draft-ietf-rtcweb-rtp-usage-04.txt   draft-ietf-rtcweb-rtp-usage-05.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: January 17, 2013 Ericsson Expires: April 25, 2013 Ericsson
J. Ott J. Ott
Aalto University Aalto University
July 16, 2012 October 22, 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-04 draft-ietf-rtcweb-rtp-usage-05
Abstract Abstract
The Web Real-Time Communication (WebRTC) framework provides support The Web Real-Time Communication (WebRTC) framework provides support
for direct interactive rich communication using audio, video, text, for direct interactive rich communication using audio, video, text,
collaboration, games, etc. between two peers' web-browsers. This collaboration, games, etc. between two peers' web-browsers. This
memo describes the media transport aspects of the WebRTC framework. memo describes the media transport aspects of the WebRTC framework.
It specifies how the Real-time Transport Protocol (RTP) is used in It specifies how the Real-time Transport Protocol (RTP) is used in
the WebRTC context, and gives requirements for which RTP features, the WebRTC context, and gives requirements for which RTP features,
profiles, and extensions need to be supported. profiles, and extensions need to be supported.
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 17, 2013. This Internet-Draft will expire on April 25, 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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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 the RTP Profile . . . . . . . . . . . . . . . . 7 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8
4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 9 4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 8
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 11 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 10
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 10
4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 11
5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12 5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 11
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . . 13 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . . 12
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 12
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 14 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 13
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 14 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 13
5.1.6. Temporary Maximum Media Stream Bit Rate Request . . . 14 5.1.6. Temporary Maximum Media Stream Bit Rate Request
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 13
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 15 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 14
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 15 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 14
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 16 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 15
6.1. Negative Acknowledgements and RTP Retransmission . . . . . 16 6.1. Negative Acknowledgements and RTP Retransmission . . . . . 15
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 17 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 16
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 17 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 16
7.1. Congestion Control Requirements . . . . . . . . . . . . . 18 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . . 17
7.2. Rate Control Boundary Conditions . . . . . . . . . . . . . 19 7.2. RTCP Limitations for Congestion Control . . . . . . . . . 18
7.3. RTCP Limitations for Congestion Control . . . . . . . . . 19 7.3. Congestion Control Interoperability With Legacy Systems . 19
7.4. Congestion Control Interoperability With Legacy Systems . 20 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 19
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 20 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 20
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 21 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 20
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 21
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 22 11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 21
11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 22 12. RTP Implementation Considerations . . . . . . . . . . . . . . 22
12. RTP Implementation Considerations . . . . . . . . . . . . . . 23 12.1. RTP Sessions and PeerConnection . . . . . . . . . . . . . 22
12.1. RTP Sessions and PeerConnection . . . . . . . . . . . . . 23 12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 24
12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 25 12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 24
12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 25 12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 25
12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 26 12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 26
12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 27 12.6. Media Synchronization . . . . . . . . . . . . . . . . . . 27
12.6. Media Synchronization . . . . . . . . . . . . . . . . . . 28 12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . . 27
12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . . 28 12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 28
12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 29
12.9. Differentiated Treatment of Flows . . . . . . . . . . . . 29 12.9. Differentiated Treatment of Flows . . . . . . . . . . . . 29
13. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31 13. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 30
14. Security Considerations . . . . . . . . . . . . . . . . . . . 31 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 31 15. Security Considerations . . . . . . . . . . . . . . . . . . . 31
16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32 16. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 32
16.1. Normative References . . . . . . . . . . . . . . . . . . . 32 17. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32
16.2. Informative References . . . . . . . . . . . . . . . . . . 34 17.1. Normative References . . . . . . . . . . . . . . . . . . . 32
17.2. Informative References . . . . . . . . . . . . . . . . . . 35
Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 36 Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 36
A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 36 A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 37
A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 39 A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 40
A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 42 A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 43
A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 42 A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 43
A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 45 A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 46
A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . . 48 A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . . 49
A.4. Translator Based . . . . . . . . . . . . . . . . . . . . . 51 A.4. Translator Based . . . . . . . . . . . . . . . . . . . . . 52
A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . . 51 A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . . 52
A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 52 A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 53
A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 54 A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 55
A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . . 58 A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . . 59
A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 59 A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 60
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 60 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] provides a framework The Real-time Transport Protocol (RTP) [RFC3550] provides a framework
for delivery of audio and video teleconferencing data and other real- for delivery of audio and video teleconferencing data and other real-
time media applications. Previous work has defined the RTP protocol, time media applications. Previous work has defined the RTP protocol,
along with numerous profiles, payload formats, and other extensions. along with numerous profiles, payload formats, and other extensions.
When combined with appropriate signalling, these form the basis for When combined with appropriate signalling, these form the basis for
many teleconferencing systems. many teleconferencing systems.
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is to be used in the WebRTC context. It proposes a baseline set of is to be used in the WebRTC context. It proposes a baseline set of
RTP features that are to be implemented by all WebRTC-aware end- RTP features that are to be implemented by all WebRTC-aware end-
points, along with suggested extensions for enhanced functionality. points, along with suggested extensions for enhanced functionality.
The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete
WebRTC framework, of which this memo is a part. WebRTC framework, of which this memo is a part.
The structure of this memo is as follows. Section 2 outlines our The structure of this memo is as follows. Section 2 outlines our
rationale in preparing this memo and choosing these RTP features. rationale in preparing this memo and choosing these RTP features.
Section 3 defines requirement terminology. Requirements for core RTP Section 3 defines requirement terminology. Requirements for core RTP
protocols are described in Section 4 and recommended RTP extensions protocols are described in Section 4 and suggested RTP extensions are
are described in Section 5. Section 6 outlines mechanisms that can 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 congestion control and rate adaptation mechanisms. The discussion of
discussion of mandated RTP mechanisms concludes in Section 8 with a mandated RTP mechanisms concludes in Section 8 with a review of
review of performance monitoring and network management tools that performance monitoring and network management tools that can be used
can be used in the WebRTC context. Section 9 gives some guidelines in the WebRTC context. Section 9 gives some guidelines for future
for future incorporation of other RTP and RTP Control Protocol (RTCP) incorporation of other RTP and RTP Control Protocol (RTCP) extensions
extensions into this framework. Section 10 describes requirements into this framework. Section 10 describes requirements placed on the
placed on the signalling channel. Section 11 discusses the signalling channel. Section 11 discusses the relationship between
relationship between features of the RTP framework and the WebRTC features of the RTP framework and the WebRTC application programming
application programming interface (API), and Section 12 discusses RTP interface (API), and Section 12 discusses RTP implementation
implementation considerations. This memo concludes with an appendix considerations. This memo concludes with an appendix discussing
discussing several different RTP Topologies, and how they affect the several different RTP Topologies, and how they affect the RTP
RTP session(s) and various implementation details of possible session(s) and various implementation details of possible realization
realization of central nodes. 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
extensions are to be supported by new implementations. The extensions are to be supported by new implementations. The
development of the WebRTC framework provides an opportunity for us to development of the WebRTC framework provides an opportunity for us to
review the available RTP features and extensions, and to define a review the available RTP features and extensions, and to define a
common baseline feature set for all WebRTC implementations of RTP. common baseline feature set for all WebRTC implementations of RTP.
This builds on the past 15 years development of RTP to mandate the This builds on the past 15 years development of RTP to mandate the
use of extensions that have shown widespread utility, while still use of extensions that have shown widespread utility, while still
remaining compatible with the wide installed base of RTP remaining compatible with the wide installed base of RTP
implementations where possible. implementations where possible.
RTP and RTCP extensions not discussed in this document can still be Other RTP and RTCP extensions not discussed in this document can be
implemented by a WebRTC end-point, but they are considered optional, implemented by WebRTC end-points if they are beneficial for new use
are not required for interoperability, and do not provide features cases. However, they are not necessary to address the WebRTC use
needed to address the WebRTC use cases and requirements cases and requirements identified to date
[I-D.ietf-rtcweb-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 targeted 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 appropriate 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 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
use of RTP features and extensions. When we describe the importance "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
of RTP extensions, or the need for implementation support, we use the document are to be interpreted as described in [RFC2119]. The RFC
following requirement levels to specify the importance of the feature 2119 interpretation of these key words applies only when written in
in the WebRTC framework: ALL CAPS. Lower- or mixed-case uses of these key words are not to be
interpreted as carrying special significance in this memo.
MUST: This word, or the terms "REQUIRED" or "SHALL", mean that the
definition is an absolute requirement of the specification.
SHOULD: This word, or the adjective "RECOMMENDED", mean that there
may exist valid reasons in particular circumstances to ignore a
particular item, but the full implications must be understood and
carefully weighed before choosing a different course.
MAY: This word, or the adjective "OPTIONAL", mean that an item is
truly optional. One vendor may choose to include the item because
a particular marketplace requires it or because the vendor feels
that it enhances the product while another vendor may omit the
same item. An implementation which does not include a particular
option MUST be prepared to interoperate with another
implementation which does include the option, though perhaps with
reduced functionality. In the same vein an implementation which
does include a particular option MUST be prepared to interoperate
with another implementation which does not include the option
(except, of course, for the feature the option provides.)
These key words are used in a manner consistent with their definition
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: We define the following terms:
RTP Media Stream: A sequence of RTP packets, and associated RTCP RTP Media Stream: A sequence of RTP packets, and associated RTCP
packets, using a single synchronisation source (SSRC) that packets, using a single synchronisation source (SSRC) that
together carries part or all of the content of a specific Media together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session. Type from a specific sender source within a given RTP session.
RTP Session: As defined by [RFC3550], the endpoints belonging to the RTP Session: As defined by [RFC3550], the endpoints belonging to the
same RTP Session are those that share a single SSRC space. That same RTP Session are those that share a single SSRC space. That
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Implementers are advised to consider the requirements for graceful Implementers are advised to consider the requirements for graceful
degradation when interoperating with legacy implementations. degradation when interoperating with legacy implementations.
Other implementation considerations are discussed in Section 12. Other implementation considerations are discussed in Section 12.
4.2. Choice of the RTP Profile 4.2. Choice of the RTP Profile
The complete specification of RTP for a particular application domain The complete specification of RTP for a particular application domain
requires the choice of an RTP Profile. For WebRTC use, the "Extended requires the choice of an RTP Profile. For WebRTC use, the "Extended
Secure RTP Profile for Real-time Transport Control Protocol (RTCP)- Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-
Based Feedback (RTP/SAVPF)" [RFC5124] is REQUIRED to be implemented. Based Feedback (RTP/SAVPF)" [RFC5124] as extended by
This builds on the basic RTP/AVP profile [RFC3551], the RTP profile [I-D.terriberry-avp-codecs] MUST be implemented. This builds on the
for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP basic RTP/AVP profile [RFC3551], the RTP profile for RTCP-based
profile (RTP/SAVP) [RFC3711]. feedback (RTP/AVPF) [RFC4585], and the secure RTP profile (RTP/SAVP)
[RFC3711].
The RTCP-based feedback extensions are needed for the improved RTCP The RTCP-based feedback extensions [RFC4585] are needed for the
timer model, that allows more flexible transmission of RTCP packets improved RTCP timer model, that allows more flexible transmission of
in response to events, rather than strictly according to bandwidth. RTCP packets in response to events, rather than strictly according to
This is vital for being able to report congestion events. These bandwidth. This is vital for being able to report congestion events.
extensions also save RTCP bandwidth, and will commonly only use the These extensions also save RTCP bandwidth, and will commonly only use
full RTCP bandwidth allocation if there are many events that require the full RTCP bandwidth allocation if there are many events that
feedback. They are also needed to make use of the RTP conferencing require feedback. They are also needed to make use of the RTP
extensions discussed in Section 5.1. conferencing 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 secure RTP profile is needed to provide SRTP media encryption, The secure RTP profile [RFC3711] is needed to provide media
integrity protection, replay protection and a limited form of source encryption, integrity protection, replay protection and a limited
authentication. form of source authentication. WebRTC implementations MUST NOT send
packets using the basic RTP/AVP profile or the RTP/AVPF profile; they
WebRTC implementations MUST NOT send packets using the basic RTP/AVP MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP
profile or the RTP/AVPF profile; they MUST employ the full RTP/SAVPF packets that are generated. The default and mandatory to implement
profile to protect all RTP and RTCP packets that are generated. The transforms listed in Section 5 of [RFC3711] SHALL apply.
default and mandatory-to-implement transforms listed in Section 5 of
[RFC3711] SHALL apply.
Implementations MUST support DTLS-SRTP [RFC5764] for key-management. Implementations MUST support DTLS-SRTP [RFC5764] for key-management.
Other key management schemes MAY be supported. Other key management schemes MAY be supported.
4.3. Choice of RTP Payload Formats 4.3. Choice of RTP Payload Formats
The requirement from Section 6 of [RFC3551] that "Audio applications Implementations MUST follow the WebRTC Audio Codec and Processing
operating under this profile SHOULD, at a minimum, be able to send Requirements [I-D.ietf-rtcweb-audio] and SHOULD follow the updated
and/or receive payload types 0 (PCMU) and 5 (DVI4)" applies, since recommendations for audio codecs in the RTP/AVP Profile
Section 4.2 of this memo mandates the use of the RTP/SAVPF profile, [I-D.terriberry-avp-codecs]. Support for other audio codecs is
which inherits this restriction from the RTP/AVP profile. OPTIONAL.
(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 (tbd: the mandatory to implement video codec is not yet decided)
configurations of a single format, provided each uses a different RTP
payload type number. An endpoint that has signalled its support for
multiple formats is REQUIRED to accept data in any of those formats
at any time, unless it has previously signalled limitations on its
decoding capability.
Endpoints MAY signal support for multiple RTP payload formats, or
multiple configurations of a single RTP payload format, provided each
payload format uses a different RTP payload type number. An endpoint
that has signalled support for multiple RTP payload formats SHOULD
accept data in any of those payload formats at any time, unless it
has previously signalled limitations on its decoding capability.
This requirement is constrained if several media types are sent in This requirement is constrained if several media types are sent in
the same RTP session. In such a case, a source (SSRC) is restricted the same RTP session. In such a case, a source (SSRC) is restricted
to switching only between the RTP payload formats signalled for the 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 media type that is being sent by that source; see Section 4.4. To
support rapid rate adaptation, RTP does not require signalling in support rapid rate adaptation by changing codec, RTP does not require
advance for changes between payload formats that were signalled advance signalling for changes between RTP payload formats that were
during session setup. signalled during session set-up.
An RTP sender that changes between two RTP payload types that use An RTP sender that changes between two RTP payload types that use
different RTP clock rates MUST follow the recommendations in Section different RTP clock rates MUST follow the recommendations in Section
4.1 of [I-D.ietf-avtext-multiple-clock-rates]. RTP receivers MUST 4.1 of [I-D.ietf-avtext-multiple-clock-rates]. RTP receivers MUST
follow the recommendations in Section 4.3 of follow the recommendations in Section 4.3 of
[I-D.ietf-avtext-multiple-clock-rates], in order to support sources [I-D.ietf-avtext-multiple-clock-rates], in order to support sources
that switch between clock rates in an RTP session (these that switch between clock rates in an RTP session (these
recommendations for receivers are backwards compatible with the case recommendations for receivers are backwards compatible with the case
where senders use only a single clock rate). where senders use only a single clock rate).
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sessions over a single UDP flow.) sessions over a single UDP flow.)
4.5. RTP and RTCP Multiplexing 4.5. RTP and RTCP Multiplexing
Historically, RTP and RTCP have been run on separate transport layer Historically, RTP and RTCP have been run on separate transport layer
addresses (e.g., two UDP ports for each RTP session, one port for RTP addresses (e.g., two UDP ports for each RTP session, one port for RTP
and one port for RTCP). With the increased use of Network Address/ and one port for RTCP). With the increased use of Network Address/
Port Translation (NAPT) this has become problematic, since Port Translation (NAPT) this has become problematic, since
maintaining multiple NAT bindings can be costly. It also complicates maintaining multiple NAT bindings can be costly. It also complicates
firewall administration, since multiple ports need to be opened to firewall administration, since multiple ports need to be opened to
allow RTP traffic. To reduce these costs and session setup times, allow RTP traffic. To reduce these costs and session set-up times,
support for multiplexing RTP data packets and RTCP control packets on support for multiplexing RTP data packets and RTCP control packets on
a single port for each RTP session is REQUIRED, as specified in a single port for each RTP session is REQUIRED, as specified in
[RFC5761]. For backwards compatibility, implementations are also [RFC5761]. For backwards compatibility, implementations are also
REQUIRED to support sending of RTP and RTCP to separate destination REQUIRED to support sending of RTP and RTCP to separate destination
ports. 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 can be useful to keep NAT if there is no active media traffic. This can be useful to keep NAT
bindings alive, and is the recommend method for application level bindings alive, and is the recommend method for application level
keep-alives of RTP sessions [RFC6263]. keep-alives of RTP sessions [RFC6263].
4.6. Reduced Size RTCP 4.6. Reduced Size RTCP
RTCP packets are usually sent as compound RTCP packets, and [RFC3550] RTCP packets are usually sent as compound RTCP packets, and [RFC3550]
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 under the RTP/AVPF Profile [RFC4585] these statistics are
unnecessarily increase the mean RTCP packet size. This can limit the not needed in every packet, and unnecessarily increase the mean RTCP
frequency at which RTCP packets can be sent within the RTCP bandwidth packet size. This can limit the frequency at which RTCP packets can
share. be sent within the RTCP bandwidth share.
To avoid this problem, [RFC5506] specifies how to reduce the mean To avoid this problem, [RFC5506] specifies how to reduce the mean
RTCP message size and allow for more frequent feedback. Frequent RTCP message size and allow for more frequent feedback. Frequent
feedback, in turn, is essential to make real-time applications feedback, in turn, is essential to make real-time applications
quickly aware of changing network conditions, and to allow them to quickly aware of changing network conditions, and to allow them to
adapt their transmission and encoding behaviour. Support for sending adapt their transmission and encoding behaviour. Support for sending
RTCP feedback packets as [RFC5506] non-compound packets is REQUIRED RTCP feedback packets as [RFC5506] non-compound packets is REQUIRED,
when signalled. For backwards compatibility, implementations are but MUST be negotiated using the signalling channel before use. For
also REQUIRED to support the use of compound RTCP feedback packets. backwards compatibility, implementations are also REQUIRED to support
the use of compound RTCP feedback packets if the remote endpoint does
not agree to the use of non-compound RTCP in the signalling exchange.
4.7. Symmetric RTP/RTCP 4.7. Symmetric RTP/RTCP
To ease traversal of NAT and firewall devices, implementations are To ease traversal of NAT and firewall devices, implementations are
REQUIRED to implement and use Symmetric RTP [RFC4961]. This requires REQUIRED to implement and use Symmetric RTP [RFC4961]. This requires
that the IP address and port used for sending and receiving RTP and that the IP address and port used for sending and receiving RTP and
RTCP 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
skipping to change at page 11, line 46 skipping to change at page 11, line 20
detected, or when the RTP application is restarted, its RTCP CNAME is detected, or when the RTP application is restarted, its RTCP CNAME is
meant to stay unchanged, so that RTP endpoints can be uniquely meant to stay unchanged, so that RTP endpoints can be uniquely
identified and associated with their RTP media streams within a set identified and associated with their RTP media streams within a set
of related RTP sessions. For proper functionality, each RTP endpoint of related RTP sessions. For proper functionality, each RTP endpoint
needs to have a unique RTCP CNAME 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, long-term persistent identifiers can be 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
specified in Section 4.2 of "Guidelines for Choosing RTP Control [I-D.rescorla-avtcore-6222bis] is RECOMMENDED.
Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is RECOMMENDED.
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 An WebRTC end-point MUST support reception of any CNAME that matches
the syntax limitations specified by the RTP specification [RFC3550] the syntax limitations specified by the RTP specification [RFC3550]
and cannot assume that any CNAME will be according to the recommended and cannot assume that any CNAME will be chosen according to the form
form above. suggested 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 needed to obtain There are a number of RTP extensions that are either needed to obtain
full functionality, or extremely useful to improve on the baseline full functionality, or extremely useful to improve on the baseline
performance, in the WebRTC application context. One set of these performance, in the WebRTC application context. One set of these
extensions is related to conferencing, while others are more generic extensions is related to conferencing, while others are more generic
in nature. The following subsections describe the various RTP in nature. The following subsections describe the various RTP
extensions mandated or suggested for use within the WebRTC context. extensions mandated or suggested for use within the WebRTC context.
5.1. Conferencing Extensions 5.1. Conferencing Extensions
RTP is inherently a group communication protocol. Groups can be RTP is inherently a group communication protocol. Groups can be
implemented using a centralised server, multi-unicast, or using IP implemented using a centralised server, multi-unicast, or using IP
multicast. While IP multicast was popular in early deployments, in multicast. While IP multicast 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 [I-D.westerlund-avtcore-rtp-topologies-update].
As discussed in Section 3.5 of [RFC5117], the use of a video As discussed in Section 3.7 of
[I-D.westerlund-avtcore-rtp-topologies-update], 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 3.8 of [I-D.westerlund-avtcore-rtp-topologies-update], the
RTCP termination breaks RTP loop detection and removes the ability use of a content modifying MCU with RTCP termination breaks RTP loop
for receivers to identify active senders. RTP Transport Translators detection and removes the ability for receivers to identify active
(Topo-Translator) are not of immediate interest to WebRTC, although senders. RTP Transport Translators (Topo-Translator) are not of
the main difference compared to point to point is the possibility of immediate interest to WebRTC, although the main difference compared
seeing multiple different transport paths in any RTCP feedback. to point to point is the possibility of seeing multiple different
Accordingly, only Point to Point (Topo-Point-to-Point), Multiple transport paths in any RTCP feedback. Accordingly, only Point to
concurrent Point to Point (Mesh) and RTP Mixers (Topo-Mixer) Point (Topo-Point-to-Point), Multiple concurrent Point to Point
topologies are needed to achieve the use-cases to be supported in (Mesh) and RTP Mixers (Topo-Mixer) topologies are needed to achieve
WebRTC initially. These RECOMMENDED topologies are expected to be the use-cases to be supported in WebRTC initially. These RECOMMENDED
supported by all WebRTC end-points (these topologies require no topologies are expected to be supported by all WebRTC end-points
special RTP-layer support in the end-point if the RTP features (these topologies require no special RTP-layer support in the end-
mandated in this memo are implemented). point if the RTP features mandated in this memo are implemented).
The RTP extensions described below to be used with centralised The RTP extensions described below to be used with centralised
conferencing -- where one RTP Mixer (e.g., a conference bridge) conferencing -- where one RTP Mixer (e.g., a conference bridge)
receives a participant's RTP media streams and distributes them to receives a participant's RTP media streams and distributes them to
the other participants -- are not necessary for interoperability; an the other participants -- are not necessary for interoperability; an
RTP endpoint that does not implement these extensions will work RTP endpoint that does not implement these extensions will work
correctly, but may offer poor performance. Support for the listed correctly, but might offer poor performance. Support for the listed
extensions will greatly improve the quality of experience and, to extensions will greatly improve the quality of experience and, to
provide a reasonable baseline quality, some these extensions are provide a reasonable baseline quality, some these extensions are
mandatory to be supported by WebRTC end-points. mandatory to be supported by WebRTC end-points.
The RTCP packets assisting in such operation are defined in the The RTCP conferencing extensions are defined in Extended RTP Profile
Extended RTP Profile for Real-time Transport Control Protocol (RTCP)- for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
Based Feedback (RTP/AVPF) [RFC4585] and the "Codec Control Messages AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
in the RTP Audio-Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
and are fully usable by the Secure variant of this profile (RTP/ usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
SAVPF) [RFC5124].
5.1.1. Full Intra Request (FIR) 5.1.1. Full Intra Request (FIR)
The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of the The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of the
Codec Control Messages [RFC5104]. This message is used to make the Codec Control Messages [RFC5104]. This message is used to make the
mixer request a new Intra picture from a participant in the session. mixer request a new Intra picture from a participant in the session.
This is used when switching between sources to ensure that the This is used when switching between sources to ensure that the
receivers can decode the video or other predictive media encoding receivers can decode the video or other predictive media encoding
with long prediction chains. It is REQUIRED that this feedback with long prediction chains. It is REQUIRED that WebRTC senders
message is supported by RTP senders in WebRTC, since it greatly understand the react to this feedback message since it greatly
improves the user experience when using centralised mixers-based improves the user experience when using centralised mixer-based
conferencing. conferencing; support for sending the FIR message is OPTIONAL.
5.1.2. Picture Loss Indication (PLI) 5.1.2. Picture Loss Indication (PLI)
The Picture Loss Indication is defined in Section 6.3.1 of the RTP/ The Picture Loss Indication is defined in Section 6.3.1 of the RTP/
AVPF profile [RFC4585]. It is used by a receiver to tell the sending AVPF profile [RFC4585]. It is used by a receiver to tell the sending
encoder that it lost the decoder context and would like to have it encoder that it lost the decoder context and would like to have it
repaired somehow. This is semantically different from the Full Intra repaired somehow. This is semantically different from the Full Intra
Request above as there there may be multiple methods to fulfill the Request above as there there could be multiple ways to fulfil the
request. It is REQUIRED that senders understand and react to this request. It is REQUIRED that WebRTC senders understand and react to
feedback message as a loss tolerance mechanism; receivers MAY send this feedback message as a loss tolerance mechanism; receivers MAY
PLI messages. send PLI messages.
5.1.3. Slice Loss Indication (SLI) 5.1.3. Slice Loss Indication (SLI)
The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF
profile [RFC4585]. It is used by a receiver to tell the encoder that profile [RFC4585]. It is used by a receiver to tell the encoder that
it has detected the loss or corruption of one or more consecutive it has detected the loss or corruption of one or more consecutive
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 (RPSI) 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 synchronisation, encoder has learned about a loss of encoder-decoder synchronisation,
a 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 RPSI message allows this to be signalled. Support for RPSI
messages is OPTIONAL.
Support for RPSI messages is OPTIONAL.
5.1.5. Temporal-Spatial Trade-off Request (TSTR) 5.1.5. Temporal-Spatial Trade-off Request (TSTR)
The temporal-spatial trade-off request and notification are defined The temporal-spatial trade-off request and notification are defined
in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used
to ask the video encoder to change the trade-off it makes between to ask the video encoder to change the trade-off it makes between
temporal and spatial resolution, for example to prefer high spatial temporal and spatial resolution, for example to prefer high spatial
image quality but low frame rate. image quality but low frame rate. Support for TSTR requests and
notifications is OPTIONAL.
Support for TSTR requests and notifications is OPTIONAL.
5.1.6. Temporary Maximum Media Stream Bit Rate Request 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
This feedback message is defined in Sections 3.5.4 and 4.2.1 of the This feedback message is defined in Sections 3.5.4 and 4.2.1 of the
Codec Control Messages [RFC5104]. This message and its notification Codec Control Messages [RFC5104]. This message and its notification
message are used by a media receiver to inform the sending party that message are used by a media receiver to inform the sending party that
there is a current limitation on the amount of bandwidth available to there is a current limitation on the amount of bandwidth available to
this receiver. This may have various reasons; for example, an RTP this receiver. This can be various reasons for this: for example, an
mixer may use this message to limit the media rate of the sender RTP mixer can use this message to limit the media rate of the sender
being forwarded by the mixer (without doing media transcoding) to fit being forwarded by the mixer (without doing media transcoding) to fit
the bottlenecks existing towards the other session participants. It the bottlenecks existing towards the other session participants. It
is REQUIRED that this feedback message is supported. A RTP media is REQUIRED that this feedback message is supported. WebRTC senders
stream sender receiving a TMMBR for its SSRC MUST follow the are REQUIRED to implement support for TMMBR messages, and MUST follow
limitations set by the message; the sending of TMMBR requests is bandwidth limitations set by a TMMBR message received for their SSRC.
OPTIONAL. The sending of TMMBR requests is OPTIONAL.
5.2. Header Extensions 5.2. Header Extensions
The RTP specification [RFC3550] provides the capability to include The RTP specification [RFC3550] provides the capability to include
RTP header extensions containing in-band data, but the format and RTP header extensions containing in-band data, but the format and
semantics of the extensions are poorly specified. The use of header semantics of the extensions are poorly specified. The use of header
extensions is OPTIONAL in the WebRTC context, but if they are used, extensions is OPTIONAL in the WebRTC context, but if they are used,
they MUST be formatted and signalled following the general mechanism they MUST be formatted and signalled following the general mechanism
for RTP header extensions defined in [RFC5285], since this gives for RTP header extensions defined in [RFC5285], since this gives
well-defined semantics to RTP header extensions. well-defined semantics to RTP header extensions.
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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 deliver 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. Negative Acknowledgements and RTP Retransmission 6.1. Negative Acknowledgements and RTP Retransmission
As a consequence of supporting the RTP/SAVPF profile, implementations As a consequence of supporting the RTP/SAVPF profile, implementations
will support negative acknowlegdements (NACKs) for RTP data packets will support negative acknowledgements (NACKs) for RTP data packets
[RFC4585]. This feedback can be used to inform a sender of the loss [RFC4585]. This feedback can be used to inform a sender of the loss
of particular RTP packets, subject to the capacity limitations of the of particular RTP packets, subject to the capacity limitations of the
RTCP feedback channel. A sender can use this information to optimise RTCP feedback channel. A sender can use this information to optimise
the user experience by adapting the media encoding to compensate for the user experience by adapting the media encoding to compensate for
known lost packets, for example. known lost packets, for example.
Senders are REQUIRED to understand the Generic NACK message defined Senders are REQUIRED to understand the Generic NACK message defined
in Section 6.2.1 of [RFC4585], but MAY choose to ignore this feedback in Section 6.2.1 of [RFC4585], but MAY choose to ignore this feedback
(following Section 4.2 of [RFC4585]). Receivers MAY send NACKs for (following Section 4.2 of [RFC4585]). Receivers MAY send NACKs for
missing RTP packets; [RFC4585] provides some guidelines on when to missing RTP packets; [RFC4585] provides some guidelines on when to
send NACKs. It is not expected that a receiver will send a NACK for send NACKs. It is not expected that a receiver will send a NACK for
every lost RTP packet, rather it should consider the cost of sending every lost RTP packet, rather it needs to consider the cost of
NACK feedback, and the importance of the lost packet, to make an sending NACK feedback, and the importance of the lost packet, to make
informed decision on whether it is worth telling the sender about a an informed decision on whether it is worth telling the sender about
packet loss event. a packet loss event.
The RTP Retransmission Payload Format [RFC4588] offers the ability to The RTP Retransmission Payload Format [RFC4588] offers the ability to
retransmit lost packets based on NACK feedback. Retransmission needs retransmit lost packets based on NACK feedback. Retransmission needs
to be used with care in interactive real-time applications to ensure to be used with care in interactive real-time applications to ensure
that the retransmitted packet arrives in time to be useful, but can that the retransmitted packet arrives in time to be useful, but can
be effective in environments with relatively low network RTT (an RTP be effective in environments with relatively low network RTT (an RTP
sender can estimate the RTT to the receivers using the information in sender can estimate the RTT to the receivers using the information in
RTCP SR and RR packets). The use of retransmissions can also RTCP SR and RR packets). The use of retransmissions can also
increase the forward RTP bandwidth, and can potentially worsen the increase the forward RTP bandwidth, and can potentially worsen the
problem if the packet loss was caused by network congestion. We problem if the packet loss was caused by network congestion. We
note, however, that retransmission of an important lost packet to note, however, that retransmission of an important lost packet to
repair decoder state may be lower cost than sending a full intra repair decoder state can have lower cost than sending a full intra
frame. It is not appropriate to blindly retransmit RTP packets in frame. It is not appropriate to blindly retransmit RTP packets in
response to a NACK. The importance of lost packets and the response to a NACK. The importance of lost packets and the
likelihood of them arriving in time to be useful needs to be likelihood of them arriving in time to be useful needs to be
considered before RTP retransmission is used. considered before RTP retransmission is used.
Receivers are REQUIRED to implement support for RTP retransmission Receivers are REQUIRED to implement support for RTP retransmission
packets [RFC4588]. Senders MAY send RTP retransmission packets in packets [RFC4588]. Senders MAY send RTP retransmission packets in
response to NACKs if the RTP retransmission payload format has been response to NACKs if the RTP retransmission payload format has been
negotiated for the session, and if the sender believes it is useful negotiated for the session, and if the sender believes it is useful
to send a retransmission of the packet(s) referenced in the NACK. An to send a retransmission of the packet(s) referenced in the NACK. An
RTP sender is not expected to retransmit every NACKed packet. RTP sender is not expected to retransmit every NACKed packet.
6.2. Forward Error Correction (FEC) 6.2. Forward Error Correction (FEC)
The use of Forward Error Correction (FEC) can provide an effective The use of Forward Error Correction (FEC) can provide an effective
protection against some degree of packet loss, at the cost of steady protection against some degree of packet loss, at the cost of steady
bandwidth overhead. There are several FEC schemes that are defined bandwidth overhead. There are several FEC schemes that are defined
for use with RTP. Some of these schemes are specific to a particular for use with RTP. Some of these schemes are specific to a particular
RTP payload format, others operate across RTP packets and can be used RTP payload format, others operate across RTP packets and can be used
with any payload format. It should be noted that using redundancy with any payload format. It needs to be noted that using redundant
encoding or FEC will lead to increased playout delay, which should be encoding or FEC will lead to increased play out delay, which needs to
considered when choosing the redundancy or FEC formats and their be considered when choosing the redundancy or FEC formats and their
respective parameters. respective parameters.
If an RTP payload format negotiated for use in a WebRTC session If an RTP payload format negotiated for use in a WebRTC session
supports redundant transmission or FEC as a standard feature of that supports redundant transmission or FEC as a standard feature of that
payload format, then that support MAY be used in the WebRTC session, payload format, then that support MAY be used in the WebRTC session,
subject to any appropriate signalling. subject to any appropriate signalling.
There are several block-based FEC schemes that are designed for use There are several block-based FEC schemes that are designed for use
with RTP independent of the chosen RTP payload format. At the time with RTP independent of the chosen RTP payload format. At the time
of this writing there is no consensus on which, if any, of these FEC of this writing there is no consensus on which, if any, of these FEC
schemes is appropriate for use in the WebRTC context. Accordingly, schemes is appropriate for use in the WebRTC context. Accordingly,
this memo makes no recommendation on the choice of block-based FEC this memo makes no recommendation on the choice of block-based FEC
for WebRTC use. for WebRTC use.
7. WebRTC Use of RTP: Rate Control and Media Adaptation 7. WebRTC Use of RTP: Rate Control and Media Adaptation
WebRTC will be used in very varied network environment with a WebRTC will be used in heterogeneous network environments using a
heterogeneous set of link technologies, including wired and wireless, variety set of link technologies, including both wired and wireless
interconnecting peers at different topological locations resulting in links, to interconnect potentially large groups of users around the
network paths with widely varying one way delays, bit-rate capacity, world. As a result, the network paths between users can have widely
load levels and traffic mixes. In addition, individual end-points varying one-way delays, available bit-rates, load levels, and traffic
will open one or more WebRTC sessions between one or more peers. mixtures. Individual end-points can open one or more RTP sessions to
Each of these session may contain different mixes of media and data each participant in a WebRTC conference, and there can be several
flows. Asymmetric usage of media bit-rates and number of RTP media participants. Each of these RTP sessions can contain different types
streams is also to be expected. A single end-point may receive zero of media, and the type of media, bit rate, and number of flows can be
to many simultaneous RTP media streams while itself transmitting one highly asymmetric. Non-RTP traffic can share the network paths RTP
or more streams. flows. Since the network environment is not predictable or stable,
WebRTC endpoints MUST ensure that the RTP traffic they generate can
The WebRTC application is very dependent from a quality perspective adapt to match changes in the available network capacity.
on the media adaptation working well so that an end-point doesn't
transmit significantly more than the path is capable of handling. If
it would, the result would be high levels of packet loss or delay
spikes causing media quality degradation.
WebRTC applications using more than a single RTP media stream of any
media type or data flows have an additional concern. In this case,
the different flows should try to avoid affecting each other
negatively. In addition, in case there is a resource limitation, the
available resources need to be shared. How to share them is
something the application should prioritize so that the limitations
in quality or capabilities are those that have the least impact on
the application.
Overall, the diversity of operating environments lead to the need for
functionality that adapts to the available capacity and that competes
fairly with other network flows. If it would not compete fairly
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
create traffic across the links in question. A possible attack
scenario is to use a web-service capable of attracting large numbers
of end-points, combined with BGP routing state to have the server
pick client pairs to drive traffic to specific paths.
The above clearly motivates the need for a well working media
adaptation mechanism. This mechanism also have a number of
requirements on what services it should provide and what performance
it needs to provide.
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
a need for the IETF to produce such a specification. Therefore, the
suggested way forward is to specify requirements on any solution for
the media adaptation. For now, we propose that these requirements be
documented in this specification. In addition, a proposed detailed
solution will be developed, but is expected to take longer time to
finalize than this document.
7.1. Congestion Control Requirements
Requirements for congestion control of WebRTC sessions are discussed
in [I-D.jesup-rtp-congestion-reqs].
Implementations are REQUIRED to implement the RTP circuit breakers The quality of experience for users of WebRTC implementation is very
described in [I-D.perkins-avtcore-rtp-circuit-breakers]. dependent on effective adaptation of the media to the limitations of
the network. End-points have to be designed so they do not transmit
significantly more data than the network path can support, except for
very short time periods, otherwise high levels of network packet loss
or delay spikes will occur, causing media quality degradation. The
limiting factor on the capacity of the network path might be the link
bandwidth, or it might be competition with other traffic on the link
(this can be non-WebRTC traffic, traffic due to other WebRTC flows,
or even competition with other WebRTC flows in the same session).
(tbd: Should add the RTP/RTCP Mechanisms that an WebRTC An effective media congestion control algorithm is therefore an
implementation is required to support. Potential candidates include essential part of the WebRTC framework. However, at the time of this
Transmission Timestamps (RFC 5450).) writing, there is no standard congestion control algorithm that can
be used for interactive media applications such as WebRTC flows.
Some requirements for congestion control algorithms for WebRTC
sessions are discussed in [I-D.jesup-rtp-congestion-reqs], and it is
expected that a future version of this memo will mandate the use of a
congestion control algorithm that satisfies these requirements.
7.2. Rate Control Boundary Conditions 7.1. Boundary Conditions and Circuit Breakers
The session establishment signalling will establish certain boundary In the absence of a concrete congestion control algorithm, all WebRTC
that the media bit-rate adaptation can act within. First of all the implementations MUST implement the RTP circuit breaker algorithm that
set of media codecs provide practical limitations in the supported is in described [I-D.ietf-avtcore-rtp-circuit-breakers]. The circuit
bit-rate span where it can provide useful quality, which breaker defines a conservative boundary condition for safe operation,
packetization choices that exist. Next the signalling can establish chosen such that applications that trigger the circuit breaker will
maximum media bit-rate boundaries using SDP b=AS or b=CT. almost certainly be causing severe network congestion. Any future
RTP congestion control algorithms are expected to operate within the
envelope allowed by the circuit breaker.
(tbd: This section needs expanding on how to use these limits) The session establishment signalling will also necessarily establish
boundaries to which the media bit-rate will conform. The choice of
media codecs provides upper- and lower-bounds on the supported bit-
rates that the application can utilise to provide useful quality, and
the packetization choices that exist. In addition, the signalling
channel can establish maximum media bit-rate boundaries using the SDP
"b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media
Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo).
The combination of media codec choice and signalled bandwidth limits
SHOULD be used to limit traffic based on known bandwidth limitations,
for example the capacity of the edge links, to the extent possible.
7.3. RTCP Limitations for Congestion Control 7.2. RTCP Limitations for Congestion Control
Experience with the congestion control algorithms of TCP [RFC5681], Experience with the congestion control algorithms of TCP [RFC5681],
TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
that feedback on packet arrivals needs to be sent roughly once per that feedback on packet arrivals needs to be sent roughly once per
round trip time. We note that the real-time media traffic may not round trip time. We note that the real-time media traffic might not
have to adapt to changing path conditions as rapidly as needed for have to adapt to changing path conditions as rapidly as needed for
the elastic applications TCP was designed for, but frequent feedback the elastic applications TCP was designed for, but frequent feedback
is still required to allow the congestion control algorithm to track is still needed to allow the congestion control algorithm to track
the path dynamics. the path dynamics.
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 RTP media stream bit rate thus limits the maximum feedback rate The RTP media stream bit rate thus limits the maximum feedback rate
as a 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 might 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 can 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 might need to be
-- to be defined in a suitable congestion control framework as noted reported -- to be defined in a suitable congestion control framework
above -- which, in turn, increase the report size again. For as noted above -- which, in turn, increase the report size again.
example, different RTCP XR report blocks (jointly) provide the For 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
algorithms, but the (compound) report size grows quickly. algorithms, but the (compound) report size grows quickly.
In group communication, the share of RTCP bandwidth needs to be In group communication, the share of RTCP bandwidth needs to be
shared by all group members, reducing the capacity and thus the shared by all group members, reducing the capacity and thus the
reporting frequency per node. reporting frequency per node.
Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP
bandwidth, split across two entities in a point-to-point session. An bandwidth, split across two entities in a point-to-point session. An
endpoint could thus send a report of 100 bytes about every 70ms or endpoint could thus send a report of 100 bytes about every 70ms or
for every other frame in a 30 fps video. for every other frame in a 30 fps video.
7.4. Congestion Control Interoperability With Legacy Systems 7.3. Congestion Control Interoperability With Legacy Systems
There are legacy implementations that do not implement RTCP, and There are legacy implementations that do not implement RTCP, and
hence do not provide any congestion feedback. Congestion control hence do not provide any congestion feedback. Congestion control
cannot be performed with these end-points. WebRTC implementations cannot be performed with these end-points. WebRTC implementations
that must interwork with such end-points MUST limit their that need to interwork with such end-points MUST limit their
transmission to a low rate, equivalent to a VoIP call using a low transmission to a low rate, equivalent to a VoIP call using a low
bandwidth codec, that is unlikely to cause any significant bandwidth codec, that is unlikely to cause any significant
congestion. congestion.
When interworking with legacy implementations that support RTCP using When interworking with legacy implementations that support RTCP using
the RTP/AVP profile [RFC3551], congestion feedback is provided in the RTP/AVP profile [RFC3551], congestion feedback is provided in
RTCP RR packets every few seconds. Implementations that are required RTCP RR packets every few seconds. Implementations that have to
to interwork with such end-points MUST ensure that they keep within interwork with such end-points MUST ensure that they keep within the
the RTP circuit breaker [I-D.perkins-avtcore-rtp-circuit-breakers] RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers]
constraints to limit the congestion they can cause. constraints to limit the congestion they can cause.
If a legacy end-point supports RTP/AVPF, this enables negotiation of If a legacy end-point supports RTP/AVPF, this enables negotiation of
important parameters for frequent reporting, such as the "trr-int" important parameters for frequent reporting, such as the "trr-int"
parameter, and the possibility that the end-point supports some parameter, and the possibility that the end-point supports some
useful feedback format for congestion control purpose such as TMMBR useful feedback format for congestion control purpose such as TMMBR
[RFC5104]. Implementations that are required to interwork with such [RFC5104]. Implementations that have to interwork with such end-
end-points MUST ensure that they stay within the RTP circuit breaker points MUST ensure that they stay within the RTP circuit breaker
[I-D.perkins-avtcore-rtp-circuit-breakers] constraints to limit the [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the
congestion they can cause, but may find that they can achieve better congestion they can cause, but might find that they can achieve
congestion response depending on the amount of feedback that is better congestion response depending on the amount of feedback that
available. 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 metrics like RTCP does contains a basic set of RTP flow monitoring metrics like
packet loss and jitter. There are a number of extensions that could packet loss and jitter. There are a number of extensions that could
be included in the set to be supported. However, in most cases which be included in the set to be supported. However, in most cases which
RTP monitoring that is needed depends on the application, which makes RTP monitoring that is needed depends on the application, which makes
it difficult to select which to include when the set of applications it difficult to select which to include when the set of 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 needs to be considered
the application to gather the measurements of interest. However, allowing the application to gather the measurements of interest.
security implications for the different data sets exposed will need However, security implications for the different data sets exposed
to be considered in this. will need to be considered in this.
(tbd: If any RTCP XR metrics should be added is still an open (tbd: If any RTCP XR metrics need to be added is still an open
question, but possible to extend at a later stage) 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 will adopt RTP
extensions that are of general utility, to enable easy gatewaying to extensions that are of general utility, to enable easy implementation
other applications using RTP, rather than adopt mechanisms that are of a gateway to other applications using RTP, rather than adopt
narrowly targeted at specific WebRTC use cases. mechanisms that are narrowly targeted at specific WebRTC use cases.
10. Signalling Considerations 10. Signalling Considerations
RTP is built with the assumption of an external signalling channel RTP is built with the assumption 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 RTP/SAVPF profile this can be inferred requires the usage of the RTP/SAVPF profile this can be inferred
as there is only a single profile, but in SDP this is still as there is only a single profile, but in SDP this is still
required information to be signalled. Interworking functions may information that has to be signalled. Interworking functions
transform this into RTP/SAVP for a legacy use case by indicating might transform this into RTP/SAVP for a legacy use case by
to the WebRTC end-point a RTP/SAVPF end-point and limiting the indicating to the WebRTC end-point a RTP/SAVPF end-point and
usage of the a=rtcp attribute to indicate a trr-int value of 4 limiting the usage of the a=rtcp attribute to indicate a trr-int
seconds. value of 4 seconds.
Transport Information: Source and destination IP address(s) and Transport Information: Source and destination IP address(s) and
ports for RTP and RTCP MUST be signalled for each RTP session. In ports for RTP and RTCP MUST be signalled for each RTP session. In
WebRTC these transport addresses will be provided by ICE that WebRTC these transport addresses will be provided by ICE that
signals candidates and arrives at nominated candidate address signals candidates and arrives at nominated candidate address
pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such
that a single port is used for RTP and RTCP flows, this MUST be that a single port is used for RTP and RTCP flows, this MUST be
signalled (see Section 4.5). If several RTP sessions are to be signalled (see Section 4.5). If several RTP sessions are to be
multiplexed onto a single transport layer flow, this MUST also be multiplexed onto a single transport layer flow, this MUST also be
signalled (see Section 4.4). signalled (see Section 4.4).
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that the other end-point will ignore. But for certain mechanisms that the other end-point will ignore. But for certain mechanisms
there is requirement for this to happen as interoperability there is requirement for this to happen as interoperability
failure otherwise happens. failure otherwise happens.
RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the
end-points will be necessary. This SHALL be done as described in end-points will be necessary. This SHALL be done as described in
"Session Description Protocol (SDP) Bandwidth Modifiers for RTP "Session Description Protocol (SDP) Bandwidth Modifiers for RTP
Control Protocol (RTCP) Bandwidth" [RFC3556], or something Control Protocol (RTCP) Bandwidth" [RFC3556], or something
semantically equivalent. This also ensures that the end-points semantically equivalent. This also ensures that the end-points
have a common view of the RTCP bandwidth, this is important as too have a common view of the RTCP bandwidth, this is important as too
different view of the bandwidths may lead to failure to different view of the bandwidths can lead to failure to
interoperate. interoperate.
These parameters are often expressed in SDP messages conveyed within These parameters are often expressed in SDP messages conveyed within
an offer/answer exchange. RTP does not depend on SDP or on the an offer/answer exchange. RTP does not depend on SDP or on the
offer/answer model, but does require all the necessary parameters to offer/answer model, but does require all the necessary parameters to
be agreed upon, and provided to the RTP implementation. We note that be agreed upon, and provided to the RTP implementation. We note that
in the WebRTC context it will depend on the signalling model and API in the WebRTC context it will depend on the signalling model and API
how these parameters need to be configured but they will be need to how these parameters need to be configured but they will be need to
either set in the API or explicitly signalled between the peers. either set in the API or explicitly signalled between the peers.
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there can be multiple different WebRTC MediaStreams containing a there can be multiple different WebRTC MediaStreams containing a
given Track (SSRC). To avoid unnecessary duplication of media at the given Track (SSRC). To avoid unnecessary duplication of media at the
transport level in such cases, a need arises for a binding defining transport level in such cases, a need arises for a binding defining
which WebRTC MediaStreams a given SSRC is associated with at the which WebRTC MediaStreams a given SSRC is associated with at the
signalling level. signalling level.
A proposal for how the binding between WebRTC MediaStreams and SSRC A proposal for how the binding between WebRTC MediaStreams and SSRC
can be done is specified in "Cross Session Stream Identification in can be done is specified in "Cross Session Stream Identification in
the Session Description Protocol" [I-D.alvestrand-rtcweb-msid]. the Session Description Protocol" [I-D.alvestrand-rtcweb-msid].
(tbd: This text must be improved and achieved consensus on. Interim (tbd: This text needs to be improved and achieved consensus on.
meeting in June 2012 shows large differences in opinions.) 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, needed by decisions made, or to enable use cases raised and
and their motivations. This discussion is from an WebRTC end-point their motivations. This discussion is from an WebRTC end-point
perspective. It will occasionally talk about central nodes, but as perspective. It will occasionally talk about central nodes, but as
this specification is for an end-point, this 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 see 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
RTP media streams being sent within the RTP session. A RTP media streams being sent within the RTP session. A
PeerConnection being a point-to-point association between an end- PeerConnection being a point-to-point association between an end-
point and another node. That peer node may be both an end-point or point and another node. That peer node can be both an end-point or
centralized processing node of some type; thus, the RTP session may centralized processing node of some type; thus, the RTP session can
terminate immediately on the far end of the PeerConnection, but it terminate immediately on the far end of the PeerConnection, but it
may also continue as further discussed below in Multiparty might also continue as further discussed below in Multiparty
(Section 12.3) and 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 interworking and network prioritization (Section 12.9) based on
flows, a WebRTC end-point needs to support a mode of operation where flows, a WebRTC end-point needs to support a mode of operation where
one RTP session per media type is used. Currently, each RTP session one RTP session per media type is used. Currently, each RTP session
must use its own UDP flow. Discussions are ongoing if a solution has to use its own UDP flow. Discussions are ongoing if a solution
enabling multiple RTP sessions over a single UDP flow, see enabling multiple RTP sessions over a single UDP flow, see
Section 4.4. Section 4.4.
The multi-unicast- or mesh-based multi-party topology (Figure 1) is a The multi-unicast- or mesh-based multi-party topology (Figure 1) is a
good example for this section as it concerns 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
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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
different 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 limitations 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 might have multiple cameras, microphones or audio
inputs and thus a single end-point can source multiple RTP media inputs and thus a single end-point can source multiple RTP media
streams of the same media type concurrently. Even if an end-point streams of the same media type concurrently. Even if an end-point
does not have multiple media sources of the same media type it will does not have multiple media sources of the same media type it has to
be required to support transmission using multiple SSRCs concurrently support transmission using multiple SSRCs concurrently in the same
in the same RTP session. This is due to the requirement on an WebRTC RTP session. This is due to the requirement on an WebRTC end-point
end-point to support multiple media types in one RTP session. For to support multiple media types in one RTP session. For example, one
example, one audio and one video source can result in the end-point audio and one video source can result in the end-point sending with
sending with two different SSRCs in the same RTP session. As multi- two different SSRCs in the same RTP session. As multi-party
party conferences are supported, as discussed below in Section 12.3, conferences are supported, as discussed below in Section 12.3, a
a WebRTC end-point will need to be capable of receiving, decoding and WebRTC end-point will need to be capable of receiving, decoding and
playout multiple RTP media streams of the same type concurrently. playout multiple RTP media streams of the same type concurrently.
tbd: Are any mechanism needed to signal limitations in the number of tbd: Are any mechanism needed to signal limitations in the number of
SSRC that an end-point can handle? active SSRC that an end-point can handle?
12.3. Multiparty 12.3. Multiparty
There are numerous situations and clear use cases for WebRTC There are numerous situations and clear use cases for WebRTC
supporting RTP sessions supporting multi-party. This can be realized supporting RTP sessions supporting multi-party. This can be realized
in a number of ways using a number of different implementation in a number of ways using a number of different implementation
strategies. In the following, the focus is on the different set of strategies. In the following, the focus is on the different set of
WebRTC end-point requirements that arise from different sets of WebRTC end-point requirements that arise from different sets of
multi-party topologies. multi-party topologies.
The multi-unicast mesh (Figure 1)-based multi-party topology The multi-unicast mesh (Figure 1)-based multi-party topology
discussed above provides a non-centralized solution but may incur a discussed above provides a non-centralized solution but can incur a
heavy tax on the end-points' outgoing paths. It may also consume heavy tax on the end-points' outgoing paths. It can also consume
large amount of encoding resources if each outgoing stream is large amount of encoding resources if each outgoing stream is
specifically encoded. If an encoding is transmitted to multiple specifically encoded. If an encoding is transmitted to multiple
parties, as in some implementations of the mesh case, a requirement parties, as in some implementations of the mesh case, a requirement
on the end-point becomes to be able to create RTP media streams on the end-point becomes to be able to create RTP media streams
suitable for multiple destinations requirements. These requirements suitable for multiple destinations requirements. These requirements
may both be dependent on transport path and the different end-points can both be dependent on transport path and the 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 RTP media streams from certain perspectives, either transmission of RTP media streams from certain perspectives, either
by only sending some of the received RTP media stream to any given by only sending some of the received RTP media stream to any given
receiver or by providing a combined RTP media stream out of a set of receiver or by providing a combined RTP media stream out of a set of
contributing streams. There are various methods of implementation as contributing streams. There are various methods of implementation as
discussed in Appendix A.3. A common aspect is that these central discussed in Appendix A.3. A common aspect is that these central
nodes may use a number of tools to control the media encoding nodes can use a number of tools to control the media encoding
provided by a WebRTC end-point. This includes functions like provided by a WebRTC end-point. This includes functions like
requesting breaking the encoding chain and have the encoder produce a requesting breaking the encoding chain and have the encoder produce a
so called Intra frame. Another is limiting the bit-rate of a given so called Intra frame. Another is limiting the bit-rate of a given
stream to better suit the mixer view of the multiple down-streams. stream to better suit the mixer view of the multiple down-streams.
Others are controlling the most suitable frame-rate, picture Others are controlling the most suitable frame-rate, picture
resolution, the trade-off between 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, source identification, manage synchronization congestion control, source identification, manage synchronization
while providing the application with suitable media optimizations. while providing the application with suitable media optimizations.
Mixers also need to be trusted nodes 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 they must be able to decrypt and then towards the end-point(s), thus they need to be able to decrypt and
encrypt it before sending it out. then encrypt it before sending it out.
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., resolve the support for detecting and handling SSRC collisions, i.e., resolve the
conflict when two different end-points use the same SSRC value. This conflict when two different end-points use the same SSRC value. This
requirement also applies to WebRTC end-points. There are several requirement also applies to WebRTC end-points. There are several
scenarios where SSRC collisions may occur. scenarios where SSRC collisions can occur.
In a point-to-point session where each SSRC is associated with either In a point-to-point session where each SSRC is associated with 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 channel, a collision identifier will be announced in the signalling channel, a collision
is less likely to occur due to the information about used SSRCs is less likely to occur due to the information about used SSRCs
provided by Source-Specific SDP Attributes [RFC5576]. Still if both provided by Source-Specific SDP Attributes [RFC5576]. Still if both
end-points start uses an new SSRC identifier prior to having end-points start uses an new SSRC identifier prior to having
signalled it to the peer and received acknowledgement on the signalled it to the peer and received acknowledgement on the
signalling message, there can be collisions. The Source-Specific SDP signalling message, there can be collisions. The Source-Specific SDP
Attributes [RFC5576] contains no mechanism to resolve SSRC collisions Attributes [RFC5576] contains no mechanism to resolve SSRC collisions
or reject a end-points usage of an SSRC. or reject a end-points usage of an SSRC.
There could also appear unsignalled SSRCs. This is more likely than There could also appear SSRC values that are not signalled. This is
it appears as certain RTP functions need extra SSRCs to provide more likely than it appears as certain RTP functions need extra SSRCs
functionality related to another (the "main") SSRC, for example, SSRC to provide functionality related to another (the "main") SSRC, for
multiplexed RTP retransmission [RFC4588]. In those cases, an end- example, SSRC multiplexed RTP retransmission [RFC4588]. In those
point can create a new SSRC that strictly doesn't need to be cases, an end-point can create a new SSRC that strictly doesn't need
announced over the signalling channel to function correctly on both to be announced over the signalling channel to function correctly on
RTP and PeerConnection level. both RTP and PeerConnection level.
The more likely case for SSRC collision is that multiple end-points The more likely case for SSRC collision is that multiple end-points
in a multiparty conference create new sources and signals those in a multiparty conference create new sources and signals those
towards the central server. In cases where the SSRC/CSRC are towards the central server. In cases where the SSRC/CSRC are
propagated between the different end-points from the central node propagated between the different end-points from the central node
collisions can occur. collisions can occur.
Another scenario is when the central node manages to connect an end- Another scenario is when the central node manages to connect an end-
point's PeerConnection to another PeerConnection the end-point point's PeerConnection to another PeerConnection the end-point
already has, thus forming a loop where the end-point will receive its already has, thus forming a loop where the end-point will receive its
skipping to change at page 27, line 45 skipping to change at page 26, line 45
Contributing Sources (CSRC) is a functionality in the RTP header that Contributing Sources (CSRC) is a functionality in the RTP header that
allows an RTP node to combine media packets from multiple sources allows an RTP node to combine media packets from multiple sources
into one and to identify which sources yielded the result. For into one and to identify which sources yielded the result. For
WebRTC end-points, supporting contributing sources is trivial. The WebRTC end-points, supporting contributing sources is trivial. The
set of CSRCs is provided in a given RTP packet. This information can set of CSRCs is provided in a given RTP packet. This information can
then be exposed to the applications using some form of API, possibly then be exposed to the applications using some form of API, possibly
a mapping back into WebRTC MediaStream identities to avoid having to a mapping back into WebRTC MediaStream identities to avoid having to
expose two namespaces and the handling of SSRC collision handling to expose two namespaces and the handling of SSRC collision handling to
the JavaScript. the JavaScript.
(tbd: should the API provide the ability to add a CSRC list to an (tbd: does the API need to provide the ability to add a CSRC list to
outgoing packet? this is only useful if the sender is mixing content) an outgoing packet? this is only useful if the sender is mixing
content)
There are also at least one extension that depends on the CRSRC list There are also at least one extension that depends on the CSRC list
being used: the Mixer-to-client audio level [RFC6465], which enhances being used: the Mixer-to-client audio level [RFC6465], which enhances
the information provided by the CSRC to actual energy levels for the information provided by the CSRC to actual energy levels for
audio for each contributing source. audio for each contributing source.
12.6. Media Synchronization 12.6. Media Synchronization
When an end-point sends media from more than one media source, it When an end-point sends media from more than one media source, it
needs to consider if (and which of) these media sources are to be needs to consider if (and which of) these media sources are to be
synchronized. In RTP/RTCP, synchronisation is provided by having a synchronized. In RTP/RTCP, synchronisation is provided by having a
set of RTP media streams be indicated as coming from the same set of RTP media streams be indicated as coming from the same
skipping to change at page 28, line 30 skipping to change at page 27, line 31
receiver and, if desired, the streams can be synchronized. The receiver and, if desired, the streams can be synchronized. The
requirement is for the media sender to provide the correlation requirement is for the media sender to provide the correlation
information; it is up to the receiver to use it or not. information; it is up to the receiver to use it or not.
12.7. Multiple RTP End-points 12.7. Multiple RTP End-points
Some usages of RTP beyond the recommend topologies result in that an Some usages of RTP beyond the recommend topologies result in that an
WebRTC end-point sending media in an RTP session out over a single WebRTC end-point sending media in an RTP session out over a single
PeerConnection will receive receiver reports from multiple RTP PeerConnection will receive receiver reports from multiple RTP
receivers. Note that receiving multiple receiver reports is expected receivers. Note that receiving multiple receiver reports is expected
because any RTP node that has multiple SSRCs is required to report to because any RTP node that has multiple SSRCs has to report to the
the media sender. The difference here is that they are multiple media sender. The difference here is that they are multiple nodes,
nodes, and thus will likely have different path characteristics. and thus will likely have different path characteristics.
RTP Mixers may create a situation where an end-point experiences a RTP Mixers can create a situation where an end-point experiences a
situation in-between a session with only two end-points and multiple situation in-between a session with only two end-points and multiple
end-points. Mixers are expected to not forward RTCP reports end-points. Mixers are expected to not forward RTCP reports
regarding RTP media streams across themselves. This is due to the regarding RTP media streams across themselves. This is due to the
difference in the RTP media streams provided to the different end- difference in the RTP media streams provided to the different end-
points. The original media source lacks information about a mixer's points. The original media source lacks information about a mixer's
manipulations prior to sending it the different receivers. This manipulations prior to sending it the different receivers. This
setup also results in that an end-point's feedback or requests goes scenario also results in that an end-point's feedback or requests
to the mixer. When the mixer can't act on this by itself, it is 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 forced to go to the original media source to fulfil the receivers
request. This will not necessarily be explicitly visible any RTP and request. This will not necessarily be explicitly visible any RTP and
RTCP traffic, but the interactions and the time to complete them will RTCP traffic, but the interactions and the time to complete them will
indicate such dependencies. indicate such dependencies.
The topologies in which an end-point receives receiver reports from The topologies in which an end-point receives receiver reports from
multiple other end-points are the centralized relay, multicast and an multiple other end-points are the centralized relay, multicast and an
end-point forwarding an RTP media stream. Having multiple RTP nodes end-point forwarding an RTP media stream. Having multiple RTP nodes
receive an RTP flow and send reports and feedback about it has receive an RTP flow and send reports and feedback about it has
several impacts. As previously discussed (Section 12.3) any codec several impacts. As previously discussed (Section 12.3) any codec
control and rate control needs to be capable of merging the control and rate control needs to be capable of merging the
skipping to change at page 29, line 41 skipping to change at page 28, line 43
feed the same set of WebRTC MediaStreams. Another method is to use feed the same set of WebRTC MediaStreams. Another method is to use
multiple WebRTC MediaStreams that are differently configured when it multiple WebRTC MediaStreams that are differently configured when it
comes to the media parameters. This would result in that multiple comes to the media parameters. This would result in that multiple
different RTP Media Streams (SSRCs) being in used with different different RTP Media Streams (SSRCs) being in used with different
encoding based on the same media source (camera, microphone). encoding based on the same media source (camera, microphone).
When intending to use simulcast it is important that this is made When intending to use simulcast it is important that this is made
explicit so that the end-points don't automatically try to optimize explicit so that the end-points don't automatically try to optimize
away the different encodings and provide a single common version. away the different encodings and provide a single common version.
Thus, some explicit indications that the intent really is to have Thus, some explicit indications that the intent really is to have
different media encodings is likely required. It should be noted different media encodings is likely needed. It is to be noted that
that it might be a central node, rather than an WebRTC end-point that it might be a central node, rather than an WebRTC end-point that
would benefit from receiving simulcasted media sources. would benefit from receiving simulcast media sources.
tbd: How to perform simulcast needs to be determined and the tbd: How to perform simulcast needs to be determined and the
appropriate API or signalling for its usage needs to be defined. appropriate API or signalling for its usage needs to be defined.
12.9. Differentiated Treatment of Flows 12.9. Differentiated Treatment of Flows
There are use cases for differentiated treatment of RTP media There are use cases for differentiated treatment of RTP media
streams. Such differentiation can happen at several places in the streams. Such differentiation can happen at several places in the
system. First of all is the prioritization within the end-point system. First of all is the prioritization within the end-point
sending the media, which controls, both which RTP media streams that sending the media, which controls, both which RTP media streams that
will be sent, and their allocation of bit-rate out of the current will be sent, and their allocation of bit-rate out of the current
available aggregate as determined by the congestion control. available aggregate as determined by the congestion control.
Secondly, the network can prioritize packet flows, including RTP Secondly, the network can prioritize packet flows, including RTP
media streams. Typically, differential treatment includes two steps, media streams. Typically, differential treatment includes two steps,
the first being identifying whether an IP packet belongs to a class the first being identifying whether an IP packet belongs to a class
which should be treated differently, the second the actual mechanism that has to be treated differently, the second the actual mechanism
to prioritize packets. This is done according to three methods; to prioritize packets. This is done according to three methods;
Diffserv: The end-point marks a packet with a diffserv code point to DiffServ: The end-point marks a packet with a DiffServ code point to
indicate to the network that the packet belongs to a particular indicate to the network that the packet belongs to a particular
class. class.
Flow based: Packets that shall be given a particular treatment are Flow based: Packets that need to be given a particular treatment are
identified using a combination of IP and port address. identified using a combination of IP and port address.
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 because multiple types of flows are aggregated and thus it issues because multiple types of flows are aggregated and thus it
becomes more difficult to analyse them. 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, i.e., relative prioritization is not possible. Moreover, 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 might 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 it so that it can provide the separation of the needs to know about it so that it can provide the separation of the
RTP 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 assumes that either the end-point or a classifier can mark DiffServ assumes that either the end-point or a classifier can mark
the packets with an appropriate DSCP so that the packets are 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
two requirements arise in the WebRTC context: 1) The WebRTC two requirements arise in the WebRTC context: 1) The WebRTC
application or browser has to know which DSCP to use and that it can application or browser has to know which DSCP to use and that it can
use them on some set of RTP media streams. 2) The information needs use them on some set of RTP media streams. 2) The information needs
to be propagated to the operating system when transmitting the to be propagated to the operating system when transmitting the
packet. packet. These issues are discussed in DSCP and other packet markings
for RTCWeb QoS [I-D.ietf-rtcweb-qos].
tbd: The model for providing differentiated treatment needs to be tbd: The model for providing differentiated treatment needs to be
evolved. This includes: evolved. Most of this is not the responsibility of this memo.
However, this memo could include:
1. How the application can prioritize MediaStreamTracks differently 1. How can the application can prioritize MediaStreamTracks
in the API differently in the API?
2. How the browser or application determine availability of 2. How MediaStreamTrack prioritization maps to the RTP level, and
transport differentiation what type of marking behaviour can occur on the RTP media stream
and its datagram?
3. How to learn about any configuration information for transport 13. Open Issues
differentiation, such as DSCPs.
13. IANA Considerations This section contains a summary of the open issues or to be done
things noted in the document:
1. Need to add references to the RTP payload format for the Video
Codec chosen in Section 4.3.
2. The methods and solutions for RTP multiplexing over a single
transport is not yet finalized in Section 4.4.
3. RTP congestion control algorithms will probably require some
feedback information to be conveyed in RTCP. Are the tools that
are mandated by this memo sufficient, or do we need additional
information?
4. RTP congestion control could be implementing using either a
sender-based algorithm or a receiver-based algorithm. To ensure
interoperability, does this memo need to mandate which end is in
charge of congestion control for a path?
5. Still open if any RTCP XR performance metrics are needed, as
discussed in Section 8.
6. The API mapping to RTP level concepts has to be agreed and
documented in Section 11.
7. An open question if any requirements are needed to agree and
limit the number of simultaneously used media sources (SSRCs)
within an RTP session. See Section 12.2.
8. Is an API needed for expressing any application level media
mixing of an RTP media stream so that the correct CSRC list can
be set as discussed in Section 12.5?
9. The method for achieving simulcast of a media source has to be
decided as discussed in Section 12.8.
10. Possible documentation of what support for differentiated
treatment that are needed on RTP level as the API and the
network level specification matures as discussed in
Section 12.9.
11. Editing of Appendix A to remove redundancy between this and the
update of RTP Topologies
[I-D.westerlund-avtcore-rtp-topologies-update].
14. IANA Considerations
This memo makes no request of IANA. 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 is to be removed on publication as
RFC. an RFC.
14. Security Considerations 15. Security Considerations
RTP and its various extensions each have their own security The security considerations for the WebRTC framework are described in
considerations. These should be taken into account when considering [I-D.ietf-rtcweb-security]. The overall security architecture for
the security properties of the complete suite. We currently don't WebRTC is described in [I-D.ietf-rtcweb-security-arch].
think this suite creates any additional security issues or
properties. The use of SRTP [RFC3711] will provide protection or
mitigation against most of the fundamental issues by offering
confidentiality, integrity and partial source authentication. A
mandatory to implement media security solution will be required to be
picked. We currently don't discuss the key-management aspect of SRTP
in this memo, that needs to be done taking the WebRTC communication
model into account.
Privacy concerns are under discussion and the generation of non- The security considerations of the RTP specification, the RTP/SAVPF
trackable CNAMEs are under discussion. profile, and the various RTP/RTCP extensions and RTP payload formats
that form the complete protocol suite described in this memo apply.
We do not believe there are any new security considerations resulting
from the combination of these various protocol extensions.
The guidelines in [RFC6562] apply when using variable bit rate (VBR) The Extended Secure RTP Profile for Real-time Transport Control
audio codecs, for example Opus or the Mixer audio level header Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
extensions. handling of fundamental issues by offering confidentiality, integrity
and partial source authentication. A mandatory to implement media
security solution is (tbd).
Security considerations for the WebRTC work are discussed in tbd: Privacy concerns, and the generation of untraceable CNAMEs, are
[I-D.ietf-rtcweb-security]. under discussion.
15. Acknowledgements The guidelines in [RFC6562] apply when using variable bit rate (VBR)
audio codecs, e.g., Opus or the Mixer audio level header extensions.
16. Acknowledgements
The authors would like to thank Harald Alvestrand, Cary Bran, Charles The authors would like to thank Harald Alvestrand, Cary Bran, Charles
Eckel and Cullen Jennings for valuable feedback. Eckel and Cullen Jennings for valuable feedback.
16. References 17. References
16.1. Normative References 17.1. Normative References
[I-D.holmberg-mmusic-sdp-bundle-negotiation] [I-D.holmberg-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
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-rtp-circuit-breakers]
Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
Breakers for Unicast Sessions",
draft-ietf-avtcore-rtp-circuit-breakers-00 (work in
progress), October 2012.
[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-02 (work in
progress), October 2011. progress), July 2012.
[I-D.ietf-avtext-multiple-clock-rates] [I-D.ietf-avtext-multiple-clock-rates]
Petit-Huguenin, M. and G. Zorn, "Support for Multiple Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", Clock Rates in an RTP Session",
draft-ietf-avtext-multiple-clock-rates-05 (work in draft-ietf-avtext-multiple-clock-rates-06 (work in
progress), May 2012. progress), October 2012.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-00 (work in
progress), September 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-04 (work based Applications", draft-ietf-rtcweb-overview-04 (work
in progress), June 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-03 (work in progress), draft-ietf-rtcweb-security-03 (work in progress),
June 2012. June 2012.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "RTCWEB Security Architecture",
draft-ietf-rtcweb-security-arch-05 (work in progress),
October 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.rescorla-avtcore-6222bis]
Perkins, C. and V. Singh, "RTP Congestion Control: Circuit Rescorla, E. and A. Begen, "Guidelines for Choosing RTP
Breakers for Unicast Sessions", Control Protocol (RTCP) Canonical Names (CNAMEs)",
draft-perkins-avtcore-rtp-circuit-breakers-00 (work in draft-rescorla-avtcore-6222bis-00 (work in progress),
progress), March 2012. October 2012.
[I-D.terriberry-avp-codecs]
Terriberry, T., "Update to Recommended Codecs for the AVP
RTP Profile", draft-terriberry-avp-codecs-00 (work in
progress), August 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-04 (work
in progress), March 2012. in progress), October 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
Payload Format Specifications", BCP 36, RFC 2736, Payload Format Specifications", BCP 36, RFC 2736,
December 1999. December 1999.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
skipping to change at page 34, line 22 skipping to change at page 34, line 45
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010. Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010. Flows", RFC 6051, November 2010.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(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 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP", RFC 6562, Variable Bit Rate Audio with Secure RTP", RFC 6562,
March 2012. March 2012.
16.2. Informative References 17.2. Informative References
[I-D.alvestrand-rtcweb-msid] [I-D.alvestrand-rtcweb-msid]
Alvestrand, H., "Cross Session Stream Identification in Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol", the Session Description Protocol",
draft-alvestrand-rtcweb-msid-02 (work in progress), draft-alvestrand-rtcweb-msid-02 (work in progress),
May 2012. May 2012.
[I-D.ietf-avt-srtp-ekt] [I-D.ietf-avt-srtp-ekt]
Wing, D., McGrew, D., and K. Fischer, "Encrypted Key Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03 Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
(work in progress), October 2011. (work in progress), October 2011.
[I-D.ietf-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS",
draft-ietf-rtcweb-qos-00 (work in progress), October 2012.
[I-D.ietf-rtcweb-use-cases-and-requirements] [I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", Time Communication Use-cases and Requirements",
draft-ietf-rtcweb-use-cases-and-requirements-09 (work in draft-ietf-rtcweb-use-cases-and-requirements-09 (work in
progress), June 2012. progress), June 2012.
[I-D.jesup-rtp-congestion-reqs] [I-D.jesup-rtp-congestion-reqs]
Jesup, R. and H. Alvestrand, "Congestion Control Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media", Requirements For Real Time Media",
draft-jesup-rtp-congestion-reqs-00 (work in progress), draft-jesup-rtp-congestion-reqs-00 (work in progress),
March 2012. March 2012.
[I-D.westerlund-avtcore-multiplex-architecture] [I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., and C. Perkins, "RTP Westerlund, M., Burman, B., Perkins, C., and H.
Multiplexing Architecture", Alvestrand, "Guidelines for using the Multiplexing
draft-westerlund-avtcore-multiplex-architecture-01 (work Features of RTP",
in progress), March 2012. draft-westerlund-avtcore-multiplex-architecture-02 (work
in progress), July 2012.
[I-D.westerlund-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies",
draft-westerlund-avtcore-rtp-topologies-update-01 (work in
progress), October 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.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
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.
[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.
skipping to change at page 36, line 18 skipping to change at page 36, line 44
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, June 2011. Control Protocol (RTCP) Flows", RFC 6263, June 2011.
Appendix A. Supported RTP Topologies Appendix A. Supported RTP Topologies
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
[I-D.westerlund-avtcore-rtp-topologies-update], 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 section includes RTP topologies beyond the RECOMMENDED ones.
This in an attempt to highlight the differencies and the in many case
This in an attempt to highlight the differences and the in many case
small differences in implementation to support a larger set of small differences in implementation to support a larger set of
possible topologies. possible topologies.
(tbd: This section needs reworking and clearer relation to
[I-D.westerlund-avtcore-rtp-topologies-update].)
A.1. Point to Point A.1. Point to Point
The point-to-point RTP topology (Figure 3) 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 |
+---+ +---+ +---+ +---+
skipping to change at page 36, line 51 skipping to change at page 37, line 35
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
become problematic, since maintaining multiple NAT bindings can be become problematic, since maintaining multiple NAT bindings can be
costly. It also complicates firewall administration, since multiple costly. It also complicates firewall administration, since multiple
ports must be opened to allow RTP traffic. To reduce these costs and ports need to be opened to allow RTP traffic. To reduce these costs
session setup times, support for multiplexing RTP data packets and and session set-up times, support for multiplexing RTP data packets
RTCP control packets on a single port [RFC5761] will be supported. and RTCP control packets on a single port [RFC5761] will be
supported.
In cases where there is only one type of media (e.g., a voice-only In cases where there is only one type of media (e.g., a voice-only
call) this topology will be implemented as a single RTP session, with call) this topology will be implemented as a single RTP session, with
bidirectional flows of RTP and RTCP packets, all then multiplexed bidirectional flows of RTP and RTCP packets, all then multiplexed
onto a single 5-tuple. If multiple types of media are to be used onto a single 5-tuple. If multiple types of media are to be used
(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
targeted by WebRTC, but they may impact interoperability with legacy targeted by WebRTC, but they can impact interoperability with legacy
systems. systems.
An RTP session have good support for simultanously transport multiple An RTP session have good support for simultaneously transport
media sources. Each media source uses an unique SSRC identifier and multiple media sources. Each media source uses an unique SSRC
each SSRC has independent RTP sequence number and timestamp spaces. identifier and each SSRC has independent RTP sequence number and
This is being utilized in WebRTC for several cases. One is to enable timestamp spaces. This is being utilized in WebRTC for several
multiple media sources of the same type, an end-point that has two cases. One is to enable multiple media sources of the same type, an
video cameras can potentially transmitt video from both to its end-point that has two video cameras can potentially transmit video
peer(s). Another usage is when a single RTP session is being used from both to its peer(s). Another usage is when a single RTP session
for both multiple media types, thus an end-point can transmit both is being used for both multiple media types, thus an end-point can
audio and video to the peer(s). Thirdly to support multi-party cases transmit both audio and video to the peer(s). Thirdly to support
as will be discussed below support for multiple SSRC of the same multi-party cases as will be discussed below support for multiple
media type are required. SSRC of the same media type is needed.
Thus we can introduce a couple of different notiations in the below Thus we can introduce a couple of different notations in the below
two alternate figures of a single peer connection in a a point to two alternate figures of a single peer connection in a point to point
point setup. The first depicting a setup where the peer connection set-up. The first depicting a setup where the peer connection
established has two different RTP sessions, one for audio and one for established has two different RTP sessions, one for audio and one for
video. The second one using a single RTP session. In both cases A video. The second one using a single RTP session. In both cases A
has two video streams to send and one audio stream. B has only one has two video streams to send and one audio stream. B has only one
audio and video stream. These are used to illustrate the relation audio and video stream. These are used to illustrate the relation
between a peerConnection, the UDP flow(s), the RTP session(s) and the between a peerConnection, the UDP flow(s), the RTP session(s) and the
SSRCs that will be used in the later cases also. In the below SSRCs that will be used in the later cases also. In the below
figures RTCP flows are not included. They will flow bi-directionally figures RTCP flows are not included. They will flow bi-directionally
between any RTP session instances in the different nodes. between any RTP session instances in the different nodes.
+-A-------------+ +-B-------------+ +-A-------------+ +-B-------------+
skipping to change at page 39, line 34 skipping to change at page 40, line 34
Figure 5: Point to Point: Single RTP session. Figure 5: Point to Point: Single RTP session.
In (Figure 5) 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) RTP media streams This RTP session carries a total of five (5) RTP media streams
(SSRCs). From A to B there is Audio (AA1) and two video (AV1 and (SSRCs). From A to B there is Audio (AA1) and two video (AV1 and
AV2). From B 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 6); unfortunately not discussed in the RTP topology (Figure 6). 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
skipping to change at page 40, line 19 skipping to change at page 41, line 18
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
for why A would need to know about the B's and C's communication. An for why A would need to know about the B's and C's communication. An
advantage of using separate RTP sessions is that it enables using advantage of using separate RTP sessions is that it enables using
different media bit-rates to the differnt peers, thus not forcing B different media bit-rates to the different peers, thus not forcing B
to endure the same quality reductions if there are limiations in the to endure the same quality reductions if there are limitations in the
transport from A to C as C will. transport from A to C as C will.
+-A------------------------+ +-B-------------+ +-A------------------------+ +-B-------------+
|+---+ +-PeerC1------| |-PeerC1------+ | |+---+ +-PeerC1------| |-PeerC1------+ |
||MIC| | +-UDP1------| |-UDP1------+ | | ||MIC| | +-UDP1------| |-UDP1------+ | |
|+---+ | | +-RTP1----| |-RTP1----+ | | | |+---+ | | +-RTP1----| |-RTP1----+ | | |
| | +----+ | | | +-Audio-| |-Audio-+ | | | | | | +----+ | | | +-Audio-| |-Audio-+ | | | |
| +->|ENC1|--+-+-+-+--->AA1|------------->| | | | | | | +->|ENC1|--+-+-+-+--->AA1|------------->| | | | | |
| | +----+ | | | | |<-------------|BA1 | | | | | | | +----+ | | | | |<-------------|BA1 | | | | |
| | | | | +-------| |-------+ | | | | | | | | | +-------| |-------+ | | | |
skipping to change at page 41, line 5 skipping to change at page 42, line 4
| | | | +-------| |-------+ | | | | | | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | | | | | +---------| |---------+ | | |
| | +-----------| |-----------+ | | | | +-----------| |-----------+ | |
| +-------------| |-------------+ | | +-------------| |-------------+ |
+--------------------------+ +---------------+ +--------------------------+ +---------------+
Figure 7: Session structure 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 7. A's microphone is and RTP sessions that are set-up 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 constrained 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 8. 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 41, line 51 skipping to change at page 42, line 50
| | | +---------| |---------+ | | | | | | +---------| |---------+ | | |
| | +-----------| |-----------+ | | | | +-----------| |-----------+ | |
| +-------------| |-------------+ | | +-------------| |-------------+ |
+------------------------+ +---------------+ +------------------------+ +---------------+
Figure 8: 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 preferred 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 decision logic does arise
several different topologies and implementation. in several different topologies and implementation.
A.3. Mixer Based A.3. Mixer Based
An mixer (Figure 9) 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 needed instead
stead of 3 in the depicted scenario (Figure 9). of 3 in the depicted scenario (Figure 9).
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Mixer | | Mixer |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 9: 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 have two downsides, the first is that the mixer has to be a
trusted node as they either performs media operations or at least trusted node as they either performs media operations or at least re-
repacketize the media. Both type of operations requires when using packetize 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 integrity
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
skipping to change at page 43, line 4 skipping to change at page 43, line 51
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 RTP 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 RTP media stream. Select which that are to be included participants RTP media stream. Select which that are to be included
in a media domain mix of the incomming RTP media streams. Then in a media domain mix of the incoming RTP media streams. Then create
create a single outgoing stream from this mix. 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 decoding N times. Then it needs to produce N or N+1 mixes, the
that different mixes are needed are so that each contributing source reasons that different mixes are needed are so that each contributing
get a mix which don't contain themselves, as this would result in an source get a mix which don't contain themselves, as this would result
echo. When N is lower than the number of all participants one may in an echo. When N is lower than the number of all participants one
produce a Mix of all N streams for the group that are curently not can produce a Mix of all N streams for the group that are curently
included in the mix, thus N+1 mixes. These audio streams are then not included in the mix, thus N+1 mixes. These audio streams are
encoded again, RTP packetized and sent out. then encoded again, RTP packetized and sent out.
Video can't really be "mixed" and produce something particular useful Video can't really be "mixed" and produce something particular useful
for the users, however creating an composition out of the contributed for the users, however creating an composition out of the contributed
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
skipping to change at page 44, line 51 skipping to change at page 45, line 51
| | +-----------| |-----------+ | | +---+ | | | | | +-----------| |-----------+ | | +---+ | | |
| +-------------| |-------------+ | +-----+ | | +-------------| |-------------+ | +-----+ |
+---------------+ |---------------+ | +---------------+ |---------------+ |
+--------------------------------+ +--------------------------------+
Figure 10: 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 10. 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 participants. 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.
The CSRC has in its turn utility in RTP extensions, like the in The CSRC has in its turn utility in RTP extensions, like the in
skipping to change at page 45, line 30 skipping to change at page 46, line 30
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 RTP media streams in party communication session and only generate RTP media streams in
each without carrying on RTP/RTCP level any identity information each without carrying on RTP/RTCP level any identity information
about the contributing sources. This removes both the functionaltiy about the contributing sources. This removes both the functionality
that CSRC can provide and the possibility to use any extensions that that CSRC can provide and the possibility to use any extensions that
build on CSRC and the loop detection. It may appear a simplification build on CSRC and the loop detection. It might appear a
if SSRC collision would occur between two different end-points as simplification if SSRC collision would occur between two different
they can be avoide to be resolved and instead remapped between the end-points as they can be avoided to be resolved and instead remapped
independent sessions if at all exposed. However, SSRC/CSRC remapping between the independent sessions if at all exposed. However, SSRC/
requiresthat SSRC/CSRC are never exposed to the WebRTC javascript CSRC remapping requires that SSRC/CSRC are never exposed to the
client to use as reference. This as they only have local importance WebRTC JavaScript client to use as reference. This as they only have
if they are used on a multi-party session scope the result would be local importance if they are used on a multi-party session scope the
missreferencing. Also SSRC collision handling will still be needed result would be mis-referencing. Also SSRC collision handling will
as it may occur between the mixer and the end-point. still be needed as it can occur between the mixer and the end-point.
Session termination may appear to resolve some issues, it however Session termination might 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
identities and never used directly between multiple end-points. identities and never used directly between multiple end-points.
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 RTP media streams and media from one of the by the mixer received RTP media streams and
forward the media using the mixers own SSRCs. The mixer can then forward the media using the mixers own SSRCs. The mixer can then
switch between available sources if that is required by the concept switch between available sources if that is needed by the concept for
for the source, like currently active speaker. 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 has to 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 is set to 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 might
to be changed if the different PeerConnections have not arrived on have to be changed if the different PeerConnections have not arrived
the same numbering for a given configuration. This also requires on the same numbering for a given configuration. This also requires
that the different end-points do support a common set of codecs, that the different end-points do support a common set of codecs,
otherwise media transcoding for codec compatibility is still otherwise media transcoding for codec compatibility is still needed.
required.
Lets consider the operation of media switching mixer that supports a Lets consider the operation of media switching mixer that supports a
video conference with six participants (A-F) where the two latest video conference with six participants (A-F) where the two latest
speakers in the conference are shown to each participants. Thus the speakers in the conference are shown to each participants. Thus the
mixer has two SSRCs sending video to each peer. mixer has two SSRCs sending video to each peer.
+-A-------------+ +-MIXER--------------------------+ +-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ | | +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | | | | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ | | | | +-RTP1----| |-RTP1------+ | | +-----+ |
skipping to change at page 48, line 16 skipping to change at page 49, line 16
To ensure that a media receiver can correctly decode the RTP media To ensure that a media receiver can correctly decode the RTP media
stream after a switch, it becomes necessary to ensure for state stream after a switch, it becomes necessary to ensure for state
saving codecs that they start from default state at the point of saving codecs that they start from default state at the point of
switching. Thus one common tool for video is to request that the switching. Thus one common tool for video is to request that the
encoding creates an intra picture, something that isn't dependent on encoding creates an intra picture, something that isn't dependent on
earlier state. This can be done using Full Intra Request RTCP codec earlier state. This can be done using Full Intra Request RTCP codec
control 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 considerations 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
session. The mixer can then select which of the potential sources session. The mixer can then select which of the potential sources
that are currently actively transmitting media, despite that the that are currently actively transmitting media, despite that the
mixer in another RTP session recieves media from that end-point. mixer in another RTP session receives media from that end-point.
This is similar to the media switching Mixer but have some important This is similar to the media switching Mixer but have some important
differences in RTP details. differences in RTP details.
+-A-------------+ +-MIXER--------------------------+ +-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ | | +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | | | | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ | | | | +-RTP1----| |-RTP1------+ | | +-----+ |
| | | | +-Video-| |-Video---+ | | | | | | | | | | +-Video-| |-Video---+ | | | | | |
| | | | | AV1|------------>|---------+-+-+-+------->| | | | | | | | AV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|BV1 <----+-+-+-+--------| | | | | | | | |<------------|BV1 <----+-+-+-+--------| | |
skipping to change at page 50, line 11 skipping to change at page 51, line 11
| | +-----------| |-------------+ | +-----+ | | | +-----------| |-------------+ | +-----+ |
| +-------------| |---------------+ | | +-------------| |---------------+ |
+---------------+ +--------------------------------+ +---------------+ +--------------------------------+
Figure 12: Media Projecting Mixer Figure 12: Media Projecting Mixer
So in this six participant conference depicted above in (Figure 12) 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 transmit 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 RTP 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 consecutively 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 independently also thus working around any SSRC collisions by
by having remapping tables between the RTP sessions. However the having remapping tables between the RTP sessions. However the
related WebRTC MediaStream signalling must be correspondlingly related WebRTC MediaStream signalling need to be correspondingly
changed to ensure consistent WebRTC MediaStream to SSRC mappings changed to ensure consistent WebRTC MediaStream to SSRC mappings
between the different PeerConnections and the same comment that between the different PeerConnections and the same comment that
higher functions must not use SSRC as references to RTP media streams higher functions MUST NOT use SSRC as references to RTP media streams
applies also here. 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 coming 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.
skipping to change at page 51, line 22 skipping to change at page 52, line 22
for a legacy end-point or simply relay packets between transport for a legacy end-point or simply relay packets between transport
domains 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 is relevant. In certain legacy deployment media transcoder
media transcoder will be necessary to ensure both codecs and bit-rate will be necessary to ensure both codecs and bit-rate falls within the
falls within the envelope the legacy end-point supports. 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 has to be
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 RTP media stream to the other end-point(s). The transcoded RTP media stream to the other end-point(s). The
congestion control loop can be terminated in the transcoder as the congestion control loop can be terminated in the transcoder as the
media bit-rate being sent by the transcoder can be adjusted media bit-rate being sent by the transcoder can be adjusted
independently of the incoming bit-rate. However, for optimizing independently of the incoming bit-rate. However, for optimizing
performance and resource consumption the translator needs to consider performance and resource consumption the translator needs to consider
what signals or bit-rate reductions it should send towards the source what signals or bit-rate reductions it needs to send towards the
end-point. For example receving a 2.5 mbps video stream and then source end-point. For example receiving a 2.5 Mbps video stream and
send out a 250 kbps video stream after transcoding is a vaste of then send out a 250 kbps video stream after transcoding is a waste of
resources. In most cases a 500 kbps video stream from the source in resources. In most cases a 500 kbps video stream from the source in
the right resolution is likely to provide equal quality after the right resolution is likely to provide equal quality after
transcoding as the 2.5 mbps source stream. At the same time transcoding as the 2.5 Mbps source stream. At the same time
increasing media bit-rate futher than what is needed to represent the increasing media bit-rate further than what is needed to represent
incoming quality accurate is also wasted resources. the 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 52, line 38 skipping to change at page 53, line 38
| | | +---------| |-----------+ | | +---+ | | | | +---------| |-----------+ | | +---+ |
| | +-----------| |-------------+ | | | | +-----------| |-------------+ | |
| +-------------| |---------------+ | | +-------------| |---------------+ |
+---------------+ +-----------------------------+ +---------------+ +-----------------------------+
Figure 13: Media Transcoder Figure 13: Media Transcoder
Figure 13 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 represented 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 sourced 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 are needed are 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 14 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 14: 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 key
keymanagement. On RTP level the main functions that may be missing management. On RTP level the main functions that might be missing in
in a legacy implementation that otherswise support RTP are RTCP in a legacy implementation that otherwise 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 needed to make it work.
+-A-------------+ +-Translator------------------+ +-A-------------+ +-Translator------------------+
| +-PeerC1------| |-PeerC1------+ | | +-PeerC1------| |-PeerC1------+ |
| | +-UDP1------| |-UDP1------+ | | | | +-UDP1------| |-UDP1------+ | |
| | | +-RTP1----| |-RTP1-----------------------+| | | | +-RTP1----| |-RTP1-----------------------+|
| | | | +-Audio-| |-Audio---+ || | | | | +-Audio-| |-Audio---+ ||
| | | | | AA1|------------>|---------+----------------+ || | | | | | AA1|------------>|---------+----------------+ ||
| | | | | |<------------|BA1 <----+--------------+ | || | | | | | |<------------|BA1 <----+--------------+ | ||
| | | | | |<---RTCP---->|<--------+----------+ | | || | | | | | |<---RTCP---->|<--------+----------+ | | ||
| | | | +-------| |---------+ +---+-+ | | || | | | | +-------| |---------+ +---+-+ | | ||
skipping to change at page 54, line 4 skipping to change at page 55, line 4
| | | | |<---RTCP---->|<--------+----------+ | | || | | | | |<---RTCP---->|<--------+----------+ | | ||
| | | | BA1|------------>|---------+--------------+ | || | | | | BA1|------------>|---------+--------------+ | ||
| | | | |<------------|AA1 <----+----------------+ || | | | | |<------------|AA1 <----+----------------+ ||
| | | +-------| |---------+ || | | | +-------| |---------+ ||
| | +---------| |----------------------------+| | | +---------| |----------------------------+|
| +-----------| |-----------+ | | +-----------| |-----------+ |
| | | | | | | |
+---------------+ +-----------------------------+ +---------------+ +-----------------------------+
Figure 15: 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 can 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 highly dependent on what functions it need to proxy
for the legacy end-point. One possibility is depicted in Figure 15 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 might need the capability
to perform media transcoding to address cases where it otherwise to perform media transcoding to address cases where it otherwise
would need to terminate media transmission. would need to terminate media transmission.
As the translator are generating RTP/RTCP traffic on behalf of B to A As the translator are generating RTP/RTCP traffic on behalf of B to A
it will need to be able to correctly protect these packets that it it will need to be able to correctly protect these packets that it
translates or generates. Thus security context information are translates or generates. Thus security context information are
required in this type of translator if it operates on the RTP/RTCP needed in this type of translator if it operates on the RTP/RTCP
packet content or media. In fact one of the more likley scenario is packet content or media. In fact one of the more likely scenario is
that the translator (gateway) will need to have two different that the translator (gateway) will need to have two different
security contexts one towards A and one towards B and for each RTP/ security contexts one towards A and one towards B and for each RTP/
RTCP packet do a authenticity verification, decryption followed by a RTCP packet do a authenticity verification, decryption followed by a
encryption and integirty protection operation to resolve missmatch in encryption and integrity protection operation to resolve mismatch 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 flavours, 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 16. feature as depicted below in Figure 16.
+---+ +------------+ +---+ +---+ +------------+ +---+
| A |<---->| |<---->| B | | A |<---->| |<---->| B |
+---+ | | +---+ +---+ | | +---+
| Translator | | Translator |
+---+ | | +---+ +---+ | | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +------------+ +---+ +---+ +------------+ +---+
Figure 16: 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 key management 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. participants, and is sometimes called a transport-layer translator.
In Figure 16, 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
skipping to change at page 57, line 5 skipping to change at page 58, line 5
| | +-----------| |-------------+ | | | | +-----------| |-------------+ | |
| +-------------| |---------------+ | | +-------------| |---------------+ |
+---------------+ +--------------------------------+ +---------------+ +--------------------------------+
Figure 17: 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 has to be capable of handling 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 have to 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
RTP 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 inconsistent and without explicit
information about it being blocked must be interpret as severe information about it being blocked has to be interpreted 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
and C carrying pt=97 and having the same packetization and encoding, and C carrying pt=97 and having the same packetization and encoding,
no entity will have manipulated the packet. no entity will have manipulated the packet.
When it comes to security there exist some additional requirements to When it comes to security there exist some additional requirements to
ensure that the property that the relay can't read the media traffic ensure that the property that the relay can't read the media traffic
is enforced. First of all the key to be used must be agreed such so is enforced. First of all the key to be used has to be agreed such
that the relay doesn't get it, e.g. no DTLS-SRTP handshake with the so that the relay doesn't get it, e.g. no DTLS-SRTP handshake with
relay, instead some other method must be used. Secondly, the keying the relay, instead some other method needs to be used. Secondly, the
structure must be capable of handling multiple end-points in the same keying structure has to be capable of handling multiple end-points in
RTP session. the same 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 18. The first one could other participants in the RTP session Figure 18. The first one could
skipping to change at page 58, line 50 skipping to change at page 59, line 50
+---+ +---+ +---+ +---+ +---+ +---+
| A |--->| B |--->| C | | A |--->| B |--->| C |
+---+ +---+ +---+ +---+ +---+ +---+
Figure 19: MediaStream Forwarding Figure 19: MediaStream Forwarding
There exist two main approaches to how B forwards the media from A to There exist two main approaches to how B forwards the media from A to
C. The first one is to simply relay the RTP media stream. The second C. The first one is to simply relay the RTP media stream. The second
one 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 approach will result in that the WebRTC end-points will have
to have the same capabilities as being discussed in Relay to have the same capabilities as being discussed in Relay
(Appendix A.4.3). Thus A will see an RTP session that is extended (Appendix A.4.3). Thus A will see an RTP session that is extended
beyond the PeerConnection and see two different receiving end-points beyond the PeerConnection and see two different receiving end-points
with different path characteristics (B and C). Thus A's congestion with different path characteristics (B and C). Thus A's congestion
control needs to be capable of handling this. The security solution control needs to be capable of handling this. The security solution
can either support mechanism that allows A to inform C about the key can either support mechanism that allows A to inform C about the key
A is using despite B and C having agreed on another set of keys. A is using despite B and C having agreed on another set of keys.
Alternatively B will decrypt and then re-encrypt using a new key. Alternatively B will decrypt and then re-encrypt using a new key.
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 coming 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 RTP media stream suitable for the B to will responsible to produce a RTP media stream suitable for the B to
C PeerConnection. This may require media transcoding for congestion C PeerConnection. This might require media transcoding for
control purpose to produce a suitable bit-rate. Thus loosing media congestion control purpose to produce a suitable bit-rate. Thus
quality in the transcoding and forcing B to spend the resource on the loosing media quality in the transcoding and forcing B to spend the
transcoding. The media transcoding does result in a separation of resource on the transcoding. The media transcoding does result in a
the two different legs removing almost all dependencies. B could separation of the two different legs removing almost all
choice to implement logic to optimize its media transcoding dependencies. B could choice to implement logic to optimize its
operation, by for example requesting media properties that are media transcoding operation, by for example requesting media
suitable for C also, thus trying to avoid it having to transcode the properties that are suitable for C also, thus trying to avoid it
content and only forward the media payloads between the two sides. having to transcode the content and only forward the media payloads
For that optimization to be practical WebRTC end-points must support between the two sides. For that optimization to be practical WebRTC
sufficiently good tools for codec control. end-points have to support 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 multiple
PeerConnection all being feed the same set of WebRTC MediaStreams. PeerConnection all being feed the same set of WebRTC MediaStreams.
Each 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 7. 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 8. 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 needs to 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 RTP media streams interested in receiving simulcast of an end-points RTP media streams
to 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
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