applications. This is going to be very common for user to
user calls. user calls.
+---+ +---+ +---+ +---+
| A |<------->| B | | A |<------->| B |
+---+ +---+ +---+ +---+
Figure 5: Point to Point Figure 3: Point to Point
This being the basic one lets use the topology to high-light a couple This being the basic one lets use the topology to high-light a couple
of details that are common for all RTP usage in the WebRTC context. of details that are common for all RTP usage in the WebRTC context.
First is the intention to multiplex RTP and RTCP over the same UDP- First is the intention to multiplex RTP and RTCP over the same UDP-
flow. Secondly is the question of using only a single RTP session or flow. Secondly is the question of using only a single RTP session or
one per media type for legacy interoperability. Thirdly is the one per media type for legacy interoperability. Thirdly is the
question of using multiple sender sources (SSRCs) per end-point. question of using multiple sender sources (SSRCs) per end-point.
Historically, RTP and RTCP have been run on separate UDP ports. With Historically, RTP and RTCP have been run on separate UDP ports. With
the increased use of Network Address/Port Translation (NAPT) this has the increased use of Network Address/Port Translation (NAPT) this has
skipping to change at page 38, line 13 skipping to change at page 37, line 21
(e.g., audio and video), then each type media can be sent as a (e.g., audio and video), then each type media can be sent as a
separate RTP session using a different 5-tuple, allowing for separate separate RTP session using a different 5-tuple, allowing for separate
transport level treatment of each type of media. Alternatively, all transport level treatment of each type of media. Alternatively, all
types of media can be multiplexed onto a single 5-tuple as a single types of media can be multiplexed onto a single 5-tuple as a single
RTP session, or as several RTP sessions if using a demultiplexing RTP session, or as several RTP sessions if using a demultiplexing
shim. Multiplexing different types of media onto a single 5-tuple shim. Multiplexing different types of media onto a single 5-tuple
places some limitations on how RTP is used, as described in "RTP places some limitations on how RTP is used, as described in "RTP
Multiplexing Architecture" Multiplexing Architecture"
[I-D.westerlund-avtcore-multiplex-architecture]. It is not expected [I-D.westerlund-avtcore-multiplex-architecture]. It is not expected
that these limitations will significantly affect the scenarios that these limitations will significantly affect the scenarios
targetted by WebRTC, but they may impact interoperability with legacy targeted by WebRTC, but they may impact interoperability with legacy
systems. systems.
An RTP session have good support for simultanously transport multiple An RTP session have good support for simultanously transport multiple
media sources. Each media source uses an unique SSRC identifier and media sources. Each media source uses an unique SSRC identifier and
each SSRC has independent RTP sequence number and timestamp spaces. each SSRC has independent RTP sequence number and timestamp spaces.
This is being utilized in WebRTC for several cases. One is to enable This is being utilized in WebRTC for several cases. One is to enable
multiple media sources of the same type, an end-point that has two multiple media sources of the same type, an end-point that has two
video cameras can potentially transmitt video from both to its video cameras can potentially transmitt video from both to its
peer(s). Another usage is when a single RTP session is being used peer(s). Another usage is when a single RTP session is being used
for both multiple media types, thus an end-point can transmit both for both multiple media types, thus an end-point can transmit both
skipping to change at page 39, line 28 skipping to change at page 38, line 28
| | | | +-Video-| |-Video-+ | | | | | | | | +-Video-| |-Video-+ | | | |
| | | | | AV1|---------------->| | | | | | | | | | | AV1|---------------->| | | | | |
| | | | | AV2|---------------->| | | | | | | | | | | AV2|---------------->| | | | | |
| | | | | |<----------------|BV1 | | | | | | | | | | |<----------------|BV1 | | | | |
| | | | +-------| |-------+ | | | | | | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | | | | | +---------| |---------+ | | |
| | +-----------| |-----------+ | | | | +-----------| |-----------+ | |
| +-------------| |-------------+ | | +-------------| |-------------+ |
+---------------+ +---------------+ +---------------+ +---------------+
Figure 6: Point to Point: Multiple RTP sessions Figure 4: Point to Point: Multiple RTP sessions
As can be seen above in the Point to Point: Multiple RTP sessions As can be seen above in the Point to Point: Multiple RTP sessions
(Figure 6) the single Peer Connection contains two RTP sessions over (Figure 4) the single Peer Connection contains two RTP sessions over
different UDP flows UDP 1 and UDP 2, i.e. their 5-tuples will be different UDP flows UDP 1 and UDP 2, i.e. their 5-tuples will be
different, normally on source and destination ports. The first RTP different, normally on source and destination ports. The first RTP
session (RTP1) carries audio, one stream in each direction AA1 and session (RTP1) carries audio, one stream in each direction AA1 and
BA1. The second RTP session contains two video streams from A (AV1 BA1. The second RTP session contains two video streams from A (AV1
and AV2) and one from B to A (BV1). and AV2) and one from B to A (BV1).
+-A-------------+ +-B-------------+ +-A-------------+ +-B-------------+
| +-PeerC1------| |-PeerC1------+ | | +-PeerC1------| |-PeerC1------+ |
| | +-UDP1------| |-UDP1------+ | | | | +-UDP1------| |-UDP1------+ | |
| | | +-RTP1----| |-RTP1----+ | | | | | | +-RTP1----| |-RTP1----+ | | |
skipping to change at page 40, line 24 skipping to change at page 39, line 24
| | | | +-Video-| |-Video-+ | | | | | | | | +-Video-| |-Video-+ | | | |
| | | | | AV1|---------------->| | | | | | | | | | | AV1|---------------->| | | | | |
| | | | | AV2|---------------->| | | | | | | | | | | AV2|---------------->| | | | | |
| | | | | |<----------------|BV1 | | | | | | | | | | |<----------------|BV1 | | | | |
| | | | +-------| |-------+ | | | | | | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | | | | | +---------| |---------+ | | |
| | +-----------| |-----------+ | | | | +-----------| |-----------+ | |
| +-------------| |-------------+ | | +-------------| |-------------+ |
+---------------+ +---------------+ +---------------+ +---------------+
Figure 7: Point to Point: Single RTP session. Figure 5: Point to Point: Single RTP session.
In (Figure 7) there is only a single UDP flow and RTP session (RTP1). In (Figure 5) there is only a single UDP flow and RTP session (RTP1).
This RTP session carries a total of five (5) media streams (SSRCs). This RTP session carries a total of five (5) RTP media streams
From A to B there is Audio (AA1) and two video (AV1 and AV2). From B (SSRCs). From A to B there is Audio (AA1) and two video (AV1 and
to A there is Audio (BA1) and Video (BV1). AV2). From B to A there is Audio (BA1) and Video (BV1).
A.2. Multi-Unicast (Mesh) A.2. Multi-Unicast (Mesh)
For small multiparty calls, it is practical to set up a multi-unicast For small multiparty calls, it is practical to set up a multi-unicast
topology (Figure 8); unfortunately not discussed in the RTP topology (Figure 6); unfortunately not discussed in the RTP
Topologies RFC [RFC5117]. In this topology, each participant sends Topologies RFC [RFC5117]. In this topology, each participant sends
individual unicast RTP/UDP/IP flows to each of the other participants individual unicast RTP/UDP/IP flows to each of the other participants
using independent PeerConnections in a full mesh. using independent PeerConnections in a full mesh.
+---+ +---+ +---+ +---+
| A |<---->| B | | A |<---->| B |
+---+ +---+ +---+ +---+
^ ^ ^ ^
\ / \ /
\ / \ /
v v v v
+---+ +---+
| C | | C |
+---+ +---+
Figure 8: Multi-unicast Figure 6: Multi-unicast
This topology has the benefit of not requiring central nodes. The This topology has the benefit of not requiring central nodes. The
downside is that it increases the used bandwidth at each sender by downside is that it increases the used bandwidth at each sender by
requiring one copy of the media streams for each participant that are requiring one copy of the RTP media streams for each participant that
part of the same session beyond the sender itself. Hence, this are part of the same session beyond the sender itself. Hence, this
topology is limited to scenarios with few participants unless the topology is limited to scenarios with few participants unless the
media is very low bandwidth. The multi-unicast topology could be media is very low bandwidth. The multi-unicast topology could be
implemented as a single RTP session, spanning multiple peer-to-peer implemented as a single RTP session, spanning multiple peer-to-peer
transport layer connections, or as several pairwise RTP sessions, one transport layer connections, or as several pairwise RTP sessions, one
between each pair of peers. To maintain a coherent mapping between between each pair of peers. To maintain a coherent mapping between
the relation between RTP sessions and PeerConnections we recommend the relation between RTP sessions and PeerConnections we recommend
that one implements this as individual RTP sessions. The only that one implements this as individual RTP sessions. The only
downside is that end-point A will not learn of the quality of any downside is that end-point A will not learn of the quality of any
transmission happening between B and C based on RTCP. This has not transmission happening between B and C based on RTCP. This has not
been seen as a significant downside as now one has yet seen a need been seen as a significant downside as now one has yet seen a need
skipping to change at page 41, line 49 skipping to change at page 40, line 49
| | | | +-RTP2----| |-RTP2----+ | | | | | | | +-RTP2----| |-RTP2----+ | | |
| | +----+ | | | +-Audio-| |-Audio-+ | | | | | | +----+ | | | +-Audio-| |-Audio-+ | | | |
| +->|ENC2|--+-+-+-+--->AA2|------------->| | | | | | | +->|ENC2|--+-+-+-+--->AA2|------------->| | | | | |
| +----+ | | | | |<-------------|CA1 | | | | | | +----+ | | | | |<-------------|CA1 | | | | |
| | | | +-------| |-------+ | | | | | | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | | | | | +---------| |---------+ | | |
| | +-----------| |-----------+ | | | | +-----------| |-----------+ | |
| +-------------| |-------------+ | | +-------------| |-------------+ |
+--------------------------+ +---------------+ +--------------------------+ +---------------+
Figure 9: Session strcuture for Multi-Unicast Setup Figure 7: Session structure for Multi-Unicast Setup
Lets review how the RTP sessions looks from A's perspective by Lets review how the RTP sessions looks from A's perspective by
considering both how the media is a handled and what PeerConnections considering both how the media is a handled and what PeerConnections
and RTP sessions that are setup in Figure 9. A's microphone is and RTP sessions that are setup in Figure 7. A's microphone is
captured and the digital audio can then be feed into two different captured and the digital audio can then be feed into two different
encoder instances each beeing associated with two different encoder instances each beeing associated with two different
PeerConnections (PeerC1 and PeerC2) each containing independent RTP PeerConnections (PeerC1 and PeerC2) each containing independent RTP
sessions (RTP1 and RTP2). The SSRCs in each RTP session will be sessions (RTP1 and RTP2). The SSRCs in each RTP session will be
completely independent and the media bit-rate produced by the encoder completely independent and the media bit-rate produced by the encoder
can also be tuned to address any congestion control requirements can also be tuned to address any congestion control requirements
between A and B differently then for the path A to C. between A and B differently then for the path A to C.
For media encodings which are more resource consuming, like video, For media encodings which are more resource consuming, like video,
one could expect that it will be common that end-points that are one could expect that it will be common that end-points that are
resource costrained will use a different implementation strategy resource costrained will use a different implementation strategy
where the encoder is shared between the different PeerConnections as where the encoder is shared between the different PeerConnections as
shown below Figure 10. shown below Figure 8.
+-A----------------------+ +-B-------------+ +-A----------------------+ +-B-------------+
|+---+ | | | |+---+ | | |
||CAM| +-PeerC1------| |-PeerC1------+ | ||CAM| +-PeerC1------| |-PeerC1------+ |
|+---+ | +-UDP1------| |-UDP1------+ | | |+---+ | +-UDP1------| |-UDP1------+ | |
| | | | +-RTP1----| |-RTP1----+ | | | | | | | +-RTP1----| |-RTP1----+ | | |
| V | | | +-Video-| |-Video-+ | | | | | V | | | +-Video-| |-Video-+ | | | |
|+----+ | | | | |<----------------|BV1 | | | | | |+----+ | | | | |<----------------|BV1 | | | | |
||ENC |----+-+-+-+--->AV1|---------------->| | | | | | ||ENC |----+-+-+-+--->AV1|---------------->| | | | | |
|+----+ | | | +-------| |-------+ | | | | |+----+ | | | +-------| |-------+ | | | |
| | | | +---------| |---------+ | | | | | | | +---------| |---------+ | | |
skipping to change at page 42, line 46 skipping to change at page 41, line 46
| | | | +-RTP2----| |-RTP2----+ | | | | | | | +-RTP2----| |-RTP2----+ | | |
| | | | | +-Video-| |-Video-+ | | | | | | | | | +-Video-| |-Video-+ | | | |
| +-------+-+-+-+--->AV2|---------------->| | | | | | | +-------+-+-+-+--->AV2|---------------->| | | | | |
| | | | | |<----------------|CV1 | | | | | | | | | | |<----------------|CV1 | | | | |
| | | | +-------| |-------+ | | | | | | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | | | | | +---------| |---------+ | | |
| | +-----------| |-----------+ | | | | +-----------| |-----------+ | |
| +-------------| |-------------+ | | +-------------| |-------------+ |
+------------------------+ +---------------+ +------------------------+ +---------------+
Figure 10: Single Encoder Multi-Unicast Setup Figure 8: Single Encoder Multi-Unicast Setup
This will clearly save resources consumed by encoding but does This will clearly save resources consumed by encoding but does
introduce the need for the end-point A to make decisions on how it introduce the need for the end-point A to make decisions on how it
encodes the media so it suites delivery to both B and C. This is not encodes the media so it suites delivery to both B and C. This is not
limited to congestion control, also prefered resolution to receive limited to congestion control, also prefered resolution to receive
based on dispaly area available is another aspect requiring based on dispaly area available is another aspect requiring
consideration. The need for this type of descion logic does arise in consideration. The need for this type of descion logic does arise in
several different topologies and implementation. several different topologies and implementation.
A.3. Mixer Based A.3. Mixer Based
An mixer (Figure 11) is a centralised point that selects or mixes An mixer (Figure 9) is a centralised point that selects or mixes
content in a conference to optimise the RTP session so that each end- content in a conference to optimise the RTP session so that each end-
point only needs connect to one entity, the mixer. The mixer can point only needs connect to one entity, the mixer. The mixer can
also reduce the bit-rate needed from the mixer down to a conference also reduce the bit-rate needed from the mixer down to a conference
participants as the media sent from the mixer to the end-point can be participants as the media sent from the mixer to the end-point can be
optimised in different ways. These optimisations include methods optimised in different ways. These optimisations include methods
like only choosing media from the currently most active speaker or like only choosing media from the currently most active speaker or
mixing together audio so that only one audio stream is required in mixing together audio so that only one audio stream is required in
stead of 3 in the depicted scenario (Figure 11). stead of 3 in the depicted scenario (Figure 9).
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Mixer | | Mixer |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 11: RTP Mixer with Only Unicast Paths Figure 9: RTP Mixer with Only Unicast Paths
Mixers has two downsides, the first is that the mixer must be a Mixers has two downsides, the first is that the mixer must be a
trusted node as they either performs media operations or at least trusted node as they either performs media operations or at least
repacketize the media. Both type of operations requires when using repacketize the media. Both type of operations requires when using
SRTP that the mixer verifies integrity, decrypts the content, perform SRTP that the mixer verifies integrity, decrypts the content, perform
its operation and form new RTP packets, encrypts and integegrity its operation and form new RTP packets, encrypts and integegrity
protect them. This applies to all types of mixers described below. protect them. This applies to all types of mixers described below.
The second downside is that all these operations and optimization of The second downside is that all these operations and optimization of
the session requires processing. How much depends on the the session requires processing. How much depends on the
implementation as will become evident below. implementation as will become evident below.
The implementation of an mixer can take several different forms and The implementation of an mixer can take several different forms and
we will discuss the main themes available that doesn't break RTP. we will discuss the main themes available that doesn't break RTP.
Please note that a Mixer could also contain translator Please note that a Mixer could also contain translator
functionalities, like a media transcoder to adjust the media bit-rate functionalities, like a media transcoder to adjust the media bit-rate
or codec used on a particular media stream. or codec used on a particular RTP media stream.
A.3.1. Media Mixing A.3.1. Media Mixing
This type of mixer is one which clearly can be called RTP mixer is This type of mixer is one which clearly can be called RTP mixer is
likely the one that most thinks of when they hear the term mixer. likely the one that most thinks of when they hear the term mixer.
Its basic patter of operation is that it will receive the different Its basic patter of operation is that it will receive the different
participants media stream. Select which that are to be included in a participants RTP media stream. Select which that are to be included
media domain mix of the incomming media streams. Then create a in a media domain mix of the incomming RTP media streams. Then
single outgoing stream from this mix. create a single outgoing stream from this mix.
Audio mixing is straight forward and commonly possible to do for a Audio mixing is straight forward and commonly possible to do for a
number of participants. Lets assume that you want to mix N number of number of participants. Lets assume that you want to mix N number of
streams from different participants. Then the mixer need to perform streams from different participants. Then the mixer need to perform
N decodings. Then it needs to produce N or N+1 mixes, the reasons N decodings. Then it needs to produce N or N+1 mixes, the reasons
that different mixes are needed are so that each contributing source that different mixes are needed are so that each contributing source
get a mix which don't contain themselves, as this would result in an get a mix which don't contain themselves, as this would result in an
echo. When N is lower than the number of all participants one may echo. When N is lower than the number of all participants one may
produce a Mix of all N streams for the group that are curently not produce a Mix of all N streams for the group that are curently not
included in the mix, thus N+1 mixes. These audio streams are then included in the mix, thus N+1 mixes. These audio streams are then
skipping to change at page 44, line 33 skipping to change at page 43, line 33
video streams can be done. In fact it can be done in a number of video streams can be done. In fact it can be done in a number of
ways, tiling the different streams creating a chessboard, selecting ways, tiling the different streams creating a chessboard, selecting
someone as more important and showing them large and a number of someone as more important and showing them large and a number of
other sources as smaller is another. Also here one commonly need to other sources as smaller is another. Also here one commonly need to
produce a number of different compositions so that the contributing produce a number of different compositions so that the contributing
part doesn't need to see themselves. Then the mixer re-encodes the part doesn't need to see themselves. Then the mixer re-encodes the
created video stream, RTP packetize it and send it out created video stream, RTP packetize it and send it out
The problem with media mixing is that it both consume large amount of The problem with media mixing is that it both consume large amount of
media processing and encoding resources. The second is the quality media processing and encoding resources. The second is the quality
degradation created by decoding and re-encoding the media stream. degradation created by decoding and re-encoding the RTP media stream.
Its advantage is that it is quite simplistic for the clients to Its advantage is that it is quite simplistic for the clients to
handle as they don't need to handle local mixing and composition. handle as they don't need to handle local mixing and composition.
+-A-------------+ +-MIXER--------------------------+ +-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ | | +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | | | | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ | | | | +-RTP1----| |-RTP1------+ | | +-----+ |
| | | | +-Audio-| |-Audio---+ | | | +---+ | | | | | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | AA1|------------>|---------+-+-+-+-|DEC|->| | | | | | | | AA1|------------>|---------+-+-+-+-|DEC|->| | |
| | | | | |<------------|MA1 <----+ | | | +---+ | | | | | | | | |<------------|MA1 <----+ | | | +---+ | | |
skipping to change at page 45, line 46 skipping to change at page 44, line 46
| | | | +-Audio-| |-Audio---+ | | | +---+ | | | | | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | CA1|------------>|---------+-+-+-+-|DEC|->| | | | | | | | CA1|------------>|---------+-+-+-+-|DEC|->| | |
| | | | | |<------------|MA3 <----+ | | | +---+ | | | | | | | | |<------------|MA3 <----+ | | | +---+ | | |
| | | | +-------| |(BA1+CA1)|\| | | +---+ | | | | | | | +-------| |(BA1+CA1)|\| | | +---+ | | |
| | | +---------| |---------+ +-+-+-|ENC|<-| A+B | | | | | +---------| |---------+ +-+-+-|ENC|<-| A+B | |
| | +-----------| |-----------+ | | +---+ | | | | | +-----------| |-----------+ | | +---+ | | |
| +-------------| |-------------+ | +-----+ | | +-------------| |-------------+ | +-----+ |
+---------------+ |---------------+ | +---------------+ |---------------+ |
+--------------------------------+ +--------------------------------+
Figure 12: Session and SSRC details for Media Mixer Figure 10: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be very straight forward as From an RTP perspective media mixing can be very straight forward as
can be seen in Figure 12. The mixer present one SSRC towards the can be seen in Figure 10. The mixer present one SSRC towards the
peer client, e.g. MA1 to Peer A, which is the media mix of the other peer client, e.g. MA1 to Peer A, which is the media mix of the other
particpants. As each peer receives a different version produced by particpants. As each peer receives a different version produced by
the mixer there are no actual relation between the different RTP the mixer there are no actual relation between the different RTP
sessions in the actual media or the transport level information. sessions in the actual media or the transport level information.
There is however one connection between RTP1-RTP3 in this figure. It There is however one connection between RTP1-RTP3 in this figure. It
has to do with the SSRC space and the identity information. When A has to do with the SSRC space and the identity information. When A
receives the MA1 stream which is a combination of BA1 and CA1 streams receives the MA1 stream which is a combination of BA1 and CA1 streams
in the other PeerConnections RTP could enable the mixer to include in the other PeerConnections RTP could enable the mixer to include
CSRC information in the MA1 stream to identify the contributing CSRC information in the MA1 stream to identify the contributing
source BA1 and CA1. source BA1 and CA1.
skipping to change at page 46, line 28 skipping to change at page 45, line 28
the different legs. For the above situation commonly nothing more the different legs. For the above situation commonly nothing more
than the Source Description (SDES) information and RTCP BYE for CSRC than the Source Description (SDES) information and RTCP BYE for CSRC
need to be exposed. The main goal would be to enable the correct need to be exposed. The main goal would be to enable the correct
binding against the application logic and other information sources. binding against the application logic and other information sources.
This also enables loop detection in the RTP session. This also enables loop detection in the RTP session.
A.3.1.1. RTP Session Termination A.3.1.1. RTP Session Termination
There exist an possible implementation choice to have the RTP There exist an possible implementation choice to have the RTP
sessions being separated between the different legs in the multi- sessions being separated between the different legs in the multi-
party communication session and only generate media streams in each party communication session and only generate RTP media streams in
without carrying on RTP/RTCP level any identity information about the each without carrying on RTP/RTCP level any identity information
contributing sources. This removes both the functionaltiy that CSRC about the contributing sources. This removes both the functionaltiy
can provide and the possibility to use any extensions that build on that CSRC can provide and the possibility to use any extensions that
CSRC and the loop detection. It may appear a simplification if SSRC build on CSRC and the loop detection. It may appear a simplification
collision would occur between two different end-points as they can be if SSRC collision would occur between two different end-points as
avoide to be resolved and instead remapped between the independent they can be avoide to be resolved and instead remapped between the
sessions if at all exposed. However, SSRC/CSRC remapping independent sessions if at all exposed. However, SSRC/CSRC remapping
requiresthat SSRC/CSRC are never exposed to the WebRTC javascript requiresthat SSRC/CSRC are never exposed to the WebRTC javascript
client to use as reference. This as they only have local importance client to use as reference. This as they only have local importance
if they are used on a multi-party session scope the result would be if they are used on a multi-party session scope the result would be
missreferencing. Also SSRC collision handling will still be needed missreferencing. Also SSRC collision handling will still be needed
as it may occur between the mixer and the end-point. as it may occur between the mixer and the end-point.
Session termination may appear to resolve some issues, it however Session termination may appear to resolve some issues, it however
creates other issues that needs resolving, like loop detection, creates other issues that needs resolving, like loop detection,
identification of contributing sources and the need to handle mapped identification of contributing sources and the need to handle mapped
identities and ensure that the right one is used towards the right identities and ensure that the right one is used towards the right
skipping to change at page 47, line 9 skipping to change at page 46, line 9
A.3.2. Media Switching A.3.2. Media Switching
An RTP Mixer based on media switching avoids the media decoding and An RTP Mixer based on media switching avoids the media decoding and
encoding cycle in the mixer, but not the decryption and re-encryption encoding cycle in the mixer, but not the decryption and re-encryption
cycle as one rewrites RTP headers. This both reduces the amount of cycle as one rewrites RTP headers. This both reduces the amount of
computational resources needed in the mixer and increases the media computational resources needed in the mixer and increases the media
quality per transmitted bit. This is achieve by letting the mixer quality per transmitted bit. This is achieve by letting the mixer
have a number of SSRCs that represents conceptual or functional have a number of SSRCs that represents conceptual or functional
streams the mixer produces. These streams are created by selecting streams the mixer produces. These streams are created by selecting
media from one of the by the mixer received media streams and forward media from one of the by the mixer received RTP media streams and
the media using the mixers own SSRCs. The mixer can then switch forward the media using the mixers own SSRCs. The mixer can then
between available sources if that is required by the concept for the switch between available sources if that is required by the concept
source, like currently active speaker. for the source, like currently active speaker.
To achieve a coherent RTP media stream from the mixer's SSRC the To achieve a coherent RTP media stream from the mixer's SSRC the
mixer is forced to rewrite the incoming RTP packet's header. First mixer is forced to rewrite the incoming RTP packet's header. First
the SSRC field must be set to the value of the Mixer's SSRC. the SSRC field must be set to the value of the Mixer's SSRC.
Secondly, the sequence number must be the next in the sequence of Secondly, the sequence number must be the next in the sequence of
outgoing packets it sent. Thirdly the RTP timestamp value needs to outgoing packets it sent. Thirdly the RTP timestamp value needs to
be adjusted using an offset that changes each time one switch media be adjusted using an offset that changes each time one switch media
source. Finally depending on the negotiation the RTP payload type source. Finally depending on the negotiation the RTP payload type
value representing this particular RTP payload configuration may have value representing this particular RTP payload configuration may have
to be changed if the different PeerConnections have not arrived on to be changed if the different PeerConnections have not arrived on
skipping to change at page 48, line 48 skipping to change at page 47, line 48
| | | | +-Video-| |-Video---+ | | | | | | | | | | +-Video-| |-Video---+ | | | | | |
| | | | | CV1|------------>|---------+-+-+-+------->| | | | | | | | CV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|MV11 <---+-+-+-+-AV1----| | | | | | | | |<------------|MV11 <---+-+-+-+-AV1----| | |
| | | | | |<------------|MV12 <---+-+-+-+-EV1----| | | | | | | | |<------------|MV12 <---+-+-+-+-EV1----| | |
| | | | +-------| |---------+ | | | | | | | | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | | | | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | +-----+ | | | +-----------| |-------------+ | +-----+ |
| +-------------| |---------------+ | | +-------------| |---------------+ |
+---------------+ +--------------------------------+ +---------------+ +--------------------------------+
Figure 13: Media Switching RTP Mixer Figure 11: Media Switching RTP Mixer
The Media Switching RTP mixer can similar to the Media Mixing one The Media Switching RTP mixer can similar to the Media Mixing one
reduce the bit-rate needed towards the different peers by selecting reduce the bit-rate needed towards the different peers by selecting
and switching in a sub-set of media streams out of the ones it and switching in a sub-set of RTP media streams out of the ones it
receives from the conference participations. receives from the conference participations.
To ensure that a media receiver can correctly decode the media stream To ensure that a media receiver can correctly decode the RTP media
after a switch, it becomes necessary to ensure for state saving stream after a switch, it becomes necessary to ensure for state
codecs that they start from default state at the point of switching. saving codecs that they start from default state at the point of
Thus one common tool for video is to request that the encoding switching. Thus one common tool for video is to request that the
creates an intra picture, something that isn't dependent on earlier encoding creates an intra picture, something that isn't dependent on
state. This can be done using Full Intra Request RTCP codec control earlier state. This can be done using Full Intra Request RTCP codec
message as discussed in Section 5.1.1. control message as discussed in Section 5.1.1.
Also in this type of mixer one could consider to terminate the RTP Also in this type of mixer one could consider to terminate the RTP
sessions fully between the different PeerConnection. The same sessions fully between the different PeerConnection. The same
arguments and conisderations as discussed in Appendix A.3.1.1 applies arguments and conisderations as discussed in Appendix A.3.1.1 applies
here. here.
A.3.3. Media Projecting A.3.3. Media Projecting
Another method for handling media in the RTP mixer is to project all Another method for handling media in the RTP mixer is to project all
potential sources (SSRCs) into a per end-point independent RTP potential sources (SSRCs) into a per end-point independent RTP
skipping to change at page 51, line 4 skipping to change at page 50, line 4
| | | | +-Video-| |-Video---+ | | | | | | | | | | +-Video-| |-Video---+ | | | | | |
| | | | | CV1|------------>|---------+-+-+-+------->| | | | | | | | CV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|AV1 <----+-+-+-+--------| | | | | | | | |<------------|AV1 <----+-+-+-+--------| | |
| | | | | | : : : |: : : : : : : : : : :| | | | | | | | | : : : |: : : : : : : : : : :| | |
| | | | | |<------------|EV1 <----+-+-+-+--------| | | | | | | | |<------------|EV1 <----+-+-+-+--------| | |
| | | | +-------| |---------+ | | | | | | | | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | | | | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | +-----+ | | | +-----------| |-------------+ | +-----+ |
| +-------------| |---------------+ | | +-------------| |---------------+ |
+---------------+ +--------------------------------+ +---------------+ +--------------------------------+
Figure 14: Media Projecting Mixer Figure 12: Media Projecting Mixer
So in this six participant conference depicted above in (Figure 14) So in this six participant conference depicted above in (Figure 12)
one can see that end-point A will in this case be aware of 5 incoming one can see that end-point A will in this case be aware of 5 incoming
SSRCs, BV1-FV1. If this mixer intend to have the same behavior as in SSRCs, BV1-FV1. If this mixer intend to have the same behavior as in
Appendix A.3.2 where the mixer provides the end-points with the two Appendix A.3.2 where the mixer provides the end-points with the two
latest speaking end-points, then only two out of these five SSRCs latest speaking end-points, then only two out of these five SSRCs
will concurrently transmitt media to A. As the mixer selects which will concurrently transmitt media to A. As the mixer selects which
source in the different RTP sessions that transmit media to the end- source in the different RTP sessions that transmit media to the end-
points each media stream will require some rewriting when being points each RTP media stream will require some rewriting when being
projected from one session into another. The main thing is that the projected from one session into another. The main thing is that the
sequence number will need to be consequitvely incremented based on sequence number will need to be consequitvely incremented based on
the packet actually being transmitted in each RTP session. Thus the the packet actually being transmitted in each RTP session. Thus the
RTP sequence number offset will change each time a source is turned RTP sequence number offset will change each time a source is turned
on in RTP session. on in RTP session.
As the RTP sessions are independent the SSRC numbers used can be As the RTP sessions are independent the SSRC numbers used can be
handled indepdentently also thus working around any SSRC collisions handled indepdentently also thus working around any SSRC collisions
by having remapping tables between the RTP sessions. However the by having remapping tables between the RTP sessions. However the
related MediaStream signalling must be correspondlingly changed to related WebRTC MediaStream signalling must be correspondlingly
ensure consistent MediaStream to SSRC mappings between the different changed to ensure consistent WebRTC MediaStream to SSRC mappings
PeerConnections and the same comment that higher functions must not between the different PeerConnections and the same comment that
use SSRC as references to media streams applies also here. higher functions must not use SSRC as references to RTP media streams
applies also here.
The mixer will also be responsible to act on any RTCP codec control The mixer will also be responsible to act on any RTCP codec control
requests comming from an end-point and decide if it can act on it requests comming from an end-point and decide if it can act on it
locally or needs to translate the request into the RTP session that locally or needs to translate the request into the RTP session that
contains the media source. Both end-points and the mixer will need contains the media source. Both end-points and the mixer will need
to implement conference related codec control functionalities to to implement conference related codec control functionalities to
provide a good experience. Full Intra Request to request from the provide a good experience. Full Intra Request to request from the
media source to provide switching points between the sources, media source to provide switching points between the sources,
Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
to aggregate congestion control response towards the media source and to aggregate congestion control response towards the media source and
have it adjust its bit-rate in case the limitation is not in the have it adjust its bit-rate in case the limitation is not in the
source to mixer link. source to mixer link.
This version of the mixer also puts different requirements on the This version of the mixer also puts different requirements on the
end-point when it comes to decoder instances and handling of the end-point when it comes to decoder instances and handling of the RTP
media streams providing media. As each projected SSRC can at any media streams providing media. As each projected SSRC can at any
time provide media the end-point either needs to handle having thus time provide media the end-point either needs to handle having thus
many allocated decoder instances or have efficient switching of many allocated decoder instances or have efficient switching of
decoder contexts in a more limited set of actual decoder instances to decoder contexts in a more limited set of actual decoder instances to
cope with the switches. The WebRTC application also gets more cope with the switches. The WebRTC application also gets more
responsibility to update how the media provides is to be presented to responsibility to update how the media provides is to be presented to
the user. the user.
A.4. Translator Based A.4. Translator Based
There is also a variety of translators. The core commonality is that There is also a variety of translators. The core commonality is that
they do not need to make themselves visible in the RTP level by they do not need to make themselves visible in the RTP level by
having an SSRC themselves. Instead they sit between one or more end- having an SSRC themselves. Instead they sit between one or more end-
point and perform translation at some level. It can be media point and perform translation at some level. It can be media
transcoding, protocol translation or covering missing functionality transcoding, protocol translation or covering missing functionality
for a legacy device or simply relay packets between transport domains for a legacy end-point or simply relay packets between transport
or to realize multi-party. We will go in details below. domains or to realize multi-party. We will go in details below.
A.4.1. Transcoder A.4.1. Transcoder
A transcoder operates on media level and really used for two A transcoder operates on media level and really used for two
purposes, the first is to allow two end-points that doesn't have a purposes, the first is to allow two end-points that doesn't have a
common set of media codecs to communicate by translating from one common set of media codecs to communicate by translating from one
codec to another. The second is to change the bit-rate to a lower codec to another. The second is to change the bit-rate to a lower
one. For WebRTC end-points communicating with each other only the one. For WebRTC end-points communicating with each other only the
first one should at all be relevant. In certain legacy deployment first one should at all be relevant. In certain legacy deployment
media transcoder will be necessary to ensure both codecs and bit-rate media transcoder will be necessary to ensure both codecs and bit-rate
falls within the envelope the legacy device supports. falls within the envelope the legacy end-point supports.
As transcoding requires access to the media the transcoder must As transcoding requires access to the media the transcoder must
within the security context and access any media encryption and within the security context and access any media encryption and
integrity keys. On the RTP plane a media transcoder will in practice integrity keys. On the RTP plane a media transcoder will in practice
fork the RTP session into two different domains that are highly fork the RTP session into two different domains that are highly
decoupled when it comes to media parameters and reporting, but not decoupled when it comes to media parameters and reporting, but not
identities. To maintain signalling bindings to SSRCs a transcoder is identities. To maintain signalling bindings to SSRCs a transcoder is
likely needing to use the SSRC of one end-point to represent the likely needing to use the SSRC of one end-point to represent the
transcoded media stream to the other end-point(s). The congestion transcoded RTP media stream to the other end-point(s). The
control loop can be terminated in the transcoder as the media bit- congestion control loop can be terminated in the transcoder as the
rate being sent by the transcoder can be adjusted independently of media bit-rate being sent by the transcoder can be adjusted
the incoming bit-rate. However, for optimizing performance and independently of the incoming bit-rate. However, for optimizing
resource consumption the translator needs to consider what signals or performance and resource consumption the translator needs to consider
bit-rate reductions it should send towards the source end-point. For what signals or bit-rate reductions it should send towards the source
example receving a 2.5 mbps video stream and then send out a 250 kbps end-point. For example receving a 2.5 mbps video stream and then
video stream after transcoding is a vaste of resources. In most send out a 250 kbps video stream after transcoding is a vaste of
cases a 500 kbps video stream from the source in the right resolution resources. In most cases a 500 kbps video stream from the source in
is likely to provide equal quality after transcoding as the 2.5 mbps the right resolution is likely to provide equal quality after
source stream. At the same time increasing media bit-rate futher transcoding as the 2.5 mbps source stream. At the same time
than what is needed to represent the incoming quality accurate is increasing media bit-rate futher than what is needed to represent the
also wasted resources. incoming quality accurate is also wasted resources.
+-A-------------+ +-Translator------------------+ +-A-------------+ +-Translator------------------+
| +-PeerC1------| |-PeerC1--------+ | | +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | | | | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | | | | | +-RTP1----| |-RTP1------+ | | |
| | | | +-Audio-| |-Audio---+ | | | +---+ | | | | | +-Audio-| |-Audio---+ | | | +---+ |
| | | | | AA1|------------>|---------+-+-+-+-|DEC|----+ | | | | | | AA1|------------>|---------+-+-+-+-|DEC|----+ |
| | | | | |<------------|BA1 <----+ | | | +---+ | | | | | | | |<------------|BA1 <----+ | | | +---+ | |
| | | | | | | |\| | | +---+ | | | | | | | | | |\| | | +---+ | |
| | | | +-------| |---------+ +-+-+-|ENC|<-+ | | | | | | +-------| |---------+ +-+-+-|ENC|<-+ | |
skipping to change at page 53, line 33 skipping to change at page 52, line 33
| | | | +-Audio-| |-Audio---+ | | | +---+ | | | | | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | BA1|------------>|---------+-+-+-+-|DEC|--+ | | | | | | | BA1|------------>|---------+-+-+-+-|DEC|--+ | |
| | | | | |<------------|AA1 <----+ | | | +---+ | | | | | | | |<------------|AA1 <----+ | | | +---+ | |
| | | | | | | |\| | | +---+ | | | | | | | | | |\| | | +---+ | |
| | | | +-------| |---------+ +-+-+-|ENC|<---+ | | | | | +-------| |---------+ +-+-+-|ENC|<---+ |
| | | +---------| |-----------+ | | +---+ | | | | +---------| |-----------+ | | +---+ |
| | +-----------| |-------------+ | | | | +-----------| |-------------+ | |
| +-------------| |---------------+ | | +-------------| |---------------+ |
+---------------+ +-----------------------------+ +---------------+ +-----------------------------+
Figure 15: Media Transcoder Figure 13: Media Transcoder
Figure 15 exposes some important details. First of all you can see Figure 13 exposes some important details. First of all you can see
the SSRC identifiers used by the translator are the corresponding the SSRC identifiers used by the translator are the corresponding
end-points. Secondly, there is a relation between the RTP sessions end-points. Secondly, there is a relation between the RTP sessions
in the two different PeerConnections that are represtented by having in the two different PeerConnections that are represtented by having
both parts be identified by the same level and they need to share both parts be identified by the same level and they need to share
certain contexts. Also certain type of RTCP messages will need to be certain contexts. Also certain type of RTCP messages will need to be
bridged between the two parts. Certain RTCP feedback messages are bridged between the two parts. Certain RTCP feedback messages are
likely needed to be soruced by the translator in response to actions likely needed to be soruced by the translator in response to actions
by the translator and its media encoder. by the translator and its media encoder.
A.4.2. Gateway / Protocol Translator A.4.2. Gateway / Protocol Translator
Gateways are used when some protocol feature that is required is not Gateways are used when some protocol feature that is required is not
supported by an end-point wants to participate in session. This RTP supported by an end-point wants to participate in session. This RTP
translator in Figure 16 takes on the role of ensuring that from the translator in Figure 14 takes on the role of ensuring that from the
perspective of participant A, participant B appears as a fully perspective of participant A, participant B appears as a fully
compliant WebRTC end-point (that is, it is the combination of the compliant WebRTC end-point (that is, it is the combination of the
Translator and participant B that looks like a WebRTC end point). Translator and participant B that looks like a WebRTC end point).
+------------+ +------------+
| | | |
+---+ | Translator | +---+ +---+ | Translator | +---+
| A |<---->| to legacy |<---->| B | | A |<---->| to legacy |<---->| B |
+---+ | end-point | +---+ +---+ | end-point | +---+
WebRTC | | Legacy WebRTC | | Legacy
+------------+ +------------+
Figure 16: Gateway (RTP translator) towards legacy end-point Figure 14: Gateway (RTP translator) towards legacy end-point
For WebRTC there are a number of requirements that could force the For WebRTC there are a number of requirements that could force the
need for a gateway if a WebRTC end-point is to communicate with a need for a gateway if a WebRTC end-point is to communicate with a
legacy end-point, such as support of ICE and DTLS-SRTP for legacy end-point, such as support of ICE and DTLS-SRTP for
keymanagement. On RTP level the main functions that may be missing keymanagement. On RTP level the main functions that may be missing
in a legacy implementation that otherswise support RTP are RTCP in in a legacy implementation that otherswise support RTP are RTCP in
general, SRTP implementation, congestion control and feedback general, SRTP implementation, congestion control and feedback
messages required to make it work. messages required to make it work.
+-A-------------+ +-Translator------------------+ +-A-------------+ +-Translator------------------+
skipping to change at page 54, line 51 skipping to change at page 53, line 51
| | | +-Audio-| |-Audio---+ +---+-+ | | || | | | +-Audio-| |-Audio---+ +---+-+ | | ||
| | | | |<---RTCP---->|<--------+----------+ | | || | | | | |<---RTCP---->|<--------+----------+ | | ||
| | | | BA1|------------>|---------+--------------+ | || | | | | BA1|------------>|---------+--------------+ | ||
| | | | |<------------|AA1 <----+----------------+ || | | | | |<------------|AA1 <----+----------------+ ||
| | | +-------| |---------+ || | | | +-------| |---------+ ||
| | +---------| |----------------------------+| | | +---------| |----------------------------+|
| +-----------| |-----------+ | | +-----------| |-----------+ |
| | | | | | | |
+---------------+ +-----------------------------+ +---------------+ +-----------------------------+
Figure 17: RTP/RTCP Protocol Translator Figure 15: RTP/RTCP Protocol Translator
The legacy gateway may be implemented in several ways and what it The legacy gateway may be implemented in several ways and what it
need to change is higly dependent on what functions it need to proxy need to change is higly dependent on what functions it need to proxy
for the legacy end-point. One possibility is depicted in Figure 17 for the legacy end-point. One possibility is depicted in Figure 15
where the RTP media streams are compatible and forward without where the RTP media streams are compatible and forward without
changes. However, their RTP header values are captured to enable the changes. However, their RTP header values are captured to enable the
RTCP translator to create RTCP reception information related to the RTCP translator to create RTCP reception information related to the
leg between the end-point and the translator. This can then be leg between the end-point and the translator. This can then be
combined with the more basic RTCP reports that the legacy endpoint combined with the more basic RTCP reports that the legacy endpoint
(B) provides to give compatible and expected RTCP reporting to A. (B) provides to give compatible and expected RTCP reporting to A.
Thus enabling at least full congestion control on the path between A Thus enabling at least full congestion control on the path between A
and the translator. If B has limited possibilities for congestion and the translator. If B has limited possibilities for congestion
response for the media then the translator may need the capabilities response for the media then the translator may need the capabilities
to perform media transcoding to address cases where it otherwise to perform media transcoding to address cases where it otherwise
skipping to change at page 55, line 37 skipping to change at page 54, line 37
encryption and integirty protection operation to resolve missmatch in encryption and integirty protection operation to resolve missmatch in
security systems. security systems.
A.4.3. Relay A.4.3. Relay
There exist a class of translators that operates on transport level There exist a class of translators that operates on transport level
below RTP and thus do not effect RTP/RTCP packets directly. They below RTP and thus do not effect RTP/RTCP packets directly. They
come in two distinct flavors, the one used to bridge between two come in two distinct flavors, the one used to bridge between two
different transport or address domains to more function as a gateway different transport or address domains to more function as a gateway
and the second one which is to to provide a group communication and the second one which is to to provide a group communication
feature as depicted below in Figure 18. feature as depicted below in Figure 16.
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Translator | | Translator |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 18: RTP Translator (Relay) with Only Unicast Paths Figure 16: RTP Translator (Relay) with Only Unicast Paths
The first kind is straight forward and is likely to exist in WebRTC The first kind is straight forward and is likely to exist in WebRTC
context when an legacy end-point is compatible with the exception for context when an legacy end-point is compatible with the exception for
ICE, and thus needs a gateway that terminates the ICE and then ICE, and thus needs a gateway that terminates the ICE and then
forwards all the RTP/RTCP traffic and keymanagment to the end-point forwards all the RTP/RTCP traffic and keymanagment to the end-point
only rewriting the IP/UDP to forward the packet to the legacy node. only rewriting the IP/UDP to forward the packet to the legacy node.
The second type is useful if one wants a less complex central node or The second type is useful if one wants a less complex central node or
a central node that is outside of the security context and thus do a central node that is outside of the security context and thus do
not have access to the media. This relay takes on the role of not have access to the media. This relay takes on the role of
forwarding the media (RTP and RTCP) packets to the other end-points forwarding the media (RTP and RTCP) packets to the other end-points
but doesn't perform any RTP or media processing. Such a device but doesn't perform any RTP or media processing. Such a device
simply forwards the media from each sender to all of the other simply forwards the media from each sender to all of the other
particpants, and is sometimes called a transport-layer translator. particpants, and is sometimes called a transport-layer translator.
In Figure 18, participant A will only need to send a media once to In Figure 16, participant A will only need to send a media once to
the relay, which will redistribute it by sending a copy of the stream the relay, which will redistribute it by sending a copy of the stream
to participants B, C, and D. Participant A will still receive three to participants B, C, and D. Participant A will still receive three
RTP streams with the media from B, C and D if they transmit RTP streams with the media from B, C and D if they transmit
simultaneously. This is from an RTP perspective resulting in an RTP simultaneously. This is from an RTP perspective resulting in an RTP
session that behaves equivalent to one transporter over an IP Any session that behaves equivalent to one transporter over an IP Any
Source Multicast (ASM). Source Multicast (ASM).
This results in one common RTP session between all participants This results in one common RTP session between all participants
despite that there will be independent PeerConnections created to the despite that there will be independent PeerConnections created to the
translator as depicted below Figure 19. translator as depicted below Figure 17.
+-A-------------+ +-RELAY--------------------------+ +-A-------------+ +-RELAY--------------------------+
| +-PeerC1------| |-PeerC1--------+ | | +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | | | | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1-------------------------+ | | | | +-RTP1----| |-RTP1-------------------------+ |
| | | | +-Video-| |-Video---+ | | | | | | +-Video-| |-Video---+ | |
| | | | | AV1|------------>|---------------------------+ | | | | | | | AV1|------------>|---------------------------+ | |
| | | | | |<------------|BV1 <--------------------+ | | | | | | | | |<------------|BV1 <--------------------+ | | |
| | | | | |<------------|CV1 <------------------+ | | | | | | | | | |<------------|CV1 <------------------+ | | | |
| | | | +-------| |---------+ | | | | | | | | | +-------| |---------+ | | | | |
skipping to change at page 57, line 48 skipping to change at page 56, line 48
| | | | +-Video-| |-Video---+ | | | | | | | | | +-Video-| |-Video---+ | | | | |
| | | | | CV1|------------>|-------------------------+ | | | | | | | | CV1|------------>|-------------------------+ | | |
| | | | | |<------------|AV1 <----------------------+ | | | | | | | |<------------|AV1 <----------------------+ | |
| | | | | |<------------|BV1 <------------------+ | | | | | | | |<------------|BV1 <------------------+ | |
| | | | +-------| |---------+ | | | | | | +-------| |---------+ | |
| | | +---------| |------------------------------+ | | | | +---------| |------------------------------+ |
| | +-----------| |-------------+ | | | | +-----------| |-------------+ | |
| +-------------| |---------------+ | | +-------------| |---------------+ |
+---------------+ +--------------------------------+ +---------------+ +--------------------------------+
Figure 19: Transport Multi-party Relay Figure 17: Transport Multi-party Relay
As the Relay RTP and RTCP packets between the UDP flows as indicated As the Relay RTP and RTCP packets between the UDP flows as indicated
by the arrows for the media flow a given WebRTC end-point, like A by the arrows for the media flow a given WebRTC end-point, like A
will see the remote sources BV1 and CV1. There will be also two will see the remote sources BV1 and CV1. There will be also two
different network paths between A, and B or C. This results in that different network paths between A, and B or C. This results in that
the client A must be capable of handlilng that when determining the client A must be capable of handlilng that when determining
congestion state that there might exist multiple destinations on the congestion state that there might exist multiple destinations on the
far side of a PeerConnection and that these paths shall be treated far side of a PeerConnection and that these paths shall be treated
differently. It also results in a requirement to combine the differently. It also results in a requirement to combine the
different congestion states into a decision to transmit a particular different congestion states into a decision to transmit a particular
media stream suitable to all participants. RTP media stream suitable to all participants.
It is also important to note that the relay can not perform selective It is also important to note that the relay can not perform selective
relaying of some sources and not others. The reason is that the RTCP relaying of some sources and not others. The reason is that the RTCP
reporting in that case becomes incosistent and without explicit reporting in that case becomes incosistent and without explicit
information about it being blocked must be interpret as severe information about it being blocked must be interpret as severe
congestion. congestion.
In this usage it is also necessary that the session management has In this usage it is also necessary that the session management has
configured a common set of RTP configuration including RTP payload configured a common set of RTP configuration including RTP payload
formats as when A sends a packet with pt=97 it will arrive at both B formats as when A sends a packet with pt=97 it will arrive at both B
skipping to change at page 58, line 40 skipping to change at page 57, line 40
RTP session. RTP session.
The second problem can basically be solved in two ways. Either a The second problem can basically be solved in two ways. Either a
common master key from which all derive their per source key for common master key from which all derive their per source key for
SRTP. The second alternative which might be more practical is that SRTP. The second alternative which might be more practical is that
each end-point has its own key used to protects all RTP/RTCP packets each end-point has its own key used to protects all RTP/RTCP packets
it sends. Each participants key are then distributed to the other it sends. Each participants key are then distributed to the other
participants. This second method could be implemented using DTLS- participants. This second method could be implemented using DTLS-
SRTP to a special key server and then use Encrypted Key Transport SRTP to a special key server and then use Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] to distribute the actual used key to the [I-D.ietf-avt-srtp-ekt] to distribute the actual used key to the
other participants in the RTP session Figure 20. The first one could other participants in the RTP session Figure 18. The first one could
be achieved using MIKEY messages in SDP. be achieved using MIKEY messages in SDP.
+---+ +---+ +---+ +---+
| | +-----------+ | | | | +-----------+ | |
| A |<------->| DTLS-SRTP |<------->| C | | A |<------->| DTLS-SRTP |<------->| C |
| |<-- -->| HOST |<-- -->| | | |<-- -->| HOST |<-- -->| |
+---+ \ / +-----------+ \ / +---+ +---+ \ / +-----------+ \ / +---+
X X X X
+---+ / \ +-----------+ / \ +---+ +---+ / \ +-----------+ / \ +---+
| |<-- -->| RTP |<-- -->| | | |<-- -->| RTP |<-- -->| |
| B |<------->| RELAY |<------->| D | | B |<------->| RELAY |<------->| D |
| | +-----------+ | | | | +-----------+ | |
+---+ +---+ +---+ +---+
Figure 20: DTLS-SRTP host and RTP Relay Separated Figure 18: DTLS-SRTP host and RTP Relay Separated
The relay can still verify that a given SSRC isn't used or spoofed by The relay can still verify that a given SSRC isn't used or spoofed by
another participant within the multi-party session by binding SSRCs another participant within the multi-party session by binding SSRCs
on their first usage to a given source address and port pair. on their first usage to a given source address and port pair.
Packets carrying that source SSRC from other addresses can be Packets carrying that source SSRC from other addresses can be
suppressed to prevent spoofing. This is possible as long as SRTP is suppressed to prevent spoofing. This is possible as long as SRTP is
used which leaves the SSRC of the packet originator in RTP and RTCP used which leaves the SSRC of the packet originator in RTP and RTCP
packets in the clear. If such packet level method for enforcing packets in the clear. If such packet level method for enforcing
source authentication within the group, then there exist source authentication within the group, then there exist
cryptographic methods such as TESLA [RFC4383] that could be used for cryptographic methods such as TESLA [RFC4383] that could be used for
true source authentication. true source authentication.
A.5. End-point Forwarding A.5. End-point Forwarding
An WebRTC end-point (B in Figure 21) will receive a MediaStream (set An WebRTC end-point (B in Figure 19) will receive a WebRTC
of SSRCs) over a PeerConnection (from A). For the moment is not MediaStream (set of SSRCs) over a PeerConnection (from A). For the
decided if the end-point is allowed or not to in its turn send that moment is not decided if the end-point is allowed or not to in its
MediaStream over another PeerConnection to C. This section discusses turn send that WebRTC MediaStream over another PeerConnection to C.
the RTP and end-point implications of allowing such functionality, This section discusses the RTP and end-point implications of allowing
which on the API level is extremely simplistic to perform. such functionality, which on the API level is extremely simplistic to
perform.
+---+ +---+ +---+ +---+ +---+ +---+
| A |--->| B |--->| C | | A |--->| B |--->| C |
+---+ +---+ +---+ +---+ +---+ +---+
Figure 21: 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 media stream. The second one C. The first one is to simply relay the RTP media stream. The second
is for B to act as a transcoder. Lets consider both approaches. one is for B to act as a transcoder. Lets consider both approaches.
A relay approache will result in that the WebRTC end-points will have A relay approache 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.
The relay based approach has the advantage that B does not need to The relay based approach has the advantage that B does not need to
transcode the media thus both maintaining the quality of the encoding transcode the media thus both maintaining the quality of the encoding
and reducing B's complexity requirements. If the right security and reducing B's complexity requirements. If the right security
solutions are supported then also C will be able to verify the solutions are supported then also C will be able to verify the
authenticity of the media comming from A. As downside A are forced to authenticity of the media comming from A. As downside A are forced to
take both B and C into consideration when delivering content. take both B and C into consideration when delivering content.
The media transcoder approach is similar to having B act as Mixer The media transcoder approach is similar to having B act as Mixer
terminating the RTP session combined with the transcoder as discussed terminating the RTP session combined with the transcoder as discussed
in Appendix A.4.1. A will only see B as receiver of its media. B in Appendix A.4.1. A will only see B as receiver of its media. B
will responsible to produce a media stream suitable for the B to C will responsible to produce a RTP media stream suitable for the B to
PeerConnection. This may require media transcoding for congestion C PeerConnection. This may require media transcoding for congestion
control purpose to produce a suitable bit-rate. Thus loosing media control purpose to produce a suitable bit-rate. Thus loosing media
quality in the transcoding and forcing B to spend the resource on the quality in the transcoding and forcing B to spend the resource on the
transcoding. The media transcoding does result in a separation of transcoding. The media transcoding does result in a separation of
the two different legs removing almost all dependencies. B could the two different legs removing almost all dependencies. B could
choice to implement logic to optimize its media transcoding choice to implement logic to optimize its media transcoding
operation, by for example requesting media properties that are operation, by for example requesting media properties that are
suitable for C also, thus trying to avoid it having to transcode the suitable for C also, thus trying to avoid it having to transcode the
content and only forward the media payloads between the two sides. content and only forward the media payloads between the two sides.
For that optimization to be practical WebRTC end-points must support For that optimization to be practical WebRTC end-points must support
sufficiently good tools for codec control. sufficiently good tools for codec control.
A.6. Simulcast A.6. Simulcast
This section discusses simulcast in the meaning of providing a node, This section discusses simulcast in the meaning of providing a node,
for example a stream switching Mixer, with multiple different encoded for example a stream switching Mixer, with multiple different encoded
version of the same media source. In the WebRTC context that appears version of the same media source. In the WebRTC context that appears
to be most easily accomplished by establishing mutliple to be most easily accomplished by establishing mutliple
PeerConnection all being feed the same set of MediaStreams. Each PeerConnection all being feed the same set of WebRTC MediaStreams.
PeerConnection is then configured to deliver a particular media Each PeerConnection is then configured to deliver a particular media
quality and thus media bit-rate. This will work well as long as the quality and thus media bit-rate. This will work well as long as the
end-point implements media encoding according to Figure 9. Then each end-point implements media encoding according to Figure 7. Then each
PeerConnection will receive an independently encoded version and the PeerConnection will receive an independently encoded version and the
codec parameters can be agreed specifically in the context of this codec parameters can be agreed specifically in the context of this
PeerConnection. PeerConnection.
For simulcast to work one needs to prevent that the end-point deliver For simulcast to work one needs to prevent that the end-point deliver
content encoded as depicted in Figure 10. If a single encoder content encoded as depicted in Figure 8. If a single encoder
instance is feed to multiple PeerConnections the intention of instance is feed to multiple PeerConnections the intention of
performing simulcast will fail. performing simulcast will fail.
Thus it should be considered to explicitly signal which of the two Thus it should be considered to explicitly signal which of the two
implementation strategies that are desired and which will be done. implementation strategies that are desired and which will be done.
At least making the application and possible the central node At least making the application and possible the central node
interested in receiving simulcast of an end-points media streams to interested in receiving simulcast of an end-points RTP media streams
be aware if it will function or not. to be aware if it will function or not.
Authors' Addresses Authors' Addresses
Colin Perkins Colin Perkins
University of Glasgow University of Glasgow
School of Computing Science School of Computing Science
Glasgow G12 8QQ Glasgow G12 8QQ
United Kingdom United Kingdom
Email: csp@csperkins.org Email: csp@csperkins.org
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