draft-ietf-rtcweb-rtp-usage-26.txt   rfc8834.txt 
RTCWEB Working Group C. Perkins Internet Engineering Task Force (IETF) C. Perkins
Internet-Draft University of Glasgow Request for Comments: 8834 University of Glasgow
Intended status: Standards Track M. Westerlund Category: Standards Track M. Westerlund
Expires: September 18, 2016 Ericsson ISSN: 2070-1721 Ericsson
J. Ott J. Ott
Aalto University Technical University Munich
March 17, 2016 January 2021
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP Media Transport and Use of RTP in WebRTC
draft-ietf-rtcweb-rtp-usage-26
Abstract Abstract
The Web Real-Time Communication (WebRTC) framework provides support The framework for Web Real-Time Communication (WebRTC) provides
for direct interactive rich communication using audio, video, text, support for direct interactive rich communication using audio, video,
collaboration, games, etc. between two peers' web-browsers. This text, collaboration, games, etc. between two peers' web browsers.
memo describes the media transport aspects of the WebRTC framework. This memo describes the media transport aspects of the WebRTC
It specifies how the Real-time Transport Protocol (RTP) is used in framework. It specifies how the Real-time Transport Protocol (RTP)
the WebRTC context, and gives requirements for which RTP features, is used in the WebRTC context and gives requirements for which RTP
profiles, and extensions need to be supported. features, profiles, and extensions need to be supported.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This is an Internet Standards Track document.
provisions of BCP 78 and BCP 79.
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time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
This Internet-Draft will expire on September 18, 2016. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8834.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Rationale
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Terminology
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5 4. WebRTC Use of RTP: Core Protocols
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5 4.1. RTP and RTCP
4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7 4.2. Choice of the RTP Profile
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8 4.3. Choice of RTP Payload Formats
4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 10 4.4. Use of RTP Sessions
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 4.5. RTP and RTCP Multiplexing
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 11 4.6. Reduced Size RTCP
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 11 4.7. Symmetric RTP/RTCP
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 12 4.8. Choice of RTP Synchronization Source (SSRC)
4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 12 4.9. Generation of the RTCP Canonical Name (CNAME)
4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 13 4.10. Handling of Leap Seconds
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 13 5. WebRTC Use of RTP: Extensions
5.1. Conferencing Extensions and Topologies . . . . . . . . . 14 5.1. Conferencing Extensions and Topologies
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 15 5.1.1. Full Intra Request (FIR)
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 15 5.1.2. Picture Loss Indication (PLI)
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 16 5.1.3. Slice Loss Indication (SLI)
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 16 5.1.4. Reference Picture Selection Indication (RPSI)
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 16 5.1.5. Temporal-Spatial Trade-Off Request (TSTR)
5.1.6. Temporary Maximum Media Stream Bit Rate Request 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 16 5.2. Header Extensions
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 17 5.2.1. Rapid Synchronization
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 17 5.2.2. Client-to-Mixer Audio Level
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 17 5.2.3. Mixer-to-Client Audio Level
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 18 5.2.4. Media Stream Identification
5.2.4. Media Stream Identification . . . . . . . . . . . . . 18 5.2.5. Coordination of Video Orientation
5.2.5. Coordination of Video Orientation . . . . . . . . . . 18 6. WebRTC Use of RTP: Improving Transport Robustness
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 19 6.1. Negative Acknowledgements and RTP Retransmission
6.1. Negative Acknowledgements and RTP Retransmission . . . . 19 6.2. Forward Error Correction (FEC)
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 20 7. WebRTC Use of RTP: Rate Control and Media Adaptation
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 20 7.1. Boundary Conditions and Circuit Breakers
7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 21 7.2. Congestion Control Interoperability and Legacy Systems
7.2. Congestion Control Interoperability and Legacy Systems . 22 8. WebRTC Use of RTP: Performance Monitoring
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22 9. WebRTC Use of RTP: Future Extensions
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 23 10. Signaling Considerations
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 23 11. WebRTC API Considerations
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 25 12. RTP Implementation Considerations
12. RTP Implementation Considerations . . . . . . . . . . . . . . 27 12.1. Configuration and Use of RTP Sessions
12.1. Configuration and Use of RTP Sessions . . . . . . . . . 27 12.1.1. Use of Multiple Media Sources within an RTP Session
12.1.1. Use of Multiple Media Sources Within an RTP Session 27 12.1.2. Use of Multiple RTP Sessions
12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 28 12.1.3. Differentiated Treatment of RTP Streams
12.1.3. Differentiated Treatment of RTP Streams . . . . . . 33 12.2. Media Source, RTP Streams, and Participant Identification
12.2. Media Source, RTP Streams, and Participant 12.2.1. Media Source Identification
Identification . . . . . . . . . . . . . . . . . . . . . 35 12.2.2. SSRC Collision Detection
12.2.1. Media Source Identification . . . . . . . . . . . . 35 12.2.3. Media Synchronization Context
12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36 13. Security Considerations
12.2.3. Media Synchronisation Context . . . . . . . . . . . 37 14. IANA Considerations
13. Security Considerations . . . . . . . . . . . . . . . . . . . 37 15. References
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39 15.1. Normative References
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39 15.2. Informative References
16. References . . . . . . . . . . . . . . . . . . . . . . . . . 39 Acknowledgements
16.1. Normative References . . . . . . . . . . . . . . . . . . 39 Authors' Addresses
16.2. Informative References . . . . . . . . . . . . . . . . . 44
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 46
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 signaling, these form the basis for
many teleconferencing systems. many teleconferencing systems.
The Web Real-Time communication (WebRTC) framework provides the The Web Real-Time Communication (WebRTC) framework provides the
protocol building blocks to support direct, interactive, real-time protocol building blocks to support direct, interactive, real-time
communication using audio, video, collaboration, games, etc., between communication using audio, video, collaboration, games, etc. between
two peers' web-browsers. This memo describes how the RTP framework two peers' web browsers. This memo describes how the RTP framework
is to be used in the WebRTC context. It proposes a baseline set of is to be used in the WebRTC context. It proposes a baseline set of
RTP features that are to be implemented by all WebRTC Endpoints, RTP features that are to be implemented by all WebRTC endpoints,
along with suggested extensions for enhanced functionality. along with suggested extensions for enhanced functionality.
This memo specifies a protocol intended for use within the WebRTC This memo specifies a protocol intended for use within the WebRTC
framework, but is not restricted to that context. An overview of the framework but is not restricted to that context. An overview of the
WebRTC framework is given in [I-D.ietf-rtcweb-overview]. WebRTC framework is given in [RFC8825].
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 for preparing this memo and choosing these RTP features.
Section 3 defines terminology. Requirements for core RTP protocols Section 3 defines terminology. Requirements for core RTP protocols
are described in Section 4 and suggested RTP extensions are described are described in Section 4, and suggested RTP extensions are
in Section 5. Section 6 outlines mechanisms that can increase described in Section 5. Section 6 outlines mechanisms that can
robustness to network problems, while Section 7 describes congestion increase robustness to network problems, while Section 7 describes
control and rate adaptation mechanisms. The discussion of mandated congestion control and rate adaptation mechanisms. The discussion of
RTP mechanisms concludes in Section 8 with a review of performance mandated RTP mechanisms concludes in Section 8 with a review of
monitoring and network management tools. Section 9 gives some performance monitoring and network management tools. Section 9 gives
guidelines for future incorporation of other RTP and RTP Control some guidelines for future incorporation of other RTP and RTP Control
Protocol (RTCP) extensions into this framework. Section 10 describes Protocol (RTCP) extensions into this framework. Section 10 describes
requirements placed on the signalling channel. Section 11 discusses requirements placed on the signaling channel. Section 11 discusses
the relationship between features of the RTP framework and the WebRTC the relationship between features of the RTP framework and the WebRTC
application programming interface (API), and Section 12 discusses RTP application programming interface (API), and Section 12 discusses RTP
implementation considerations. The memo concludes with security implementation considerations. The memo concludes with security
considerations (Section 13) and IANA considerations (Section 14). considerations (Section 13) and IANA considerations (Section 14).
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 support many new media encodings, but it 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 to review development of the WebRTC framework provides an opportunity to review
the available RTP features and extensions, and to define a common the available RTP features and extensions and define a common
baseline RTP feature set for all WebRTC Endpoints. This builds on baseline RTP feature set for all WebRTC endpoints. This builds on
the past 20 years of RTP development to mandate the use of extensions the past 20 years of RTP development to mandate the use of extensions
that have shown widespread utility, while still remaining compatible that have shown widespread utility, while still remaining compatible
with the wide installed base of RTP implementations where possible. with the wide installed base of RTP implementations where possible.
RTP and RTCP extensions that are not discussed in this document can RTP and RTCP extensions that are not discussed in this document can
be implemented by WebRTC Endpoints if they are beneficial for new use be implemented by WebRTC endpoints if they are beneficial for new use
cases. However, they are not necessary to address the WebRTC use cases. However, they are not necessary to address the WebRTC use
cases and requirements identified in [RFC7478]. cases and requirements identified in [RFC7478].
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
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
document are to be interpreted as described in [RFC2119]. The RFC "OPTIONAL" in this document are to be interpreted as described in BCP
2119 interpretation of these key words applies only when written in 14 [RFC2119] [RFC8174] when, and only when, they appear in all
ALL CAPS. Lower- or mixed-case uses of these key words are not to be capitals, as shown here. Lower- or mixed-case uses of these key
interpreted as carrying special significance in this memo. words are not to be interpreted as carrying special significance in
this memo.
We define the following additional terms: We define the following additional terms:
WebRTC MediaStream: The MediaStream concept defined by the W3C in WebRTC MediaStream: The MediaStream concept defined by the W3C in
the WebRTC API [W3C.WD-mediacapture-streams-20130903]. A the WebRTC API [W3C.WD-mediacapture-streams]. A MediaStream
MediaStream consists of zero or more MediaStreamTracks. consists of zero or more MediaStreamTracks.
MediaStreamTrack: Part of the MediaStream concept defined by the W3C MediaStreamTrack: Part of the MediaStream concept defined by the W3C
in the WebRTC API [W3C.WD-mediacapture-streams-20130903]. A in the WebRTC API [W3C.WD-mediacapture-streams]. A
MediaStreamTrack is an individual stream of media from any type of MediaStreamTrack is an individual stream of media from any type of
media source like a microphone or a camera, but also conceptual media source such as a microphone or a camera, but conceptual
sources, like a audio mix or a video composition, are possible. sources such as an audio mix or a video composition are also
possible.
Transport-layer Flow: A uni-directional flow of transport packets Transport-layer flow: A unidirectional flow of transport packets
that are identified by having a particular 5-tuple of source IP that are identified by a particular 5-tuple of source IP address,
address, source port, destination IP address, destination port, source port, destination IP address, destination port, and
and transport protocol used. transport protocol.
Bi-directional Transport-layer Flow: A bi-directional transport- Bidirectional transport-layer flow: A bidirectional transport-layer
layer flow is a transport-layer flow that is symmetric. That is, flow is a transport-layer flow that is symmetric. That is, the
the transport-layer flow in the reverse direction has a 5-tuple transport-layer flow in the reverse direction has a 5-tuple where
where the source and destination address and ports are swapped the source and destination address and ports are swapped compared
compared to the forward path transport-layer flow, and the to the forward path transport-layer flow, and the transport
transport protocol is the same. protocol is the same.
This document uses the terminology from This document uses the terminology from [RFC7656] and [RFC8825].
[I-D.ietf-avtext-rtp-grouping-taxonomy] and Other terms are used according to their definitions from the RTP
[I-D.ietf-rtcweb-overview]. Other terms are used according to their specification [RFC3550]. In particular, note the following
definitions from the RTP Specification [RFC3550]. Especially note frequently used terms: RTP stream, RTP session, and endpoint.
the following frequently used terms: RTP Stream, RTP Session, and
Endpoint.
4. WebRTC Use of RTP: Core Protocols 4. WebRTC Use of RTP: Core Protocols
The following sections describe the core features of RTP and RTCP The following sections describe the core features of RTP and RTCP
that need to be implemented, along with the mandated RTP profiles. that need to be implemented, along with the mandated RTP profiles.
Also described are the core extensions providing essential features Also described are the core extensions providing essential features
that all WebRTC Endpoints need to implement to function effectively that all WebRTC endpoints need to implement to function effectively
on today's networks. on today's networks.
4.1. RTP and RTCP 4.1. RTP and RTCP
The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be
implemented as the media transport protocol for WebRTC. RTP itself implemented as the media transport protocol for WebRTC. RTP itself
comprises two parts: the RTP data transfer protocol, and the RTP comprises two parts: the RTP data transfer protocol and the RTP
control protocol (RTCP). RTCP is a fundamental and integral part of Control Protocol (RTCP). RTCP is a fundamental and integral part of
RTP, and MUST be implemented and used in all WebRTC Endpoints. RTP and MUST be implemented and used in all WebRTC endpoints.
The following RTP and RTCP features are sometimes omitted in limited The following RTP and RTCP features are sometimes omitted in limited-
functionality implementations of RTP, but are REQUIRED in all WebRTC functionality implementations of RTP, but they are REQUIRED in all
Endpoints: WebRTC endpoints:
o Support for use of multiple simultaneous SSRC values in a single * Support for use of multiple simultaneous synchronization source
RTP session, including support for RTP endpoints that send many (SSRC) values in a single RTP session, including support for RTP
SSRC values simultaneously, following [RFC3550] and endpoints that send many SSRC values simultaneously, following
[I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for [RFC3550] and [RFC8108]. The RTCP optimizations for multi-SSRC
multi-SSRC sessions defined in sessions defined in [RFC8861] MAY be supported; if supported, the
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] MAY be supported; usage MUST be signaled.
if supported the usage MUST be signalled.
o Random choice of SSRC on joining a session; collision detection * Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (see also Section 4.8). and resolution for SSRC values (see also Section 4.8).
o Support for reception of RTP data packets containing CSRC lists, * Support for reception of RTP data packets containing contributing
as generated by RTP mixers, and RTCP packets relating to CSRCs. source (CSRC) lists, as generated by RTP mixers, and RTCP packets
relating to CSRCs.
o Sending correct synchronisation information in the RTCP Sender * Sending correct synchronization information in the RTCP Sender
Reports, to allow receivers to implement lip-synchronisation; see Reports, to allow receivers to implement lip synchronization; see
Section 5.2.1 regarding support for the rapid RTP synchronisation Section 5.2.1 regarding support for the rapid RTP synchronization
extensions. extensions.
o Support for multiple synchronisation contexts. Participants that * Support for multiple synchronization contexts. Participants that
send multiple simultaneous RTP packet streams SHOULD do so as part send multiple simultaneous RTP packet streams SHOULD do so as part
of a single synchronisation context, using a single RTCP CNAME for of a single synchronization context, using a single RTCP CNAME for
all streams and allowing receivers to play the streams out in a all streams and allowing receivers to play the streams out in a
synchronised manner. For compatibility with potential future synchronized manner. For compatibility with potential future
versions of this specification, or for interoperability with non- versions of this specification, or for interoperability with non-
WebRTC devices through a gateway, receivers MUST support multiple WebRTC devices through a gateway, receivers MUST support multiple
synchronisation contexts, indicated by the use of multiple RTCP synchronization contexts, indicated by the use of multiple RTCP
CNAMEs in an RTP session. This specification mandates the usage CNAMEs in an RTP session. This specification mandates the usage
of a single CNAME when sending RTP Streams in some circumstances, of a single CNAME when sending RTP streams in some circumstances;
see Section 4.9. see Section 4.9.
o Support for sending and receiving RTCP SR, RR, SDES, and BYE * Support for sending and receiving RTCP Sender Report (SR),
packet types. Note that support for other RTCP packet types is Receiver Report (RR), Source Description (SDES), and BYE packet
OPTIONAL, unless mandated by other parts of this specification. types. Note that support for other RTCP packet types is OPTIONAL
Note that additional RTCP Packet types are used by the RTP/SAVPF unless mandated by other parts of this specification. Note that
Profile (Section 4.2) and the other RTCP extensions (Section 5). additional RTCP packet types are used by the RTP/SAVPF profile
WebRTC endpoints that implement the SDP bundle negotiation (Section 4.2) and the other RTCP extensions (Section 5). WebRTC
extension will use the SDP grouping framework 'mid' attribute to endpoints that implement the Session Description Protocol (SDP)
identify media streams. Such endpoints MUST implement the RTCP bundle negotiation extension will use the SDP Grouping Framework
SDES MID item described in "mid" attribute to identify media streams. Such endpoints MUST
[I-D.ietf-mmusic-sdp-bundle-negotiation]. implement the RTCP SDES media identification (MID) item described
in [RFC8843].
o Support for multiple endpoints in a single RTP session, and for * Support for multiple endpoints in a single RTP session, and for
scaling the RTCP transmission interval according to the number of scaling the RTCP transmission interval according to the number of
participants in the session; support for randomised RTCP participants in the session; support for randomized RTCP
transmission intervals to avoid synchronisation of RTCP reports; transmission intervals to avoid synchronization of RTCP reports;
support for RTCP timer reconsideration (Section 6.3.6 of support for RTCP timer reconsideration (Section 6.3.6 of
[RFC3550]) and reverse reconsideration (Section 6.3.4 of [RFC3550]) and reverse reconsideration (Section 6.3.4 of
[RFC3550]). [RFC3550]).
o Support for configuring the RTCP bandwidth as a fraction of the * Support for configuring the RTCP bandwidth as a fraction of the
media bandwidth, and for configuring the fraction of the RTCP media bandwidth, and for configuring the fraction of the RTCP
bandwidth allocated to senders, e.g., using the SDP "b=" line bandwidth allocated to senders -- e.g., using the SDP "b=" line
[RFC4566][RFC3556]. [RFC4566] [RFC3556].
o Support for the reduced minimum RTCP reporting interval described * Support for the reduced minimum RTCP reporting interval described
in Section 6.2 of [RFC3550]. When using the reduced minimum RTCP in Section 6.2 of [RFC3550]. When using the reduced minimum RTCP
reporting interval, the fixed (non-reduced) minimum interval MUST reporting interval, the fixed (nonreduced) minimum interval MUST
be used when calculating the participant timeout interval (see be used when calculating the participant timeout interval (see
Sections 6.2 and 6.3.5 of [RFC3550]). The delay before sending Sections 6.2 and 6.3.5 of [RFC3550]). The delay before sending
the initial compound RTCP packet can be set to zero (see the initial compound RTCP packet can be set to zero (see
Section 6.2 of [RFC3550] as updated by Section 6.2 of [RFC3550] as updated by [RFC8108]).
[I-D.ietf-avtcore-rtp-multi-stream]).
o Support for discontinuous transmission. RTP allows endpoints to * Support for discontinuous transmission. RTP allows endpoints to
pause and resume transmission at any time. When resuming, the RTP pause and resume transmission at any time. When resuming, the RTP
sequence number will increase by one, as usual, while the increase sequence number will increase by one, as usual, while the increase
in the RTP timestamp value will depend on the duration of the in the RTP timestamp value will depend on the duration of the
pause. Discontinuous transmission is most commonly used with some pause. Discontinuous transmission is most commonly used with some
audio payload formats, but is not audio specific, and can be used audio payload formats, but it is not audio specific and can be
with any RTP payload format. used with any RTP payload format.
o Ignore unknown RTCP packet types and RTP header extensions. This * Ignore unknown RTCP packet types and RTP header extensions. This
is to ensure robust handling of future extensions, middlebox is to ensure robust handling of future extensions, middlebox
behaviours, etc., that can result in not signalled RTCP packet behaviors, etc., that can result in receiving RTP header
types or RTP header extensions being received. If a compound RTCP extensions or RTCP packet types that were not signaled. If a
packet is received that contains a mixture of known and unknown compound RTCP packet that contains a mixture of known and unknown
RTCP packet types, the known packets types need to be processed as RTCP packet types is received, the known packet types need to be
usual, with only the unknown packet types being discarded. processed as usual, with only the unknown packet types being
discarded.
It is known that a significant number of legacy RTP implementations, It is known that a significant number of legacy RTP implementations,
especially those targeted at VoIP-only systems, do not support all of especially those targeted at systems with only Voice over IP (VoIP),
the above features, and in some cases do not support RTCP at all. do not support all of the above features and in some cases do not
Implementers are advised to consider the requirements for graceful support RTCP at all. Implementers are advised to consider the
degradation when interoperating with legacy implementations. requirements for graceful degradation when interoperating with legacy
implementations.
Other implementation considerations are discussed in Section 12. Other implementation considerations are discussed in Section 12.
4.2. Choice of 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 RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as secure RTP profile for RTCP-based feedback (RTP/SAVPF) [RFC5124], as
extended by [RFC7007], MUST be implemented. The RTP/SAVPF profile is extended by [RFC7007], MUST be implemented. The RTP/SAVPF profile is
the combination of basic RTP/AVP profile [RFC3551], the RTP profile the combination of the basic RTP/AVP profile [RFC3551], the RTP
for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP profile for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure
profile (RTP/SAVP) [RFC3711]. RTP profile (RTP/SAVP) [RFC3711].
The RTCP-based feedback extensions [RFC4585] are needed for the The RTCP-based feedback extensions [RFC4585] are needed for the
improved RTCP timer model. This allows more flexible transmission of improved RTCP timer model. This allows more flexible transmission of
RTCP packets in response to events, rather than strictly according to RTCP packets in response to events, rather than strictly according to
bandwidth, and is vital for being able to report congestion signals bandwidth, and is vital for being able to report congestion signals
as well as media events. These extensions also allow saving RTCP as well as media events. These extensions also allow saving RTCP
bandwidth, and an endpoint will commonly only use the full RTCP bandwidth, and an endpoint will commonly only use the full RTCP
bandwidth allocation if there are many events that require feedback. bandwidth allocation if there are many events that require feedback.
The timer rules are also needed to make use of the RTP conferencing The timer rules are also needed to make use of the RTP conferencing
extensions discussed in Section 5.1. extensions discussed in Section 5.1.
Note: The enhanced RTCP timer model defined in the RTP/AVPF | Note: The enhanced RTCP timer model defined in the RTP/AVPF
profile is backwards compatible with legacy systems that implement | profile is backwards compatible with legacy systems that
only the RTP/AVP or RTP/SAVP profile, given some constraints on | implement only the RTP/AVP or RTP/SAVP profile, given some
parameter configuration such as the RTCP bandwidth value and "trr- | constraints on parameter configuration such as the RTCP
int" (the most important factor for interworking with RTP/(S)AVP | bandwidth value and "trr-int". The most important factor for
endpoints via a gateway is to set the trr-int parameter to a value | interworking with RTP/(S)AVP endpoints via a gateway is to set
representing 4 seconds, see Section 6.1 in | the "trr-int" parameter to a value representing 4 seconds; see
[I-D.ietf-avtcore-rtp-multi-stream]). | Section 7.1.3 of [RFC8108].
The secure RTP (SRTP) profile extensions [RFC3711] are needed to The secure RTP (SRTP) profile extensions [RFC3711] are needed to
provide media encryption, integrity protection, replay protection and provide media encryption, integrity protection, replay protection,
a limited form of source authentication. WebRTC Endpoints MUST NOT and a limited form of source authentication. WebRTC endpoints MUST
send packets using the basic RTP/AVP profile or the RTP/AVPF profile; NOT send packets using the basic RTP/AVP profile or the RTP/AVPF
they MUST employ the full RTP/SAVPF profile to protect all RTP and profile; they MUST employ the full RTP/SAVPF profile to protect all
RTCP packets that are generated (i.e., implementations MUST use SRTP RTP and RTCP packets that are generated. In other words,
and SRTCP). The RTP/SAVPF profile MUST be configured using the implementations MUST use SRTP and Secure RTCP (SRTCP). The RTP/SAVPF
cipher suites, DTLS-SRTP protection profiles, keying mechanisms, and profile MUST be configured using the cipher suites, DTLS-SRTP
other parameters described in [I-D.ietf-rtcweb-security-arch]. protection profiles, keying mechanisms, and other parameters
described in [RFC8827].
4.3. Choice of RTP Payload Formats 4.3. Choice of RTP Payload Formats
Mandatory to implement audio codecs and RTP payload formats for Mandatory-to-implement audio codecs and RTP payload formats for
WebRTC endpoints are defined in [I-D.ietf-rtcweb-audio]. Mandatory WebRTC endpoints are defined in [RFC7874]. Mandatory-to-implement
to implement video codecs and RTP payload formats for WebRTC video codecs and RTP payload formats for WebRTC endpoints are defined
endpoints are defined in [I-D.ietf-rtcweb-video]. WebRTC endpoints in [RFC7742]. WebRTC endpoints MAY additionally implement any other
MAY additionally implement any other codec for which an RTP payload codec for which an RTP payload format and associated signaling has
format and associated signalling has been defined. been defined.
WebRTC Endpoints cannot assume that the other participants in an RTP WebRTC endpoints cannot assume that the other participants in an RTP
session understand any RTP payload format, no matter how common. The session understand any RTP payload format, no matter how common. The
mapping between RTP payload type numbers and specific configurations mapping between RTP payload type numbers and specific configurations
of particular RTP payload formats MUST be agreed before those payload of particular RTP payload formats MUST be agreed before those payload
types/formats can be used. In an SDP context, this can be done using types/formats can be used. In an SDP context, this can be done using
the "a=rtpmap:" and "a=fmtp:" attributes associated with an "m=" the "a=rtpmap:" and "a=fmtp:" attributes associated with an "m="
line, along with any other SDP attributes needed to configure the RTP line, along with any other SDP attributes needed to configure the RTP
payload format. payload format.
Endpoints can signal support for multiple RTP payload formats, or Endpoints can signal support for multiple RTP payload formats or
multiple configurations of a single RTP payload format, as long as multiple configurations of a single RTP payload format, as long as
each unique RTP payload format configuration uses a different RTP each unique RTP payload format configuration uses a different RTP
payload type number. As outlined in Section 4.8, the RTP payload payload type number. As outlined in Section 4.8, the RTP payload
type number is sometimes used to associate an RTP packet stream with type number is sometimes used to associate an RTP packet stream with
a signalling context. This association is possible provided unique a signaling context. This association is possible provided unique
RTP payload type numbers are used in each context. For example, an RTP payload type numbers are used in each context. For example, an
RTP packet stream can be associated with an SDP "m=" line by RTP packet stream can be associated with an SDP "m=" line by
comparing the RTP payload type numbers used by the RTP packet stream comparing the RTP payload type numbers used by the RTP packet stream
with payload types signalled in the "a=rtpmap:" lines in the media with payload types signaled in the "a=rtpmap:" lines in the media
sections of the SDP. This leads to the following considerations: sections of the SDP. This leads to the following considerations:
If RTP packet streams are being associated with signalling If RTP packet streams are being associated with signaling contexts
contexts based on the RTP payload type, then the assignment of RTP based on the RTP payload type, then the assignment of RTP payload
payload type numbers MUST be unique across signalling contexts. type numbers MUST be unique across signaling contexts.
If the same RTP payload format configuration is used in multiple If the same RTP payload format configuration is used in multiple
contexts, then a different RTP payload type number has to be contexts, then a different RTP payload type number has to be
assigned in each context to ensure uniqueness. assigned in each context to ensure uniqueness.
If the RTP payload type number is not being used to associate RTP If the RTP payload type number is not being used to associate RTP
packet streams with a signalling context, then the same RTP packet streams with a signaling context, then the same RTP payload
payload type number can be used to indicate the exact same RTP type number can be used to indicate the exact same RTP payload
payload format configuration in multiple contexts. format configuration in multiple contexts.
A single RTP payload type number MUST NOT be assigned to different A single RTP payload type number MUST NOT be assigned to different
RTP payload formats, or different configurations of the same RTP RTP payload formats, or different configurations of the same RTP
payload format, within a single RTP session (note that the "m=" lines payload format, within a single RTP session (note that the "m=" lines
in an SDP bundle group [I-D.ietf-mmusic-sdp-bundle-negotiation] form in an SDP BUNDLE group [RFC8843] form a single RTP session).
a single RTP session).
An endpoint that has signalled support for multiple RTP payload An endpoint that has signaled support for multiple RTP payload
formats MUST be able to accept data in any of those payload formats formats MUST be able to accept data in any of those payload formats
at any time, unless it has previously signalled limitations on its at any time, unless it has previously signaled limitations on its
decoding capability. This requirement is constrained if several decoding capability. This requirement is constrained if several
types of media (e.g., audio and video) are sent in the same RTP types of media (e.g., audio and video) are sent in the same RTP
session. In such a case, a source (SSRC) is restricted to switching session. In such a case, a source (SSRC) is restricted to switching
only between the RTP payload formats signalled for the type of media only between the RTP payload formats signaled for the type of media
that is being sent by that source; see Section 4.4. To support rapid that is being sent by that source; see Section 4.4. To support rapid
rate adaptation by changing codec, RTP does not require advance rate adaptation by changing codecs, RTP does not require advance
signalling for changes between RTP payload formats used by a single signaling for changes between RTP payload formats used by a single
SSRC that were signalled during session set-up. SSRC that were signaled during session setup.
If performing changes between two RTP payload types that use If performing changes between two RTP payload types that use
different RTP clock rates, an RTP sender MUST follow the different RTP clock rates, an RTP sender MUST follow the
recommendations in Section 4.1 of [RFC7160]. RTP receivers MUST recommendations in Section 4.1 of [RFC7160]. RTP receivers MUST
follow the recommendations in Section 4.3 of [RFC7160] in order to follow the recommendations in Section 4.3 of [RFC7160] in order to
support sources that switch between clock rates in an RTP session support sources that switch between clock rates in an RTP session.
(these recommendations for receivers are backwards compatible with These recommendations for receivers are backwards compatible with the
the case where senders use only a single clock rate). case where senders use only a single clock rate.
4.4. Use of RTP Sessions 4.4. Use of RTP Sessions
An association amongst a set of endpoints communicating using RTP is An association amongst a set of endpoints communicating using RTP is
known as an RTP session [RFC3550]. An endpoint can be involved in known as an RTP session [RFC3550]. An endpoint can be involved in
several RTP sessions at the same time. In a multimedia session, each several RTP sessions at the same time. In a multimedia session, each
type of media has typically been carried in a separate RTP session type of media has typically been carried in a separate RTP session
(e.g., using one RTP session for the audio, and a separate RTP (e.g., using one RTP session for the audio and a separate RTP session
session using a different transport-layer flow for the video). using a different transport-layer flow for the video). WebRTC
WebRTC Endpoints are REQUIRED to implement support for multimedia endpoints are REQUIRED to implement support for multimedia sessions
sessions in this way, separating each RTP session using different in this way, separating each RTP session using different transport-
transport-layer flows for compatibility with legacy systems (this is layer flows for compatibility with legacy systems (this is sometimes
sometimes called session multiplexing). called session multiplexing).
In modern day networks, however, with the widespread use of network In modern-day networks, however, with the widespread use of network
address/port translators (NAT/NAPT) and firewalls, it is desirable to address/port translators (NAT/NAPT) and firewalls, it is desirable to
reduce the number of transport-layer flows used by RTP applications. reduce the number of transport-layer flows used by RTP applications.
This can be done by sending all the RTP packet streams in a single This can be done by sending all the RTP packet streams in a single
RTP session, which will comprise a single transport-layer flow (this RTP session, which will comprise a single transport-layer flow. This
will prevent the use of some quality-of-service mechanisms, as will prevent the use of some quality-of-service mechanisms, as
discussed in Section 12.1.3). Implementations are therefore also discussed in Section 12.1.3. Implementations are therefore also
REQUIRED to support transport of all RTP packet streams, independent REQUIRED to support transport of all RTP packet streams, independent
of media type, in a single RTP session using a single transport layer of media type, in a single RTP session using a single transport-layer
flow, according to [I-D.ietf-avtcore-multi-media-rtp-session] (this flow, according to [RFC8860] (this is sometimes called SSRC
is sometimes called SSRC multiplexing). If multiple types of media multiplexing). If multiple types of media are to be used in a single
are to be used in a single RTP session, all participants in that RTP RTP session, all participants in that RTP session MUST agree to this
session MUST agree to this usage. In an SDP context, usage. In an SDP context, the mechanisms described in [RFC8843] can
[I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to signal such a be used to signal such a bundle of RTP packet streams forming a
bundle of RTP packet streams forming a single RTP session. single RTP session.
Further discussion about the suitability of different RTP session Further discussion about the suitability of different RTP session
structures and multiplexing methods to different scenarios can be structures and multiplexing methods to different scenarios can be
found in [I-D.ietf-avtcore-multiplex-guidelines]. found in [RFC8872].
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
flows (e.g., two UDP ports for each RTP session, one port for RTP and flows (e.g., two UDP ports for each RTP session, one for RTP and one
one port for RTCP). With the increased use of Network Address/Port for RTCP). With the increased use of Network Address/Port
Translation (NAT/NAPT) this has become problematic, since maintaining Translation (NAT/NAPT), this has become problematic, since
multiple NAT bindings can be costly. It also complicates firewall maintaining multiple NAT bindings can be costly. It also complicates
administration, since multiple ports need to be opened to allow RTP firewall administration, since multiple ports need to be opened to
traffic. To reduce these costs and session set-up times, allow RTP traffic. To reduce these costs and session setup times,
implementations are REQUIRED to support multiplexing RTP data packets implementations are REQUIRED to support multiplexing RTP data packets
and RTCP control packets on a single transport-layer flow [RFC5761]. and RTCP control packets on a single transport-layer flow [RFC5761].
Such RTP and RTCP multiplexing MUST be negotiated in the signalling Such RTP and RTCP multiplexing MUST be negotiated in the signaling
channel before it is used. If SDP is used for signalling, this channel before it is used. If SDP is used for signaling, this
negotiation MUST use the mechanism defined in [RFC5761]. negotiation MUST use the mechanism defined in [RFC5761].
Implementations can also support sending RTP and RTCP on separate Implementations can also support sending RTP and RTCP on separate
transport-layer flows, but this is OPTIONAL to implement. If an transport-layer flows, but this is OPTIONAL to implement. If an
implementation does not support RTP and RTCP sent on separate implementation does not support RTP and RTCP sent on separate
transport-layer flows, it MUST indicate that using the mechanism transport-layer flows, it MUST indicate that using the mechanism
defined in [I-D.ietf-mmusic-mux-exclusive]. defined in [RFC8858].
Note that the use of RTP and RTCP multiplexed onto a single Note that the use of RTP and RTCP multiplexed onto a single
transport-layer flow ensures that there is occasional traffic sent on transport-layer flow ensures that there is occasional traffic sent on
that port, even if there is no active media traffic. This can be that port, even if there is no active media traffic. This can be
useful to keep NAT bindings alive [RFC6263]. useful to keep NAT bindings alive [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 SR or RR packet.
or Receiver Report (RR) packet. When using frequent RTCP feedback When using frequent RTCP feedback messages under the RTP/AVPF profile
messages under the RTP/AVPF Profile [RFC4585] these statistics are [RFC4585], these statistics are not needed in every packet, and they
not needed in every packet, and unnecessarily increase the mean RTCP unnecessarily increase the mean RTCP packet size. This can limit the
packet size. This can limit the frequency at which RTCP packets can frequency at which RTCP packets can be sent within the RTCP bandwidth
be sent within the RTCP bandwidth share. share.
To avoid this problem, [RFC5506] specifies how to reduce the mean To avoid this problem, [RFC5506] specifies how to reduce the mean
RTCP message 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. Implementations adapt their transmission and encoding behavior. Implementations MUST
MUST support sending and receiving non-compound RTCP feedback packets support sending and receiving noncompound RTCP feedback packets
[RFC5506]. Use of non-compound RTCP packets MUST be negotiated using [RFC5506]. Use of noncompound RTCP packets MUST be negotiated using
the signalling channel. If SDP is used for signalling, this the signaling channel. If SDP is used for signaling, this
negotiation MUST use the attributes defined in [RFC5506]. For negotiation MUST use the attributes defined in [RFC5506]. For
backwards compatibility, implementations are also REQUIRED to support backwards compatibility, implementations are also REQUIRED to support
the use of compound RTCP feedback packets if the remote endpoint does 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. not agree to the use of noncompound RTCP in the signaling 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]. The reason REQUIRED to implement and use symmetric RTP [RFC4961]. The reason
for using symmetric RTP is primarily to avoid issues with NATs and for using symmetric RTP is primarily to avoid issues with NATs and
Firewalls by ensuring that the send and receive RTP packet streams, firewalls by ensuring that the send and receive RTP packet streams,
as well as RTCP, are actually bi-directional transport-layer flows. as well as RTCP, are actually bidirectional transport-layer flows.
This will keep alive the NAT and firewall pinholes, and help indicate This will keep alive the NAT and firewall pinholes and help indicate
consent that the receive direction is a transport-layer flow the consent that the receive direction is a transport-layer flow the
intended recipient actually wants. In addition, it saves resources, intended recipient actually wants. In addition, it saves resources,
specifically ports at the endpoints, but also in the network as NAT specifically ports at the endpoints, but also in the network, because
mappings or firewall state is not unnecessary bloated. The amount of the NAT mappings or firewall state is not unnecessarily bloated. The
per flow QoS state kept in the network is also reduced. amount of per-flow QoS state kept in the network is also reduced.
4.8. Choice of RTP Synchronisation Source (SSRC) 4.8. Choice of RTP Synchronization Source (SSRC)
Implementations are REQUIRED to support signalled RTP synchronisation Implementations are REQUIRED to support signaled RTP synchronization
source (SSRC) identifiers. If SDP is used, this MUST be done using source (SSRC) identifiers. If SDP is used, this MUST be done using
the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of the "a=ssrc:" SDP attribute defined in Sections 4.1 and 5 of
[RFC5576] and the "previous-ssrc" source attribute defined in [RFC5576] and the "previous-ssrc" source attribute defined in
Section 6.2 of [RFC5576]; other per-SSRC attributes defined in Section 6.2 of [RFC5576]; other per-SSRC attributes defined in
[RFC5576] MAY be supported. [RFC5576] MAY be supported.
While support for signalled SSRC identifiers is mandated, their use While support for signaled SSRC identifiers is mandated, their use in
in an RTP session is OPTIONAL. Implementations MUST be prepared to an RTP session is OPTIONAL. Implementations MUST be prepared to
accept RTP and RTCP packets using SSRCs that have not been explicitly accept RTP and RTCP packets using SSRCs that have not been explicitly
signalled ahead of time. Implementations MUST support random SSRC signaled ahead of time. Implementations MUST support random SSRC
assignment, and MUST support SSRC collision detection and resolution, assignment and MUST support SSRC collision detection and resolution,
according to [RFC3550]. When using signalled SSRC values, collision according to [RFC3550]. When using signaled SSRC values, collision
detection MUST be performed as described in Section 5 of [RFC5576]. detection MUST be performed as described in Section 5 of [RFC5576].
It is often desirable to associate an RTP packet stream with a non- It is often desirable to associate an RTP packet stream with a non-
RTP context. For users of the WebRTC API a mapping between SSRCs and RTP context. For users of the WebRTC API, a mapping between SSRCs
MediaStreamTracks are provided per Section 11. For gateways or other and MediaStreamTracks is provided per Section 11. For gateways or
usages it is possible to associate an RTP packet stream with an "m=" other usages, it is possible to associate an RTP packet stream with
line in a session description formatted using SDP. If SSRCs are an "m=" line in a session description formatted using SDP. If SSRCs
signalled this is straightforward (in SDP the "a=ssrc:" line will be are signaled, this is straightforward (in SDP, the "a=ssrc:" line
at the media level, allowing a direct association with an "m=" line). will be at the media level, allowing a direct association with an
If SSRCs are not signalled, the RTP payload type numbers used in an "m=" line). If SSRCs are not signaled, the RTP payload type numbers
RTP packet stream are often sufficient to associate that packet used in an RTP packet stream are often sufficient to associate that
stream with a signalling context (e.g., if RTP payload type numbers packet stream with a signaling context. For example, if RTP payload
are assigned as described in Section 4.3 of this memo, the RTP type numbers are assigned as described in Section 4.3 of this memo,
payload types used by an RTP packet stream can be compared with the RTP payload types used by an RTP packet stream can be compared
values in SDP "a=rtpmap:" lines, which are at the media level in SDP, with values in SDP "a=rtpmap:" lines, which are at the media level in
and so map to an "m=" line). SDP and so map to an "m=" line.
4.9. Generation of the RTCP Canonical Name (CNAME) 4.9. Generation of the RTCP Canonical Name (CNAME)
The RTCP Canonical Name (CNAME) provides a persistent transport-level The RTCP Canonical Name (CNAME) provides a persistent transport-level
identifier for an RTP endpoint. While the Synchronisation Source identifier for an RTP endpoint. While the SSRC identifier for an RTP
(SSRC) identifier for an RTP endpoint can change if a collision is endpoint can change if a collision is detected or when the RTP
detected, or when the RTP application is restarted, its RTCP CNAME is application is restarted, its RTCP CNAME is meant to stay unchanged
meant to stay unchanged for the duration of a RTCPeerConnection for the duration of an RTCPeerConnection [W3C.WebRTC], so that RTP
[W3C.WD-webrtc-20130910], so that RTP endpoints can be uniquely endpoints can be uniquely identified and associated with their RTP
identified and associated with their RTP packet streams within a set packet streams within a set of related RTP sessions.
of related RTP sessions.
Each RTP endpoint MUST have at least one RTCP CNAME, and that RTCP Each RTP endpoint MUST have at least one RTCP CNAME, and that RTCP
CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs
identify a particular synchronisation context, i.e., all SSRCs identify a particular synchronization context -- i.e., all SSRCs
associated with a single RTCP CNAME share a common reference clock. associated with a single RTCP CNAME share a common reference clock.
If an endpoint has SSRCs that are associated with several If an endpoint has SSRCs that are associated with several
unsynchronised reference clocks, and hence different synchronisation unsynchronized reference clocks, and hence different synchronization
contexts, it will need to use multiple RTCP CNAMEs, one for each contexts, it will need to use multiple RTCP CNAMEs, one for each
synchronisation context. synchronization context.
Taking the discussion in Section 11 into account, a WebRTC Endpoint Taking the discussion in Section 11 into account, a WebRTC endpoint
MUST NOT use more than one RTCP CNAME in the RTP sessions belonging MUST NOT use more than one RTCP CNAME in the RTP sessions belonging
to single RTCPeerConnection (that is, an RTCPeerConnection forms a to a single RTCPeerConnection (that is, an RTCPeerConnection forms a
synchronisation context). RTP middleboxes MAY generate RTP packet synchronization context). RTP middleboxes MAY generate RTP packet
streams associated with more than one RTCP CNAME, to allow them to streams associated with more than one RTCP CNAME, to allow them to
avoid having to resynchronize media from multiple different endpoints avoid having to resynchronize media from multiple different endpoints
part of a multi-party RTP session. that are part of a multiparty RTP session.
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 (Section 13). Accordingly, a problematic from a privacy viewpoint (Section 13). Accordingly, a
WebRTC Endpoint MUST generate a new, unique, short-term persistent WebRTC endpoint MUST generate a new, unique, short-term persistent
RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a
single exception; if explicitly requested at creation an single exception; if explicitly requested at creation, an
RTCPeerConnection MAY use the same CNAME as as an existing RTCPeerConnection MAY use the same CNAME as an existing
RTCPeerConnection within their common same-origin context. RTCPeerConnection within their common same-origin context.
An WebRTC Endpoint MUST support reception of any CNAME that matches A WebRTC endpoint 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 chosen according to the form and cannot assume that any CNAME will be chosen according to the form
suggested above. suggested above.
4.10. Handling of Leap Seconds 4.10. Handling of Leap Seconds
The guidelines regarding handling of leap seconds to limit their The guidelines given in [RFC7164] regarding handling of leap seconds
impact on RTP media play-out and synchronization given in [RFC7164] to limit their impact on RTP media play-out and synchronization
SHOULD be followed. SHOULD be followed.
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 context. One set of these extensions is performance, in the WebRTC context. One set of these extensions is
related to conferencing, while others are more generic in nature. related to conferencing, while others are more generic in nature.
The following subsections describe the various RTP extensions The following subsections describe the various RTP extensions
mandated or suggested for use within WebRTC. mandated or suggested for use within WebRTC.
5.1. Conferencing Extensions and Topologies 5.1. Conferencing Extensions and Topologies
RTP is a protocol that inherently supports group communication. RTP is a protocol that inherently supports group communication.
Groups can be implemented by having each endpoint send its RTP packet Groups can be implemented by having each endpoint send its RTP packet
streams to an RTP middlebox that redistributes the traffic, by using streams to an RTP middlebox that redistributes the traffic, by using
a mesh of unicast RTP packet streams between endpoints, or by using a mesh of unicast RTP packet streams between endpoints, or by using
an IP multicast group to distribute the RTP packet streams. These an IP multicast group to distribute the RTP packet streams. These
topologies can be implemented in a number of ways as discussed in topologies can be implemented in a number of ways as discussed in
[I-D.ietf-avtcore-rtp-topologies-update]. [RFC7667].
While the use of IP multicast groups is popular in IPTV systems, the While the use of IP multicast groups is popular in IPTV systems, the
topologies based on RTP middleboxes are dominant in interactive video topologies based on RTP middleboxes are dominant in interactive
conferencing environments. Topologies based on a mesh of unicast video-conferencing environments. Topologies based on a mesh of
transport-layer flows to create a common RTP session have not seen unicast transport-layer flows to create a common RTP session have not
widespread deployment to date. Accordingly, WebRTC Endpoints are not seen widespread deployment to date. Accordingly, WebRTC endpoints
expected to support topologies based on IP multicast groups or to are not expected to support topologies based on IP multicast groups
support mesh-based topologies, such as a point-to-multipoint mesh or mesh-based topologies, such as a point-to-multipoint mesh
configured as a single RTP session (Topo-Mesh in the terminology of configured as a single RTP session ("Topo-Mesh" in the terminology of
[I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to- [RFC7667]). However, a point-to-multipoint mesh constructed using
multipoint mesh constructed using several RTP sessions, implemented several RTP sessions, implemented in WebRTC using independent
in WebRTC using independent RTCPeerConnections RTCPeerConnections [W3C.WebRTC], can be expected to be used in WebRTC
[W3C.WD-webrtc-20130910], can be expected to be used in WebRTC, and and needs to be supported.
needs to be supported.
WebRTC Endpoints implemented according to this memo are expected to WebRTC endpoints implemented according to this memo are expected to
support all the topologies described in support all the topologies described in [RFC7667] where the RTP
[I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send endpoints send and receive unicast RTP packet streams to and from
and receive unicast RTP packet streams to and from some peer device, some peer device, provided that peer can participate in performing
provided that peer can participate in performing congestion control congestion control on the RTP packet streams. The peer device could
on the RTP packet streams. The peer device could be another RTP be another RTP endpoint, or it could be an RTP middlebox that
endpoint, or it could be an RTP middlebox that redistributes the RTP redistributes the RTP packet streams to other RTP endpoints. This
packet streams to other RTP endpoints. This limitation means that limitation means that some of the RTP middlebox-based topologies are
some of the RTP middlebox-based topologies are not suitable for use not suitable for use in WebRTC. Specifically:
in WebRTC. Specifically:
o Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used, * Video-switching Multipoint Control Units (MCUs) (Topo-Video-
since they make the use of RTCP for congestion control and quality switch-MCU) SHOULD NOT be used, since they make the use of RTCP
of service reports problematic (see Section 3.8 of for congestion control and quality-of-service reports problematic
[I-D.ietf-avtcore-rtp-topologies-update]). (see Section 3.8 of [RFC7667]).
o The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology * The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
SHOULD NOT be used because its safe use requires a congestion SHOULD NOT be used, because its safe use requires a congestion
control algorithm or RTP circuit breaker that handles point to control algorithm or RTP circuit breaker that handles point to
multipoint, which has not yet been standardised. multipoint, which has not yet been standardized.
The following topology can be used, however it has some issues worth The following topology can be used, however it has some issues worth
noting: noting:
o Content modifying MCUs with RTCP termination (Topo-RTCP- * Content-modifying MCUs with RTCP termination (Topo-RTCP-
terminating-MCU) MAY be used. Note that in this RTP Topology, RTP terminating-MCU) MAY be used. Note that in this RTP topology, RTP
loop detection and identification of active senders is the loop detection and identification of active senders is the
responsibility of the WebRTC application; since the clients are responsibility of the WebRTC application; since the clients are
isolated from each other at the RTP layer, RTP cannot assist with isolated from each other at the RTP layer, RTP cannot assist with
these functions (see section 3.9 of these functions (see Section 3.9 of [RFC7667]).
[I-D.ietf-avtcore-rtp-topologies-update]).
The RTP extensions described in Section 5.1.1 to Section 5.1.6 are The RTP extensions described in Sections 5.1.1 to 5.1.6 are designed
designed to be used with centralised conferencing, where an RTP to be used with centralized conferencing, where an RTP middlebox
middlebox (e.g., a conference bridge) receives a participant's RTP (e.g., a conference bridge) receives a participant's RTP packet
packet streams and distributes them to the other participants. These streams and distributes them to the other participants. These
extensions are not necessary for interoperability; an RTP endpoint extensions are not necessary for interoperability; an RTP endpoint
that does not implement these extensions will work correctly, but that does not implement these extensions will work correctly but
might offer poor performance. Support for the listed extensions will might offer poor performance. Support for the listed extensions will
greatly improve the quality of experience and, to provide a greatly improve the quality of experience; to provide a reasonable
reasonable baseline quality, some of these extensions are mandatory baseline quality, some of these extensions are mandatory to be
to be supported by WebRTC Endpoints. supported by WebRTC endpoints.
The RTCP conferencing extensions are defined in Extended RTP Profile The RTCP conferencing extensions are defined in "Extended RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/ for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
AVPF) [RFC4585] and the memo on Codec Control Messages (CCM) in RTP/ AVPF)" [RFC4585] and "Codec Control Messages in the RTP Audio-Visual
AVPF [RFC5104]; they are fully usable by the Secure variant of this Profile with Feedback (AVPF)" [RFC5104]; they are fully usable by the
profile (RTP/SAVPF) [RFC5124]. secure variant of this profile (RTP/SAVPF) [RFC5124].
5.1.1. Full Intra Request (FIR) 5.1.1. Full Intra Request (FIR)
The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1 The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1
of the Codec Control Messages [RFC5104]. It is used to make the of Codec Control Messages [RFC5104]. It is used to make the mixer
mixer request a new Intra picture from a participant in the session. request a new Intra picture from a participant in the session. This
This is used when switching between sources to ensure that the is used when switching between sources to ensure that the receivers
receivers can decode the video or other predictive media encoding can decode the video or other predictive media encoding with long
with long prediction chains. WebRTC Endpoints that are sending media prediction chains. WebRTC endpoints that are sending media MUST
MUST understand and react to FIR feedback messages they receive, understand and react to FIR feedback messages they receive, since
since this greatly improves the user experience when using this greatly improves the user experience when using centralized
centralised mixer-based conferencing. Support for sending FIR mixer-based conferencing. Support for sending FIR messages is
messages is OPTIONAL. OPTIONAL.
5.1.2. Picture Loss Indication (PLI) 5.1.2. Picture Loss Indication (PLI)
The Picture Loss Indication message is defined in Section 6.3.1 of The Picture Loss Indication message is defined in Section 6.3.1 of
the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the
sending encoder that it lost the decoder context and would like to sending encoder that it lost the decoder context and would like to
have it repaired somehow. This is semantically different from the have it repaired somehow. This is semantically different from the
Full Intra Request above as there could be multiple ways to fulfil Full Intra Request above, as there could be multiple ways to fulfill
the request. WebRTC Endpoints that are sending media MUST understand the request. WebRTC endpoints that are sending media MUST understand
and react to PLI feedback messages as a loss tolerance mechanism. and react to PLI feedback messages as a loss-tolerance mechanism.
Receivers MAY send PLI messages. Receivers MAY send PLI messages.
5.1.3. Slice Loss Indication (SLI) 5.1.3. Slice Loss Indication (SLI)
The Slice Loss Indication message is defined in Section 6.3.2 of the The Slice Loss Indication message is defined in Section 6.3.2 of the
RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the
encoder that it has detected the loss or corruption of one or more encoder that it has detected the loss or corruption of one or more
consecutive macro blocks, and would like to have these repaired consecutive macro blocks and would like to have these repaired
somehow. It is RECOMMENDED that receivers generate SLI feedback somehow. It is RECOMMENDED that receivers generate SLI feedback
messages if slices are lost when using a codec that supports the messages if slices are lost when using a codec that supports the
concept of macro blocks. A sender that receives an SLI feedback concept of macro blocks. A sender that receives an SLI feedback
message SHOULD attempt to repair the lost slice(s). message SHOULD attempt to repair the lost slice(s).
5.1.4. Reference Picture Selection Indication (RPSI) 5.1.4. Reference Picture Selection Indication (RPSI)
Reference Picture Selection Indication (RPSI) messages are defined in Reference Picture Selection Indication (RPSI) messages are defined in
Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video encoding Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video-encoding
standards allow the use of older reference pictures than the most standards allow the use of older reference pictures than the most
recent one for predictive coding. If such a codec is in use, and if recent one for predictive coding. If such a codec is in use, and if
the encoder has learnt that encoder-decoder synchronisation has been the encoder has learned that encoder-decoder synchronization has been
lost, then a known as correct reference picture can be used as a base lost, then a known-as-correct reference picture can be used as a base
for future coding. The RPSI message allows this to be signalled. for future coding. The RPSI message allows this to be signaled.
Receivers that detect that encoder-decoder synchronisation has been Receivers that detect that encoder-decoder synchronization has been
lost SHOULD generate an RPSI feedback message if codec being used lost SHOULD generate an RPSI feedback message if the codec being used
supports reference picture selection. A RTP packet stream sender supports reference-picture selection. An RTP packet-stream sender
that receives such an RPSI message SHOULD act on that messages to that receives such an RPSI message SHOULD act on that messages to
change the reference picture, if it is possible to do so within the change the reference picture, if it is possible to do so within the
available bandwidth constraints, and with the codec being used. available bandwidth constraints and with the codec being used.
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
image quality but low frame rate. Support for TSTR requests and spatial image quality but low frame rate. Support for TSTR requests
notifications is OPTIONAL. and notifications is OPTIONAL.
5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR) 5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of The Temporary Maximum Media Stream Bit Rate Request (TMMBR) feedback
the Codec Control Messages [RFC5104]. This request and its message is defined in Sections 3.5.4 and 4.2.1 of Codec Control
notification message are used by a media receiver to inform the Messages [RFC5104]. This request and its corresponding Temporary
sending party that there is a current limitation on the amount of Maximum Media Stream Bit Rate Notification (TMMBN) message [RFC5104]
bandwidth available to this receiver. There can be various reasons are used by a media receiver to inform the sending party that there
for this: for example, an RTP mixer can use this message to limit the is a current limitation on the amount of bandwidth available to this
media rate of the sender being forwarded by the mixer (without doing receiver. There can be various reasons for this: for example, an RTP
media transcoding) to fit the bottlenecks existing towards the other mixer can use this message to limit the media rate of the sender
session participants. WebRTC Endpoints that are sending media are being forwarded by the mixer (without doing media transcoding) to fit
REQUIRED to implement support for TMMBR messages, and MUST follow the bottlenecks existing towards the other session participants.
bandwidth limitations set by a TMMBR message received for their SSRC. WebRTC endpoints that are sending media are REQUIRED to implement
The sending of TMMBR requests is OPTIONAL. support for TMMBR messages and MUST follow bandwidth limitations set
by a TMMBR message received for their SSRC. The sending of TMMBR
messages 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 WebRTC, but if they are used, they MUST be extensions is OPTIONAL in WebRTC, but if they are used, they MUST be
formatted and signalled following the general mechanism for RTP formatted and signaled following the general mechanism for RTP header
header extensions defined in [RFC5285], since this gives well-defined extensions defined in [RFC8285], since this gives well-defined
semantics to RTP header extensions. semantics to RTP header extensions.
As noted in [RFC5285], the requirement from the RTP specification As noted in [RFC8285], the requirement from the RTP specification
that header extensions are "designed so that the header extension may that header extensions are "designed so that the header extension may
be ignored" [RFC3550] stands. To be specific, header extensions MUST be ignored" [RFC3550] stands. To be specific, header extensions MUST
only be used for data that can safely be ignored by the recipient only be used for data that can safely be ignored by the recipient
without affecting interoperability, and MUST NOT be used when the without affecting interoperability and MUST NOT be used when the
presence of the extension has changed the form or nature of the rest presence of the extension has changed the form or nature of the rest
of the packet in a way that is not compatible with the way the stream of the packet in a way that is not compatible with the way the stream
is signalled (e.g., as defined by the payload type). Valid examples is signaled (e.g., as defined by the payload type). Valid examples
of RTP header extensions might include metadata that is additional to of RTP header extensions might include metadata that is additional to
the usual RTP information, but that can safely be ignored without the usual RTP information but that can safely be ignored without
compromising interoperability. compromising interoperability.
5.2.1. Rapid Synchronisation 5.2.1. Rapid Synchronization
Many RTP sessions require synchronisation between audio, video, and Many RTP sessions require synchronization between audio, video, and
other content. This synchronisation is performed by receivers, using other content. This synchronization is performed by receivers, using
information contained in RTCP SR packets, as described in the RTP information contained in RTCP SR packets, as described in the RTP
specification [RFC3550]. This basic mechanism can be slow, however, specification [RFC3550]. This basic mechanism can be slow, however,
so it is RECOMMENDED that the rapid RTP synchronisation extensions so it is RECOMMENDED that the rapid RTP synchronization extensions
described in [RFC6051] be implemented in addition to RTCP SR-based described in [RFC6051] be implemented in addition to RTCP SR-based
synchronisation. synchronization.
This header extension uses the [RFC5285] generic header extension This header extension uses the generic header extension framework
framework, and so needs to be negotiated before it can be used. described in [RFC8285] and so needs to be negotiated before it can be
used.
5.2.2. Client-to-Mixer Audio Level 5.2.2. Client-to-Mixer Audio Level
The Client to Mixer Audio Level extension [RFC6464] is an RTP header The client-to-mixer audio level extension [RFC6464] is an RTP header
extension used by an endpoint to inform a mixer about the level of extension used by an endpoint to inform a mixer about the level of
audio activity in the packet to which the header is attached. This audio activity in the packet to which the header is attached. This
enables an RTP middlebox to make mixing or selection decisions enables an RTP middlebox to make mixing or selection decisions
without decoding or detailed inspection of the payload, reducing the without decoding or detailed inspection of the payload, reducing the
complexity in some types of mixers. It can also save decoding complexity in some types of mixers. It can also save decoding
resources in receivers, which can choose to decode only the most resources in receivers, which can choose to decode only the most
relevant RTP packet streams based on audio activity levels. relevant RTP packet streams based on audio activity levels.
The Client-to-Mixer Audio Level [RFC6464] header extension MUST be The client-to-mixer audio level header extension [RFC6464] MUST be
implemented. It is REQUIRED that implementations are capable of implemented. It is REQUIRED that implementations be capable of
encrypting the header extension according to [RFC6904] since the encrypting the header extension according to [RFC6904], since the
information contained in these header extensions can be considered information contained in these header extensions can be considered
sensitive. The use of this encryption is RECOMMENDED, however usage sensitive. The use of this encryption is RECOMMENDED; however, usage
of the encryption can be explicitly disabled through API or of the encryption can be explicitly disabled through API or
signalling. signaling.
This header extension uses the [RFC5285] generic header extension This header extension uses the generic header extension framework
framework, and so needs to be negotiated before it can be used. described in [RFC8285] and so needs to be negotiated before it can be
used.
5.2.3. Mixer-to-Client Audio Level 5.2.3. Mixer-to-Client Audio Level
The Mixer to Client Audio Level header extension [RFC6465] provides The mixer-to-client audio level header extension [RFC6465] provides
an endpoint with the audio level of the different sources mixed into an endpoint with the audio level of the different sources mixed into
a common source stream by a RTP mixer. This enables a user interface a common source stream by an RTP mixer. This enables a user
to indicate the relative activity level of each session participant, interface to indicate the relative activity level of each session
rather than just being included or not based on the CSRC field. This participant, rather than just being included or not based on the CSRC
is a pure optimisation of non critical functions, and is hence field. This is a pure optimization of non-critical functions and is
OPTIONAL to implement. If this header extension is implemented, it hence OPTIONAL to implement. If this header extension is
is REQUIRED that implementations are capable of encrypting the header implemented, it is REQUIRED that implementations be capable of
extension according to [RFC6904] since the information contained in encrypting the header extension according to [RFC6904], since the
these header extensions can be considered sensitive. It is further information contained in these header extensions can be considered
RECOMMENDED that this encryption is used, unless the encryption has sensitive. It is further RECOMMENDED that this encryption be used,
been explicitly disabled through API or signalling. unless the encryption has been explicitly disabled through API or
signaling.
This header extension uses the [RFC5285] generic header extension This header extension uses the generic header extension framework
framework, and so needs to be negotiated before it can be used. described in [RFC8285] and so needs to be negotiated before it can be
used.
5.2.4. Media Stream Identification 5.2.4. Media Stream Identification
WebRTC endpoints that implement the SDP bundle negotiation extension WebRTC endpoints that implement the SDP bundle negotiation extension
will use the SDP grouping framework 'mid' attribute to identify media will use the SDP Grouping Framework "mid" attribute to identify media
streams. Such endpoints MUST implement the RTP MID header extension streams. Such endpoints MUST implement the RTP MID header extension
described in [I-D.ietf-mmusic-sdp-bundle-negotiation]. described in [RFC8843].
This header extension uses the [RFC5285] generic header extension This header extension uses the generic header extension framework
framework, and so needs to be negotiated before it can be used. described in [RFC8285] and so needs to be negotiated before it can be
used.
5.2.5. Coordination of Video Orientation 5.2.5. Coordination of Video Orientation
WebRTC endpoints that send or receive video MUST implement the WebRTC endpoints that send or receive video MUST implement the
coordination of video orientation (CVO) RTP header extension as coordination of video orientation (CVO) RTP header extension as
described in Section 4 of [I-D.ietf-rtcweb-video]. described in Section 4 of [RFC7742].
This header extension uses the [RFC5285] generic header extension This header extension uses the generic header extension framework
framework, and so needs to be negotiated before it can be used. described in [RFC8285] and so needs to be negotiated before it can be
used.
6. WebRTC Use of RTP: Improving Transport Robustness 6. WebRTC Use of RTP: Improving Transport Robustness
There are tools that can make RTP packet streams robust against There are tools that can make RTP packet streams robust against
packet loss and reduce the impact of loss on media quality. However, packet loss and reduce the impact of loss on media quality. However,
they generally add some overhead compared to a non-robust stream. they generally add some overhead compared to a non-robust stream.
The overhead needs to be considered, and the aggregate bit-rate MUST The overhead needs to be considered, and the aggregate bitrate MUST
be rate controlled to avoid causing network congestion (see be rate controlled to avoid causing network congestion (see
Section 7). As a result, improving robustness might require a lower Section 7). As a result, improving robustness might require a lower
base encoding quality, but has the potential to deliver that quality base encoding quality but has the potential to deliver that quality
with fewer errors. The mechanisms described in the following sub- with fewer errors. The mechanisms described in the following
sections can be used to improve tolerance to packet loss. subsections can be used to improve tolerance to packet loss.
6.1. Negative Acknowledgements and RTP Retransmission 6.1. Negative Acknowledgements and RTP Retransmission
As a consequence of supporting the RTP/SAVPF profile, implementations As a consequence of supporting the RTP/SAVPF profile, implementations
can send negative acknowledgements (NACKs) for RTP data packets can send 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 optimize
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. known lost packets.
RTP packet stream senders are REQUIRED to understand the Generic NACK RTP packet stream senders are REQUIRED to understand the generic NACK
message defined in Section 6.2.1 of [RFC4585], but MAY choose to message defined in Section 6.2.1 of [RFC4585], but they MAY choose to
ignore some or all of this feedback (following Section 4.2 of ignore some or all of this feedback (following Section 4.2 of
[RFC4585]). Receivers MAY send NACKs for missing RTP packets. [RFC4585]). Receivers MAY send NACKs for missing RTP packets.
Guidelines on when to send NACKs are provided in [RFC4585]. It is Guidelines on when to send NACKs are provided in [RFC4585]. It is
not expected that a receiver will send a NACK for every lost RTP not expected that a receiver will send a NACK for every lost RTP
packet, rather it needs to consider the cost of sending NACK packet; rather, it needs to consider the cost of sending NACK
feedback, and the importance of the lost packet, to make an informed feedback and the importance of the lost packet to make an informed
decision on whether it is worth telling the sender about a packet decision on whether it is worth telling the sender about a packet-
loss event. 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 it
be effective in environments with relatively low network RTT (an RTP can be effective in environments with relatively low network RTT.
sender can estimate the RTT to the receivers using the information in (An RTP sender can estimate the RTT to the receivers using the
RTCP SR and RR packets, as described at the end of Section 6.4.1 of information in RTCP SR and RR packets, as described at the end of
[RFC3550]). The use of retransmissions can also increase the forward Section 6.4.1 of [RFC3550]). The use of retransmissions can also
RTP bandwidth, and can potentially caused increased packet loss if increase the forward RTP bandwidth and can potentially cause
the original packet loss was caused by network congestion. Note, increased packet loss if the original packet loss was caused by
however, that retransmission of an important lost packet to repair network congestion. Note, however, that retransmission of an
decoder state can have lower cost than sending a full intra frame. important lost packet to repair decoder state can have lower cost
It is not appropriate to blindly retransmit RTP packets in response than sending a full intra frame. It is not appropriate to blindly
to a NACK. The importance of lost packets and the likelihood of them retransmit RTP packets in response to a NACK. The importance of lost
arriving in time to be useful needs to be considered before RTP packets and the likelihood of them arriving in time to be useful need
retransmission is used. to be considered before RTP retransmission is used.
Receivers are REQUIRED to implement support for RTP retransmission Receivers are REQUIRED to implement support for RTP retransmission
packets [RFC4588] sent using SSRC multiplexing, and MAY also support packets [RFC4588] sent using SSRC multiplexing and MAY also support
RTP retransmission packets sent using session multiplexing. Senders RTP retransmission packets sent using session multiplexing. Senders
MAY send RTP retransmission packets in response to NACKs if support MAY send RTP retransmission packets in response to NACKs if support
for the RTP retransmission payload format has been negotiated, and if for the RTP retransmission payload format has been negotiated and the
the sender believes it is useful to send a retransmission of the sender believes it is useful to send a retransmission of the
packet(s) referenced in the NACK. Senders do not need to retransmit packet(s) referenced in the NACK. Senders do not need to retransmit
every NACKed packet. 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, and others operate across RTP packets and can be
with any payload format. It needs to be noted that using redundant used with any payload format. Note that using redundant encoding or
encoding or FEC will lead to increased play out delay, which needs to FEC will lead to increased play-out delay, which needs to be
be considered when choosing FEC schemes and their parameters. considered when choosing FEC schemes and their parameters.
WebRTC endpoints MUST follow the recommendations for FEC use given in WebRTC endpoints MUST follow the recommendations for FEC use given in
[I-D.ietf-rtcweb-fec]. WebRTC endpoints MAY support other types of [RFC8854]. WebRTC endpoints MAY support other types of FEC, but
FEC, but these MUST be negotiated before they are used. these MUST be negotiated before they are used.
7. WebRTC Use of RTP: Rate Control and Media Adaptation 7. WebRTC Use of RTP: Rate Control and Media Adaptation
WebRTC will be used in heterogeneous network environments using a WebRTC will be used in heterogeneous network environments using a
variety of link technologies, including both wired and wireless variety of link technologies, including both wired and wireless
links, to interconnect potentially large groups of users around the links, to interconnect potentially large groups of users around the
world. As a result, the network paths between users can have widely world. As a result, the network paths between users can have widely
varying one-way delays, available bit-rates, load levels, and traffic varying one-way delays, available bitrates, load levels, and traffic
mixtures. Individual endpoints can send one or more RTP packet mixtures. Individual endpoints can send one or more RTP packet
streams to each participant, and there can be several participants. streams to each participant, and there can be several participants.
Each of these RTP packet streams can contain different types of Each of these RTP packet streams can contain different types of
media, and the type of media, bit rate, and number of RTP packet media, and the type of media, bitrate, and number of RTP packet
streams as well as transport-layer flows can be highly asymmetric. streams as well as transport-layer flows can be highly asymmetric.
Non-RTP traffic can share the network paths with RTP transport-layer Non-RTP traffic can share the network paths with RTP transport-layer
flows. Since the network environment is not predictable or stable, flows. Since the network environment is not predictable or stable,
WebRTC Endpoints MUST ensure that the RTP traffic they generate can WebRTC endpoints MUST ensure that the RTP traffic they generate can
adapt to match changes in the available network capacity. adapt to match changes in the available network capacity.
The quality of experience for users of WebRTC is very dependent on The quality of experience for users of WebRTC is very dependent on
effective adaptation of the media to the limitations of the network. effective adaptation of the media to the limitations of the network.
Endpoints have to be designed so they do not transmit significantly Endpoints have to be designed so they do not transmit significantly
more data than the network path can support, except for very short more data than the network path can support, except for very short
time periods, otherwise high levels of network packet loss or delay time periods; otherwise, high levels of network packet loss or delay
spikes will occur, causing media quality degradation. The limiting spikes will occur, causing media quality degradation. The limiting
factor on the capacity of the network path might be the link factor on the capacity of the network path might be the link
bandwidth, or it might be competition with other traffic on the link bandwidth, or it might be competition with other traffic on the link
(this can be non-WebRTC traffic, traffic due to other WebRTC flows, (this can be non-WebRTC traffic, traffic due to other WebRTC flows,
or even competition with other WebRTC flows in the same session). or even competition with other WebRTC flows in the same session).
An effective media congestion control algorithm is therefore an An effective media congestion control algorithm is therefore an
essential part of the WebRTC framework. However, at the time of this essential part of the WebRTC framework. However, at the time of this
writing, there is no standard congestion control algorithm that can writing, there is no standard congestion control algorithm that can
be used for interactive media applications such as WebRTC's flows. be used for interactive media applications such as WebRTC's flows.
Some requirements for congestion control algorithms for Some requirements for congestion control algorithms for
RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements]. RTCPeerConnections are discussed in [RFC8836]. If a standardized
If a standardized congestion control algorithm that satisfies these congestion control algorithm that satisfies these requirements is
requirements is developed in the future, this memo will need to be be developed in the future, this memo will need to be updated to mandate
updated to mandate its use. its use.
7.1. Boundary Conditions and Circuit Breakers 7.1. Boundary Conditions and Circuit Breakers
WebRTC Endpoints MUST implement the RTP circuit breaker algorithm WebRTC endpoints MUST implement the RTP circuit breaker algorithm
that is described in [I-D.ietf-avtcore-rtp-circuit-breakers]. The that is described in [RFC8083]. The RTP circuit breaker is designed
RTP circuit breaker is designed to enable applications to recognise to enable applications to recognize and react to situations of
and react to situations of extreme network congestion. However, extreme network congestion. However, since the RTP circuit breaker
since the RTP circuit breaker might not be triggered until congestion might not be triggered until congestion becomes extreme, it cannot be
becomes extreme, it cannot be considered a substitute for congestion considered a substitute for congestion control, and applications MUST
control, and applications MUST also implement congestion control to also implement congestion control to allow them to adapt to changes
allow them to adapt to changes in network capacity. The congestion in network capacity. The congestion control algorithm will have to
control algorithm will have to be proprietary until a standardized be proprietary until a standardized congestion control algorithm is
congestion control algorithm is available. Any future RTP congestion available. Any future RTP congestion control algorithms are expected
control algorithms are expected to operate within the envelope to operate within the envelope allowed by the circuit breaker.
allowed by the circuit breaker.
The session establishment signalling will also necessarily establish The session-establishment signaling will also necessarily establish
boundaries to which the media bit-rate will conform. The choice of boundaries to which the media bitrate will conform. The choice of
media codecs provides upper- and lower-bounds on the supported bit- media codecs provides upper and lower bounds on the supported
rates that the application can utilise to provide useful quality, and bitrates that the application can utilize to provide useful quality,
the packetisation choices that exist. In addition, the signalling and the packetization choices that exist. In addition, the signaling
channel can establish maximum media bit-rate boundaries using, for channel can establish maximum media bitrate boundaries using, for
example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF Temporary example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF TMMBR
Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of messages (see Section 5.1.6 of this memo). Signaled bandwidth
this memo). Signalled bandwidth limitations, such as SDP "b=AS:" or limitations, such as SDP "b=AS:" or "b=CT:" lines received from the
"b=CT:" lines received from the peer, MUST be followed when sending peer, MUST be followed when sending RTP packet streams. A WebRTC
RTP packet streams. A WebRTC Endpoint receiving media SHOULD signal endpoint receiving media SHOULD signal its bandwidth limitations.
its bandwidth limitations. These limitations have to be based on These limitations have to be based on known bandwidth limitations,
known bandwidth limitations, for example the capacity of the edge for example the capacity of the edge links.
links.
7.2. Congestion Control Interoperability and Legacy Systems 7.2. Congestion Control Interoperability and Legacy Systems
All endpoints that wish to interwork with WebRTC MUST implement RTCP All endpoints that wish to interwork with WebRTC MUST implement RTCP
and provide congestion feedback via the defined RTCP reporting and provide congestion feedback via the defined RTCP reporting
mechanisms. mechanisms.
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 have to RTCP RR packets every few seconds. Implementations that have to
interwork with such endpoints MUST ensure that they keep within the interwork with such endpoints MUST ensure that they keep within the
RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers] RTP circuit breaker [RFC8083] constraints to limit the congestion
constraints to limit the congestion they can cause. they can cause.
If a legacy endpoint supports RTP/AVPF, this enables negotiation of If a legacy endpoint 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 endpoint supports some useful parameter, and the possibility that the endpoint supports some useful
feedback format for congestion control purpose such as TMMBR feedback format for congestion control purposes such as TMMBR
[RFC5104]. Implementations that have to interwork with such [RFC5104]. Implementations that have to interwork with such
endpoints MUST ensure that they stay within the RTP circuit breaker endpoints MUST ensure that they stay within the RTP circuit breaker
[I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the [RFC8083] constraints to limit the congestion they can cause, but
congestion they can cause, but might find that they can achieve they might find that they can achieve better congestion response
better congestion response depending on the amount of feedback that depending on the amount of feedback that is available.
is available.
With proprietary congestion control algorithms issues can arise when With proprietary congestion control algorithms, issues can arise when
different algorithms and implementations interact in a communication different algorithms and implementations interact in a communication
session. If the different implementations have made different session. If the different implementations have made different
choices in regards to the type of adaptation, for example one sender choices in regards to the type of adaptation, for example one sender
based, and one receiver based, then one could end up in situation based, and one receiver based, then one could end up in a situation
where one direction is dual controlled, when the other direction is where one direction is dual controlled when the other direction is
not controlled. This memo cannot mandate behaviour for proprietary not controlled. This memo cannot mandate behavior for proprietary
congestion control algorithms, but implementations that use such congestion control algorithms, but implementations that use such
algorithms ought to be aware of this issue, and try to ensure that algorithms ought to be aware of this issue and try to ensure that
effective congestion control is negotiated for media flowing in both effective congestion control is negotiated for media flowing in both
directions. If the IETF were to standardise both sender- and directions. If the IETF were to standardize both sender- and
receiver-based congestion control algorithms for WebRTC traffic in receiver-based congestion control algorithms for WebRTC traffic in
the future, the issues of interoperability, control, and ensuring the future, the issues of interoperability, control, and ensuring
that both directions of media flow are congestion controlled would that both directions of media flow are congestion controlled would
also need to be considered. also need to be considered.
8. WebRTC Use of RTP: Performance Monitoring 8. WebRTC Use of RTP: Performance Monitoring
As described in Section 4.1, implementations are REQUIRED to generate As described in Section 4.1, implementations are REQUIRED to generate
RTCP Sender Report (SR) and Reception Report (RR) packets relating to RTCP Sender Report (SR) and Receiver Report (RR) packets relating to
the RTP packet streams they send and receive. These RTCP reports can the RTP packet streams they send and receive. These RTCP reports can
be used for performance monitoring purposes, since they include basic be used for performance monitoring purposes, since they include basic
packet loss and jitter statistics. packet-loss and jitter statistics.
A large number of additional performance metrics are supported by the A large number of additional performance metrics are supported by the
RTCP Extended Reports (XR) framework, see [RFC3611][RFC6792]. At the RTCP Extended Reports (XR) framework; see [RFC3611] and [RFC6792].
time of this writing, it is not clear what extended metrics are At the time of this writing, it is not clear what extended metrics
suitable for use in WebRTC, so there is no requirement that are suitable for use in WebRTC, so there is no requirement that
implementations generate RTCP XR packets. However, implementations implementations generate RTCP XR packets. However, implementations
that can use detailed performance monitoring data MAY generate RTCP that can use detailed performance monitoring data MAY generate RTCP
XR packets as appropriate. The use of RTCP XR packets SHOULD be XR packets as appropriate. The use of RTCP XR packets SHOULD be
signalled; implementations MUST ignore RTCP XR packets that are signaled; implementations MUST ignore RTCP XR packets that are
unexpected or not understood. unexpected or not understood.
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. In this case, future updates to this memo have to be made of WebRTC. In this case, future updates to this memo have to be made
following the Guidelines for Writers of RTP Payload Format following "Guidelines for Writers of RTP Payload Format
Specifications [RFC2736], How to Write an RTP Payload Format Specifications" [RFC2736], "How to Write an RTP Payload Format"
[I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP [RFC8088], and "Guidelines for Extending the RTP Control Protocol
Control Protocol [RFC5968], and SHOULD take into account any future (RTCP)" [RFC5968]. They also 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 will adopt RTP in others. Where possible, the WebRTC framework will adopt RTP
extensions that are of general utility, to enable easy implementation extensions that are of general utility, to enable easy implementation
of a gateway to other applications using RTP, rather than adopt of a gateway to other applications using RTP, rather than adopt
mechanisms that are narrowly targeted at specific WebRTC use cases. mechanisms that are narrowly targeted at specific WebRTC use cases.
10. Signalling Considerations 10. Signaling Considerations
RTP is built with the assumption that an external signalling channel RTP is built with the assumption that an external signaling channel
exists, and can be used to configure RTP sessions and their features. exists and can be used to configure RTP sessions and their features.
The basic configuration of an RTP session consists of the following The basic configuration of an RTP session consists of the following
parameters: parameters:
RTP Profile: The name of the RTP profile to be used in session. The RTP profile: The name of the RTP profile to be used in the session.
RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate The RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can
on basic level, as can their secure variants RTP/SAVP [RFC3711] interoperate on a basic level, as can their secure variants, RTP/
and RTP/SAVPF [RFC5124]. The secure variants of the profiles do SAVP [RFC3711] and RTP/SAVPF [RFC5124]. The secure variants of
not directly interoperate with the non-secure variants, due to the the profiles do not directly interoperate with the nonsecure
presence of additional header fields for authentication in SRTP variants, due to the presence of additional header fields for
packets and cryptographic transformation of the payload. WebRTC authentication in SRTP packets and cryptographic transformation of
requires the use of the RTP/SAVPF profile, and this MUST be the payload. WebRTC requires the use of the RTP/SAVPF profile,
signalled. Interworking functions might transform this into the and this MUST be signaled. Interworking functions might transform
RTP/SAVP profile for a legacy use case, by indicating to the this into the RTP/SAVP profile for a legacy use case by indicating
WebRTC Endpoint that the RTP/SAVPF is used and configuring a trr- to the WebRTC endpoint that the RTP/SAVPF is used and configuring
int value of 4 seconds. a "trr-int" value of 4 seconds.
Transport Information: Source and destination IP address(s) and Transport information: Source and destination IP address(es) and
ports for RTP and RTCP MUST be signalled for each RTP session. In ports for RTP and RTCP MUST be signaled for each RTP session. In
WebRTC these transport addresses will be provided by ICE [RFC5245] WebRTC, these transport addresses will be provided by Interactive
that signals candidates and arrives at nominated candidate address Connectivity Establishment (ICE) [RFC8445] that signals candidates
pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such and arrives at nominated candidate address pairs. If RTP and RTCP
that a single port, i.e. transport-layer flow, is used for RTP and multiplexing [RFC5761] is to be used such that a single port --
RTCP flows, this MUST be signalled (see Section 4.5). i.e., transport-layer flow -- is used for RTP and RTCP flows, this
MUST be signaled (see Section 4.5).
RTP Payload Types, media formats, and format parameters: The mapping RTP payload types, media formats, and format parameters: The mapping
between media type names (and hence the RTP payload formats to be between media type names (and hence the RTP payload formats to be
used), and the RTP payload type numbers MUST be signalled. Each used) and the RTP payload type numbers MUST be signaled. Each
media type MAY also have a number of media type parameters that media type MAY also have a number of media type parameters that
MUST also be signalled to configure the codec and RTP payload MUST also be signaled to configure the codec and RTP payload
format (the "a=fmtp:" line from SDP). Section 4.3 of this memo format (the "a=fmtp:" line from SDP). Section 4.3 of this memo
discusses requirements for uniqueness of payload types. discusses requirements for uniqueness of payload types.
RTP Extensions: The use of any additional RTP header extensions and RTP extensions: The use of any additional RTP header extensions and
RTCP packet types, including any necessary parameters, MUST be RTCP packet types, including any necessary parameters, MUST be
signalled. This signalling is to ensure that a WebRTC Endpoint's signaled. This signaling ensures that a WebRTC endpoint's
behaviour, especially when sending, of any extensions is behavior, especially when sending, is predictable and consistent.
predictable and consistent. For robustness, and for compatibility For robustness and compatibility with non-WebRTC systems that
with non-WebRTC systems that might be connected to a WebRTC might be connected to a WebRTC session via a gateway,
session via a gateway, implementations are REQUIRED to ignore implementations are REQUIRED to ignore unknown RTCP packets and
unknown RTCP packets and RTP header extensions (see also RTP header extensions (see also Section 4.1).
Section 4.1).
RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the RTCP bandwidth: Support for exchanging RTCP bandwidth values with
endpoints will be necessary. This SHALL be done as described in the endpoints will be necessary. This SHALL be done as described
"Session Description Protocol (SDP) Bandwidth Modifiers for RTP in "Session Description Protocol (SDP) Bandwidth Modifiers for RTP
Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or
something semantically equivalent. This also ensures that the something semantically equivalent. This also ensures that the
endpoints have a common view of the RTCP bandwidth. A common view endpoints have a common view of the RTCP bandwidth. A common view
of the RTCP bandwidth among different endpoints is important, to of the RTCP bandwidth among different endpoints is important to
prevent differences in RTCP packet timing and timeout intervals prevent differences in RTCP packet timing and timeout intervals
causing interoperability problems. causing interoperability problems.
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 the offer/
offer/answer model, but does require all the necessary parameters to answer model but does require all the necessary parameters to be
be agreed upon, and provided to the RTP implementation. Note that in agreed upon and provided to the RTP implementation. Note that in
WebRTC it will depend on the signalling model and API how these WebRTC, it will depend on the signaling model and API how these
parameters need to be configured but they will be need to either be parameters need to be configured, but they will need to either be set
set in the API or explicitly signalled between the peers. in the API or explicitly signaled between the peers.
11. WebRTC API Considerations 11. WebRTC API Considerations
The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and The WebRTC API [W3C.WebRTC] and the Media Capture and Streams API
Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses [W3C.WD-mediacapture-streams] define and use the concept of a
the concept of a MediaStream that consists of zero or more MediaStream that consists of zero or more MediaStreamTracks. A
MediaStreamTracks. A MediaStreamTrack is an individual stream of MediaStreamTrack is an individual stream of media from any type of
media from any type of media source like a microphone or a camera, media source, such as a microphone or a camera, but conceptual
but also conceptual sources, like a audio mix or a video composition, sources, like an audio mix or a video composition, are also possible.
are possible. The MediaStreamTracks within a MediaStream might need The MediaStreamTracks within a MediaStream might need to be
to be synchronized during play back. synchronized during playback.
A MediaStreamTrack's realisation in RTP in the context of an A MediaStreamTrack's realization in RTP, in the context of an
RTCPeerConnection consists of a source packet stream identified with RTCPeerConnection, consists of a source packet stream, identified by
an SSRC within an RTP session part of the RTCPeerConnection. The an SSRC, sent within an RTP session that is part of the
MediaStreamTrack can also result in additional packet streams, and RTCPeerConnection. The MediaStreamTrack can also result in
thus SSRCs, in the same RTP session. These can be dependent packet additional packet streams, and thus SSRCs, in the same RTP session.
streams from scalable encoding of the source stream associated with These can be dependent packet streams from scalable encoding of the
the MediaStreamTrack, if such a media encoder is used. They can also source stream associated with the MediaStreamTrack, if such a media
be redundancy packet streams, these are created when applying Forward encoder is used. They can also be redundancy packet streams; these
Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to are created when applying Forward Error Correction (Section 6.2) or
the source packet stream. RTP retransmission (Section 6.1) to the source packet stream.
It is important to note that the same media source can be feeding It is important to note that the same media source can be feeding
multiple MediaStreamTracks. As different sets of constraints or multiple MediaStreamTracks. As different sets of constraints or
other parameters can be applied to the MediaStreamTrack, each other parameters can be applied to the MediaStreamTrack, each
MediaStreamTrack instance added to a RTCPeerConnection SHALL result MediaStreamTrack instance added to an RTCPeerConnection SHALL result
in an independent source packet stream, with its own set of in an independent source packet stream with its own set of associated
associated packet streams, and thus different SSRC(s). It will packet streams and thus different SSRC(s). It will depend on applied
depend on applied constraints and parameters if the source stream and constraints and parameters if the source stream and the encoding
the encoding configuration will be identical between different configuration will be identical between different MediaStreamTracks
MediaStreamTracks sharing the same media source. If the encoding sharing the same media source. If the encoding parameters and
parameters and constraints are the same, an implementation could constraints are the same, an implementation could choose to use only
choose to use only one encoded stream to create the different RTP one encoded stream to create the different RTP packet streams. Note
packet streams. Note that such optimisations would need to take into that such optimizations would need to take into account that the
account that the constraints for one of the MediaStreamTracks can at constraints for one of the MediaStreamTracks can change at any
any moment change, meaning that the encoding configurations might no moment, meaning that the encoding configurations might no longer be
longer be identical and two different encoder instances would then be identical, and two different encoder instances would then be needed.
needed.
The same MediaStreamTrack can also be included in multiple The same MediaStreamTrack can also be included in multiple
MediaStreams, thus multiple sets of MediaStreams can implicitly need MediaStreams; thus, multiple sets of MediaStreams can implicitly need
to use the same synchronisation base. To ensure that this works in to use the same synchronization base. To ensure that this works in
all cases, and does not force an endpoint to disrupt the media by all cases and does not force an endpoint to disrupt the media by
changing synchronisation base and CNAME during delivery of any changing synchronization base and CNAME during delivery of any
ongoing packet streams, all MediaStreamTracks and their associated ongoing packet streams, all MediaStreamTracks and their associated
SSRCs originating from the same endpoint need to be sent using the SSRCs originating from the same endpoint need to be sent using the
same CNAME within one RTCPeerConnection. This is motivating the use same CNAME within one RTCPeerConnection. This is motivating the use
of a single CNAME in Section 4.9. of a single CNAME in Section 4.9.
The requirement on using the same CNAME for all SSRCs that | The requirement to use the same CNAME for all SSRCs that
originate from the same endpoint, does not require a middlebox | originate from the same endpoint does not require a middlebox
that forwards traffic from multiple endpoints to only use a single | that forwards traffic from multiple endpoints to only use a
CNAME. | single CNAME.
Different CNAMEs normally need to be used for different Different CNAMEs normally need to be used for different
RTCPeerConnection instances, as specified in Section 4.9. Having two RTCPeerConnection instances, as specified in Section 4.9. Having two
communication sessions with the same CNAME could enable tracking of a communication sessions with the same CNAME could enable tracking of a
user or device across different services (see Section 4.4.1 of user or device across different services (see Section 4.4.1 of
[I-D.ietf-rtcweb-security] for details). A web application can [RFC8826] for details). A web application can request that the
request that the CNAMEs used in different RTCPeerConnections (within CNAMEs used in different RTCPeerConnections (within a same-origin
a same-orign context) be the same, this allows for synchronization of context) be the same; this allows for synchronization of the
the endpoint's RTP packet streams across the different endpoint's RTP packet streams across the different
RTCPeerConnections. RTCPeerConnections.
Note: this doesn't result in a tracking issue, since the creation | Note: This doesn't result in a tracking issue, since the
of matching CNAMEs depends on existing tracking within a single | creation of matching CNAMEs depends on existing tracking within
origin. | a single origin.
The above will currently force a WebRTC Endpoint that receives a The above will currently force a WebRTC endpoint that receives a
MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing MediaStreamTrack on one RTCPeerConnection and adds it as outgoing one
on any RTCPeerConnection to perform resynchronisation of the stream. on any RTCPeerConnection to perform resynchronization of the stream.
Since the sending party needs to change the CNAME to the one it uses, Since the sending party needs to change the CNAME to the one it uses,
this implies it has to use a local system clock as timebase for the this implies it has to use a local system clock as the timebase for
synchronisation. Thus, the relative relation between the timebase of the synchronization. Thus, the relative relation between the
the incoming stream and the system sending out needs to be defined. timebase of the incoming stream and the system sending out needs to
This relation also needs monitoring for clock drift and likely be defined. This relation also needs monitoring for clock drift and
adjustments of the synchronisation. The sending entity is also likely adjustments of the synchronization. The sending entity is
responsible for congestion control for its sent streams. In cases of also responsible for congestion control for its sent streams. In
packet loss the loss of incoming data also needs to be handled. This cases of packet loss, the loss of incoming data also needs to be
leads to the observation that the method that is least likely to handled. This leads to the observation that the method that is least
cause issues or interruptions in the outgoing source packet stream is likely to cause issues or interruptions in the outgoing source packet
a model of full decoding, including repair etc., followed by encoding stream is a model of full decoding, including repair, followed by
of the media again into the outgoing packet stream. Optimisations of encoding of the media again into the outgoing packet stream.
this method are clearly possible and implementation specific. Optimizations of this method are clearly possible and implementation
specific.
A WebRTC Endpoint MUST support receiving multiple MediaStreamTracks, A WebRTC endpoint MUST support receiving multiple MediaStreamTracks,
where each of the different MediaStreamTracks (and their sets of where each of the different MediaStreamTracks (and its sets of
associated packet streams) uses different CNAMEs. However, associated packet streams) uses different CNAMEs. However,
MediaStreamTracks that are received with different CNAMEs have no MediaStreamTracks that are received with different CNAMEs have no
defined synchronisation. defined synchronization.
Note: The motivation for supporting reception of multiple CNAMEs | Note: The motivation for supporting reception of multiple
is to allow for forward compatibility with any future changes that | CNAMEs is to allow for forward compatibility with any future
enable more efficient stream handling when endpoints relay/forward | changes that enable more efficient stream handling when
streams. It also ensures that endpoints can interoperate with | endpoints relay/forward streams. It also ensures that
certain types of multi-stream middleboxes or endpoints that are | endpoints can interoperate with certain types of multistream
not WebRTC. | middleboxes or endpoints that are not WebRTC.
Javascript Session Establishment Protocol [I-D.ietf-rtcweb-jsep] "JavaScript Session Establishment Protocol (JSEP)" [RFC8829]
specifies that the binding between the WebRTC MediaStreams, specifies that the binding between the WebRTC MediaStreams,
MediaStreamTracks and the SSRC is done as specified in "Cross Session MediaStreamTracks, and the SSRC is done as specified in "WebRTC
Stream Identification in the Session Description Protocol" MediaStream Identification in the Session Description Protocol"
[I-D.ietf-mmusic-msid]. The MSID document [I-D.ietf-mmusic-msid] [RFC8830]. Section 4.1 of the MediaStream Identification (MSID)
also defines, in section 4.1, how to map unknown source packet stream document [RFC8830] also defines how to map source packet streams with
SSRCs to MediaStreamTracks and MediaStreams. This later is relevant unknown SSRCs to MediaStreamTracks and MediaStreams. This later is
to handle some cases of legacy interoperability. Commonly the RTP relevant to handle some cases of legacy interoperability. Commonly,
Payload Type of any incoming packets will reveal if the packet stream the RTP payload type of any incoming packets will reveal if the
is a source stream or a redundancy or dependent packet stream. The packet stream is a source stream or a redundancy or dependent packet
association to the correct source packet stream depends on the stream. The association to the correct source packet stream depends
payload format in use for the packet stream. on the payload format in use for the packet stream.
Finally this specification puts a requirement on the WebRTC API to Finally, this specification puts a requirement on the WebRTC API to
realize a method for determining the CSRC list (Section 4.1) as well realize a method for determining the CSRC list (Section 4.1) as well
as the Mixer-to-Client audio levels (Section 5.2.3) (when supported) as the mixer-to-client audio levels (Section 5.2.3) (when supported);
and the basic requirements for this is further discussed in the basic requirements for this is further discussed in
Section 12.2.1. Section 12.2.1.
12. RTP Implementation Considerations 12. RTP Implementation Considerations
The following discussion provides some guidance on the implementation The following discussion provides some guidance on the implementation
of the RTP features described in this memo. The focus is on a WebRTC of the RTP features described in this memo. The focus is on a WebRTC
Endpoint implementation perspective, and while some mention is made endpoint implementation perspective, and while some mention is made
of the behaviour of middleboxes, that is not the focus of this memo. of the behavior of middleboxes, that is not the focus of this memo.
12.1. Configuration and Use of RTP Sessions 12.1. Configuration and Use of RTP Sessions
A WebRTC Endpoint will be a simultaneous participant in one or more A WebRTC endpoint will be a simultaneous participant in one or more
RTP sessions. Each RTP session can convey multiple media sources, RTP sessions. Each RTP session can convey multiple media sources and
and can include media data from multiple endpoints. In the include media data from multiple endpoints. In the following, some
following, some ways in which WebRTC Endpoints can configure and use ways in which WebRTC endpoints can configure and use RTP sessions are
RTP sessions are outlined. outlined.
12.1.1. Use of Multiple Media Sources Within an RTP Session 12.1.1. Use of Multiple Media Sources within an RTP Session
RTP is a group communication protocol, and every RTP session can RTP is a group communication protocol, and every RTP session can
potentially contain multiple RTP packet streams. There are several potentially contain multiple RTP packet streams. There are several
reasons why this might be desirable: reasons why this might be desirable:
Multiple media types: Outside of WebRTC, it is common to use one RTP * Multiple media types:
session for each type of media source (e.g., one RTP session for
audio sources and one for video sources, each sent over different
transport layer flows). However, to reduce the number of UDP
ports used, the default in WebRTC is to send all types of media in
a single RTP session, as described in Section 4.4, using RTP and
RTCP multiplexing (Section 4.5) to further reduce the number of
UDP ports needed. This RTP session then uses only one bi-
directional transport-layer flow, but will contain multiple RTP
packet streams, each containing a different type of media. A
common example might be an endpoint with a camera and microphone
that sends two RTP packet streams, one video and one audio, into a
single RTP session.
Multiple Capture Devices: A WebRTC Endpoint might have multiple Outside of WebRTC, it is common to use one RTP session for each
cameras, microphones, or other media capture devices, and so might type of media source (e.g., one RTP session for audio sources and
want to generate several RTP packet streams of the same media one for video sources, each sent over different transport-layer
type. Alternatively, it might want to send media from a single flows). However, to reduce the number of UDP ports used, the
capture device in several different formats or quality settings at default in WebRTC is to send all types of media in a single RTP
once. Both can result in a single endpoint sending multiple RTP session, as described in Section 4.4, using RTP and RTCP
packet streams of the same media type into a single RTP session at multiplexing (Section 4.5) to further reduce the number of UDP
the same time. ports needed. This RTP session then uses only one bidirectional
transport-layer flow but will contain multiple RTP packet streams,
each containing a different type of media. A common example might
be an endpoint with a camera and microphone that sends two RTP
packet streams, one video and one audio, into a single RTP
session.
Associated Repair Data: An endpoint might send a RTP packet stream * Multiple capture devices:
that is somehow associated with another stream. For example, it
might send an RTP packet stream that contains FEC or
retransmission data relating to another stream. Some RTP payload
formats send this sort of associated repair data as part of the
source packet stream, while others send it as a separate packet
stream.
Layered or Multiple Description Coding: An endpoint can use a A WebRTC endpoint might have multiple cameras, microphones, or
layered media codec, for example H.264 SVC, or a multiple other media capture devices, and so it might want to generate
description codec, that generates multiple RTP packet streams, several RTP packet streams of the same media type. Alternatively,
each with a distinct RTP SSRC, within a single RTP session. it might want to send media from a single capture device in
several different formats or quality settings at once. Both can
result in a single endpoint sending multiple RTP packet streams of
the same media type into a single RTP session at the same time.
RTP Mixers, Translators, and Other Middleboxes: An RTP session, in * Associated repair data:
the WebRTC context, is a point-to-point association between an
endpoint and some other peer device, where those devices share a An endpoint might send an RTP packet stream that is somehow
common SSRC space. The peer device might be another WebRTC associated with another stream. For example, it might send an RTP
Endpoint, or it might be an RTP mixer, translator, or some other packet stream that contains FEC or retransmission data relating to
form of media processing middlebox. In the latter cases, the another stream. Some RTP payload formats send this sort of
middlebox might send mixed or relayed RTP streams from several associated repair data as part of the source packet stream, while
participants, that the WebRTC Endpoint will need to render. Thus, others send it as a separate packet stream.
even though a WebRTC Endpoint might only be a member of a single
RTP session, the peer device might be extending that RTP session * Layered or multiple-description coding:
to incorporate other endpoints. WebRTC is a group communication
environment and endpoints need to be capable of receiving, Within a single RTP session, an endpoint can use a layered media
decoding, and playing out multiple RTP packet streams at once, codec -- for example, H.264 Scalable Video Coding (SVC) -- or a
even in a single RTP session. multiple-description codec that generates multiple RTP packet
streams, each with a distinct RTP SSRC.
* RTP mixers, translators, and other middleboxes:
An RTP session, in the WebRTC context, is a point-to-point
association between an endpoint and some other peer device, where
those devices share a common SSRC space. The peer device might be
another WebRTC endpoint, or it might be an RTP mixer, translator,
or some other form of media-processing middlebox. In the latter
cases, the middlebox might send mixed or relayed RTP streams from
several participants, which the WebRTC endpoint will need to
render. Thus, even though a WebRTC endpoint might only be a
member of a single RTP session, the peer device might be extending
that RTP session to incorporate other endpoints. WebRTC is a
group communication environment, and endpoints need to be capable
of receiving, decoding, and playing out multiple RTP packet
streams at once, even in a single RTP session.
12.1.2. Use of Multiple RTP Sessions 12.1.2. Use of Multiple RTP Sessions
In addition to sending and receiving multiple RTP packet streams In addition to sending and receiving multiple RTP packet streams
within a single RTP session, a WebRTC Endpoint might participate in within a single RTP session, a WebRTC endpoint might participate in
multiple RTP sessions. There are several reasons why a WebRTC multiple RTP sessions. There are several reasons why a WebRTC
Endpoint might choose to do this: endpoint might choose to do this:
To interoperate with legacy devices: The common practice in the non- * To interoperate with legacy devices:
WebRTC world is to send different types of media in separate RTP
sessions, for example using one RTP session for audio and another
RTP session, on a separate transport layer flow, for video. All
WebRTC Endpoints need to support the option of sending different
types of media on different RTP sessions, so they can interwork
with such legacy devices. This is discussed further in
Section 4.4.
To provide enhanced quality of service: Some network-based quality The common practice in the non-WebRTC world is to send different
of service mechanisms operate on the granularity of transport types of media in separate RTP sessions -- for example, using one
layer flows. If it is desired to use these mechanisms to provide RTP session for audio and another RTP session, on a separate
differentiated quality of service for some RTP packet streams, transport-layer flow, for video. All WebRTC endpoints need to
then those RTP packet streams need to be sent in a separate RTP support the option of sending different types of media on
session using a different transport-layer flow, and with different RTP sessions so they can interwork with such legacy
appropriate quality of service marking. This is discussed further devices. This is discussed further in Section 4.4.
in Section 12.1.3.
To separate media with different purposes: An endpoint might want to * To provide enhanced quality of service:
send RTP packet streams that have different purposes on different
RTP sessions, to make it easy for the peer device to distinguish
them. For example, some centralised multiparty conferencing
systems display the active speaker in high resolution, but show
low resolution "thumbnails" of other participants. Such systems
might configure the endpoints to send simulcast high- and low-
resolution versions of their video using separate RTP sessions, to
simplify the operation of the RTP middlebox. In the WebRTC
context this is currently possible by establishing multiple WebRTC
MediaStreamTracks that have the same media source in one (or more)
RTCPeerConnection. Each MediaStreamTrack is then configured to
deliver a particular media quality and thus media bit-rate, and
will produce an independently encoded version with the codec
parameters agreed specifically in the context of that
RTCPeerConnection. The RTP middlebox can distinguish packets
corresponding to the low- and high-resolution streams by
inspecting their SSRC, RTP payload type, or some other information
contained in RTP payload, RTP header extension or RTCP packets,
but it can be easier to distinguish the RTP packet streams if they
arrive on separate RTP sessions on separate transport-layer flows.
To directly connect with multiple peers: A multi-party conference Some network-based quality-of-service mechanisms operate on the
does not need to use an RTP middlebox. Rather, a multi-unicast granularity of transport-layer flows. If use of these mechanisms
mesh can be created, comprising several distinct RTP sessions, to provide differentiated quality of service for some RTP packet
with each participant sending RTP traffic over a separate RTP streams is desired, then those RTP packet streams need to be sent
session (that is, using an independent RTCPeerConnection object) in a separate RTP session using a different transport-layer flow,
to every other participant, as shown in Figure 1. This topology and with appropriate quality-of-service marking. This is
has the benefit of not requiring an RTP middlebox node that is discussed further in Section 12.1.3.
trusted to access and manipulate the media data. The downside is
that it increases the used bandwidth at each sender by requiring
one copy of the RTP packet streams for each participant that are
part of the same session beyond the sender itself.
+---+ +---+ * To separate media with different purposes:
| A |<--->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 1: Multi-unicast using several RTP sessions An endpoint might want to send RTP packet streams that have
different purposes on different RTP sessions, to make it easy for
the peer device to distinguish them. For example, some
centralized multiparty conferencing systems display the active
speaker in high resolution but show low-resolution "thumbnails" of
other participants. Such systems might configure the endpoints to
send simulcast high- and low-resolution versions of their video
using separate RTP sessions to simplify the operation of the RTP
middlebox. In the WebRTC context, this is currently possible by
establishing multiple WebRTC MediaStreamTracks that have the same
media source in one (or more) RTCPeerConnection. Each
MediaStreamTrack is then configured to deliver a particular media
quality and thus media bitrate, and it will produce an
independently encoded version with the codec parameters agreed
specifically in the context of that RTCPeerConnection. The RTP
middlebox can distinguish packets corresponding to the low- and
high-resolution streams by inspecting their SSRC, RTP payload
type, or some other information contained in RTP payload, RTP
header extension, or RTCP packets. However, it can be easier to
distinguish the RTP packet streams if they arrive on separate RTP
sessions on separate transport-layer flows.
* To directly connect with multiple peers:
A multiparty conference does not need to use an RTP middlebox.
Rather, a multi-unicast mesh can be created, comprising several
distinct RTP sessions, with each participant sending RTP traffic
over a separate RTP session (that is, using an independent
RTCPeerConnection object) to every other participant, as shown in
Figure 1. This topology has the benefit of not requiring an RTP
middlebox node that is trusted to access and manipulate the media
data. The downside is that it increases the used bandwidth at
each sender by requiring one copy of the RTP packet streams for
each participant that is part of the same session beyond the
sender itself.
+---+ +---+
| A |<--->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 1: Multi-unicast Using Several RTP Sessions
The multi-unicast topology could also be implemented as a single The multi-unicast topology could also be implemented as a single
RTP session, spanning multiple peer-to-peer transport layer RTP session, spanning multiple peer-to-peer transport-layer
connections, or as several pairwise RTP sessions, one between each connections, or as several pairwise RTP sessions, one between each
pair of peers. To maintain a coherent mapping of the relationship pair of peers. To maintain a coherent mapping of the relationship
between RTP sessions and RTCPeerConnection objects it is recommend between RTP sessions and RTCPeerConnection objects, it is
that this is implemented as several individual RTP sessions. The RECOMMENDED that this be implemented as several individual RTP
only downside is that endpoint A will not learn of the quality of sessions. The only downside is that endpoint A will not learn of
any transmission happening between B and C, since it will not see the quality of any transmission happening between B and C, since
RTCP reports for the RTP session between B and C, whereas it would it will not see RTCP reports for the RTP session between B and C,
if all three participants were part of a single RTP session. whereas it would if all three participants were part of a single
Experience with the Mbone tools (experimental RTP-based multicast RTP session. Experience with the Mbone tools (experimental RTP-
conferencing tools from the late 1990s) has showed that RTCP based multicast conferencing tools from the late 1990s) has shown
reception quality reports for third parties can be presented to that RTCP reception quality reports for third parties can be
users in a way that helps them understand asymmetric network presented to users in a way that helps them understand asymmetric
problems, and the approach of using separate RTP sessions prevents network problems, and the approach of using separate RTP sessions
this. However, an advantage of using separate RTP sessions is prevents this. However, an advantage of using separate RTP
that it enables using different media bit-rates and RTP session sessions is that it enables using different media bitrates and RTP
configurations between the different peers, thus not forcing B to session configurations between the different peers, thus not
endure the same quality reductions if there are limitations in the forcing B to endure the same quality reductions as C will if there
transport from A to C as C will. It is believed that these are limitations in the transport from A to C. It is believed that
advantages outweigh the limitations in debugging power. these advantages outweigh the limitations in debugging power.
To indirectly connect with multiple peers: A common scenario in * To indirectly connect with multiple peers:
multi-party conferencing is to create indirect connections to
multiple peers, using an RTP mixer, translator, or some other type
of RTP middlebox. Figure 2 outlines a simple topology that might
be used in a four-person centralised conference. The middlebox
acts to optimise the transmission of RTP packet streams from
certain perspectives, either by only sending some of the received
RTP packet stream to any given receiver, or by providing a
combined RTP packet stream out of a set of contributing streams.
+---+ +-------------+ +---+ A common scenario in multiparty conferencing is to create indirect
| A |<---->| |<---->| B | connections to multiple peers, using an RTP mixer, translator, or
+---+ | RTP mixer, | +---+ some other type of RTP middlebox. Figure 2 outlines a simple
| translator, | topology that might be used in a four-person centralized
| or other | conference. The middlebox acts to optimize the transmission of
+---+ | middlebox | +---+ RTP packet streams from certain perspectives, either by only
| C |<---->| |<---->| D | sending some of the received RTP packet stream to any given
+---+ +-------------+ +---+ receiver, or by providing a combined RTP packet stream out of a
set of contributing streams.
Figure 2: RTP mixer with only unicast paths +---+ +-------------+ +---+
| A |<---->| |<---->| B |
+---+ | RTP mixer, | +---+
| translator, |
| or other |
+---+ | middlebox | +---+
| C |<---->| |<---->| D |
+---+ +-------------+ +---+
Figure 2: RTP Mixer with Only Unicast Paths
There are various methods of implementation for the middlebox. If There are various methods of implementation for the middlebox. If
implemented as a standard RTP mixer or translator, a single RTP implemented as a standard RTP mixer or translator, a single RTP
session will extend across the middlebox and encompass all the session will extend across the middlebox and encompass all the
endpoints in one multi-party session. Other types of middlebox endpoints in one multiparty session. Other types of middleboxes
might use separate RTP sessions between each endpoint and the might use separate RTP sessions between each endpoint and the
middlebox. A common aspect is that these RTP middleboxes can use middlebox. A common aspect is that these RTP middleboxes can use
a number of tools to control the media encoding provided by a a number of tools to control the media encoding provided by a
WebRTC Endpoint. This includes functions like requesting the WebRTC endpoint. This includes functions like requesting the
breaking of the encoding chain and have the encoder produce a so breaking of the encoding chain and having the encoder produce a
called Intra frame. Another is limiting the bit-rate of a given so-called Intra frame. Another common aspect is limiting the
stream to better suit the mixer view of the multiple down-streams. bitrate of a stream to better match the mixed output. Other
Others are controlling the most suitable frame-rate, picture aspects are controlling the most suitable frame rate, picture
resolution, the trade-off between frame-rate and spatial quality. resolution, and the trade-off between frame rate and spatial
The middlebox has the responsibility to correctly perform quality. The middlebox has the responsibility to correctly
congestion control, source identification, manage synchronisation perform congestion control, identify sources, and manage
while providing the application with suitable media optimisations. synchronization while providing the application with suitable
The middlebox also has to be a trusted node when it comes to media optimizations. The middlebox also has to be a trusted node
security, since it manipulates either the RTP header or the media when it comes to security, since it manipulates either the RTP
itself (or both) received from one endpoint, before sending it on header or the media itself (or both) received from one endpoint
towards the endpoint(s), thus they need to be able to decrypt and before sending them on towards the endpoint(s); thus they need to
then re-encrypt the RTP packet stream before sending it out. be able to decrypt and then re-encrypt the RTP packet stream
before sending it out.
RTP Mixers can create a situation where an endpoint experiences a Mixers are expected to not forward RTCP reports regarding RTP
situation in-between a session with only two endpoints and packet streams across themselves. This is due to the difference
multiple RTP sessions. Mixers are expected to not forward RTCP between the RTP packet streams provided to the different
reports regarding RTP packet streams across themselves. This is endpoints. The original media source lacks information about a
due to the difference in the RTP packet streams provided to the mixer's manipulations prior to being sent to the different
different endpoints. The original media source lacks information receivers. This scenario also results in an endpoint's feedback
about a mixer's manipulations prior to sending it the different or requests going to the mixer. When the mixer can't act on this
receivers. This scenario also results in that an endpoint's by itself, it is forced to go to the original media source to
feedback or requests go to the mixer. When the mixer can't act on fulfill the receiver's request. This will not necessarily be
this by itself, it is forced to go to the original media source to
fulfil the receivers request. This will not necessarily be
explicitly visible to any RTP and RTCP traffic, but the explicitly visible to any RTP and RTCP traffic, but the
interactions and the time to complete them will indicate such interactions and the time to complete them will indicate such
dependencies. dependencies.
Providing source authentication in multi-party scenarios is a Providing source authentication in multiparty scenarios is a
challenge. In the mixer-based topologies, endpoints source challenge. In the mixer-based topologies, endpoints source
authentication is based on, firstly, verifying that media comes authentication is based on, firstly, verifying that media comes
from the mixer by cryptographic verification and, secondly, trust from the mixer by cryptographic verification and, secondly, trust
in the mixer to correctly identify any source towards the in the mixer to correctly identify any source towards the
endpoint. In RTP sessions where multiple endpoints are directly endpoint. In RTP sessions where multiple endpoints are directly
visible to an endpoint, all endpoints will have knowledge about visible to an endpoint, all endpoints will have knowledge about
each others' master keys, and can thus inject packets claimed to each others' master keys and can thus inject packets claiming to
come from another endpoint in the session. Any node performing come from another endpoint in the session. Any node performing
relay can perform non-cryptographic mitigation by preventing relay can perform noncryptographic mitigation by preventing
forwarding of packets that have SSRC fields that came from other forwarding of packets that have SSRC fields that came from other
endpoints before. For cryptographic verification of the source, endpoints before. For cryptographic verification of the source,
SRTP would require additional security mechanisms, for example SRTP would require additional security mechanisms -- for example,
TESLA for SRTP [RFC4383], that are not part of the base WebRTC Timed Efficient Stream Loss-Tolerant Authentication (TESLA) for
standards. SRTP [RFC4383] -- that are not part of the base WebRTC standards.
To forward media between multiple peers: It is sometimes desirable * To forward media between multiple peers:
for an endpoint that receives an RTP packet stream to be able to
forward that RTP packet stream to a third party. The are some It is sometimes desirable for an endpoint that receives an RTP
obvious security and privacy implications in supporting this, but packet stream to be able to forward that RTP packet stream to a
also potential uses. This is supported in the W3C API by taking third party. The are some obvious security and privacy
the received and decoded media and using it as media source that implications in supporting this, but also potential uses. This is
is re-encoding and transmitted as a new stream. supported in the W3C API by taking the received and decoded media
and using it as a media source that is re-encoded and transmitted
as a new stream.
At the RTP layer, media forwarding acts as a back-to-back RTP At the RTP layer, media forwarding acts as a back-to-back RTP
receiver and RTP sender. The receiving side terminates the RTP receiver and RTP sender. The receiving side terminates the RTP
session and decodes the media, while the sender side re-encodes session and decodes the media, while the sender side re-encodes
and transmits the media using an entirely separate RTP session. and transmits the media using an entirely separate RTP session.
The original sender will only see a single receiver of the media, The original sender will only see a single receiver of the media,
and will not be able to tell that forwarding is happening based on and will not be able to tell that forwarding is happening based on
RTP-layer information since the RTP session that is used to send RTP-layer information, since the RTP session that is used to send
the forwarded media is not connected to the RTP session on which the forwarded media is not connected to the RTP session on which
the media was received by the node doing the forwarding. the media was received by the node doing the forwarding.
The endpoint that is performing the forwarding is responsible for The endpoint that is performing the forwarding is responsible for
producing an RTP packet stream suitable for onwards transmission. producing an RTP packet stream suitable for onwards transmission.
The outgoing RTP session that is used to send the forwarded media The outgoing RTP session that is used to send the forwarded media
is entirely separate to the RTP session on which the media was is entirely separate from the RTP session on which the media was
received. This will require media transcoding for congestion received. This will require media transcoding for congestion
control purpose to produce a suitable bit-rate for the outgoing control purposes to produce a suitable bitrate for the outgoing
RTP session, reducing media quality and forcing the forwarding RTP session, reducing media quality and forcing the forwarding
endpoint to spend the resource on the transcoding. The media endpoint to spend the resource on the transcoding. The media
transcoding does result in a separation of the two different legs transcoding does result in a separation of the two different legs,
removing almost all dependencies, and allowing the forwarding removing almost all dependencies, and allowing the forwarding
endpoint to optimise its media transcoding operation. The cost is endpoint to optimize its media transcoding operation. The cost is
greatly increased computational complexity on the forwarding node. greatly increased computational complexity on the forwarding node.
Receivers of the forwarded stream will see the forwarding device Receivers of the forwarded stream will see the forwarding device
as the sender of the stream, and will not be able to tell from the as the sender of the stream and will not be able to tell from the
RTP layer that they are receiving a forwarded stream rather than RTP layer that they are receiving a forwarded stream rather than
an entirely new RTP packet stream generated by the forwarding an entirely new RTP packet stream generated by the forwarding
device. device.
12.1.3. Differentiated Treatment of RTP Streams 12.1.3. Differentiated Treatment of RTP Streams
There are use cases for differentiated treatment of RTP packet There are use cases for differentiated treatment of RTP packet
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 endpoint system. First of all is the prioritization within the endpoint
sending the media, which controls, both which RTP packet streams that sending the media, which controls both which RTP packet streams will
will be sent, and their allocation of bit-rate out of the current be sent and their allocation of bitrate out of the current available
available aggregate as determined by the congestion control. aggregate, as determined by the congestion control.
It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will It is expected that the WebRTC API [W3C.WebRTC] will allow the
allow the application to indicate relative priorities for different application to indicate relative priorities for different
MediaStreamTracks. These priorities can then be used to influence MediaStreamTracks. These priorities can then be used to influence
the local RTP processing, especially when it comes to congestion the local RTP processing, especially when it comes to determining how
control response in how to divide the available bandwidth between the to divide the available bandwidth between the RTP packet streams for
RTP packet streams. Any changes in relative priority will also need the sake of congestion control. Any changes in relative priority
to be considered for RTP packet streams that are associated with the will also need to be considered for RTP packet streams that are
main RTP packet streams, such as redundant streams for RTP associated with the main RTP packet streams, such as redundant
retransmission and FEC. The importance of such redundant RTP packet streams for RTP retransmission and FEC. The importance of such
streams is dependent on the media type and codec used, in regards to redundant RTP packet streams is dependent on the media type and codec
how robust that codec is to packet loss. However, a default policy used, with regard to how robust that codec is against packet loss.
might to be to use the same priority for redundant RTP packet stream However, a default policy might be to use the same priority for a
as for the source RTP packet stream. redundant RTP packet stream as for the source RTP packet stream.
Secondly, the network can prioritize transport-layer flows and sub- Secondly, the network can prioritize transport-layer flows and
flows, including RTP packet streams. Typically, differential subflows, including RTP packet streams. Typically, differential
treatment includes two steps, the first being identifying whether an treatment includes two steps, the first being identifying whether an
IP packet belongs to a class that has to be treated differently, the IP packet belongs to a class that has to be treated differently, the
second consisting of the actual mechanism to prioritize packets. second consisting of the actual mechanism for prioritizing packets.
Three common methods for classifying IP packets are: Three common methods for classifying IP packets are:
DiffServ: The endpoint marks a packet with a DiffServ code point to DiffServ: The endpoint 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 need to 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.
Flow-based differentiation will provide the same treatment to all Flow-based differentiation will provide the same treatment to all
packets within a transport-layer flow, i.e., relative prioritization packets within a transport-layer flow, i.e., relative prioritization
is not possible. Moreover, if the resources are limited it might not is not possible. Moreover, if the resources are limited, it might
be possible to provide differential treatment compared to best-effort not be possible to provide differential treatment compared to best
for all the RTP packet streams used in a WebRTC session. The use of effort for all the RTP packet streams used in a WebRTC session. The
flow-based differentiation needs to be coordinated between the WebRTC use of flow-based differentiation needs to be coordinated between the
system and the network(s). The WebRTC endpoint needs to know that WebRTC system and the network(s). The WebRTC endpoint needs to know
flow-based differentiation might be used to provide the separation of that flow-based differentiation might be used to provide the
the RTP packet streams onto different UDP flows to enable a more separation of the RTP packet streams onto different UDP flows to
granular usage of flow based differentiation. The used flows, their enable a more granular usage of flow-based differentiation. The used
5-tuples and prioritization will need to be communicated to the flows, their 5-tuples, and prioritization will need to be
network so that it can identify the flows correctly to enable communicated to the network so that it can identify the flows
prioritization. No specific protocol support for this is specified. correctly to enable prioritization. No specific protocol support for
this is specified.
DiffServ assumes that either the endpoint or a classifier can mark DiffServ assumes that either the endpoint or a classifier can mark
the packets with an appropriate DSCP so that the packets are treated the packets with an appropriate Differentiated Services Code Point
according to that marking. If the endpoint is to mark the traffic (DSCP) so that the packets are treated according to that marking. If
two requirements arise in the WebRTC context: 1) The WebRTC Endpoint the endpoint is to mark the traffic, two requirements arise in the
has to know which DSCP to use and that it can use them on some set of WebRTC context: 1) The WebRTC endpoint has to know which DSCPs to use
RTP packet streams. 2) The information needs to be propagated to the and know that it can use them on some set of RTP packet streams. 2)
operating system when transmitting the packet. Details of this The information needs to be propagated to the operating system when
process are outside the scope of this memo and are further discussed transmitting the packet. Details of this process are outside the
in "DSCP and other packet markings for RTCWeb QoS" scope of this memo and are further discussed in "Differentiated
[I-D.ietf-tsvwg-rtcweb-qos]. Services Code Point (DSCP) Packet Markings for WebRTC QoS" [RFC8837].
Deep Packet Inspectors will, despite the SRTP media encryption, still Despite the SRTP media encryption, deep packet inspectors will still
be fairly capable at classifying the RTP streams. The reason is that be fairly capable of classifying the RTP streams. The reason is that
SRTP leaves the first 12 bytes of the RTP header unencrypted. This SRTP leaves the first 12 bytes of the RTP header unencrypted. This
enables easy RTP stream identification using the SSRC and provides enables easy RTP stream identification using the SSRC and provides
the classifier with useful information that can be correlated to the classifier with useful information that can be correlated to
determine for example the stream's media type. Using packet sizes, determine, for example, the stream's media type. Using packet sizes,
reception times, packet inter-spacing, RTP timestamp increments and reception times, packet inter-spacing, RTP timestamp increments, and
sequence numbers, fairly reliable classifications are achieved. sequence numbers, fairly reliable classifications are achieved.
For packet based marking schemes it might be possible to mark For packet-based marking schemes, it might be possible to mark
individual RTP packets differently based on the relative priority of individual RTP packets differently based on the relative priority of
the RTP payload. For example video codecs that have I, P, and B the RTP payload. For example, video codecs that have I, P, and B
pictures could prioritise any payloads carrying only B frames less, pictures could prioritize any payloads carrying only B frames less,
as these are less damaging to loose. However, depending on the QoS as these are less damaging to lose. However, depending on the QoS
mechanism and what markings that are applied, this can result in not mechanism and what markings are applied, this can result in not only
only different packet drop probabilities but also packet reordering, different packet-drop probabilities but also packet reordering; see
see [I-D.ietf-tsvwg-rtcweb-qos] and [I-D.ietf-dart-dscp-rtp] for [RFC8837] and [RFC7657] for further discussion. As a default policy,
further discussion. As a default policy all RTP packets related to a all RTP packets related to an RTP packet stream ought to be provided
RTP packet stream ought to be provided with the same prioritization; with the same prioritization; per-packet prioritization is outside
per-packet prioritization is outside the scope of this memo, but the scope of this memo but might be specified elsewhere in future.
might be specified elsewhere in future.
It is also important to consider how RTCP packets associated with a It is also important to consider how RTCP packets associated with a
particular RTP packet stream need to be marked. RTCP compound particular RTP packet stream need to be marked. RTCP compound
packets with Sender Reports (SR), ought to be marked with the same packets with Sender Reports (SRs) ought to be marked with the same
priority as the RTP packet stream itself, so the RTCP-based round- priority as the RTP packet stream itself, so the RTCP-based round-
trip time (RTT) measurements are done using the same transport-layer trip time (RTT) measurements are done using the same transport-layer
flow priority as the RTP packet stream experiences. RTCP compound flow priority as the RTP packet stream experiences. RTCP compound
packets containing RR packet ought to be sent with the priority used packets containing an RR packet ought to be sent with the priority
by the majority of the RTP packet streams reported on. RTCP packets used by the majority of the RTP packet streams reported on. RTCP
containing time-critical feedback packets can use higher priority to packets containing time-critical feedback packets can use higher
improve the timeliness and likelihood of delivery of such feedback. priority to improve the timeliness and likelihood of delivery of such
feedback.
12.2. Media Source, RTP Streams, and Participant Identification 12.2. Media Source, RTP Streams, and Participant Identification
12.2.1. Media Source Identification 12.2.1. Media Source Identification
Each RTP packet stream is identified by a unique synchronisation Each RTP packet stream is identified by a unique synchronization
source (SSRC) identifier. The SSRC identifier is carried in each of source (SSRC) identifier. The SSRC identifier is carried in each of
the RTP packets comprising a RTP packet stream, and is also used to the RTP packets comprising an RTP packet stream, and is also used to
identify that stream in the corresponding RTCP reports. The SSRC is identify that stream in the corresponding RTCP reports. The SSRC is
chosen as discussed in Section 4.8. The first stage in chosen as discussed in Section 4.8. The first stage in
demultiplexing RTP and RTCP packets received on a single transport demultiplexing RTP and RTCP packets received on a single transport-
layer flow at a WebRTC Endpoint is to separate the RTP packet streams layer flow at a WebRTC endpoint is to separate the RTP packet streams
based on their SSRC value; once that is done, additional based on their SSRC value; once that is done, additional
demultiplexing steps can determine how and where to render the media. demultiplexing steps can determine how and where to render the media.
RTP allows a mixer, or other RTP-layer middlebox, to combine encoded RTP allows a mixer, or other RTP-layer middlebox, to combine encoded
streams from multiple media sources to form a new encoded stream from streams from multiple media sources to form a new encoded stream from
a new media source (the mixer). The RTP packets in that new RTP a new media source (the mixer). The RTP packets in that new RTP
packet stream can include a Contributing Source (CSRC) list, packet stream can include a contributing source (CSRC) list,
indicating which original SSRCs contributed to the combined source indicating which original SSRCs contributed to the combined source
stream. As described in Section 4.1, implementations need to support stream. As described in Section 4.1, implementations need to support
reception of RTP data packets containing a CSRC list and RTCP packets reception of RTP data packets containing a CSRC list and RTCP packets
that relate to sources present in the CSRC list. The CSRC list can that relate to sources present in the CSRC list. The CSRC list can
change on a packet-by-packet basis, depending on the mixing operation change on a packet-by-packet basis, depending on the mixing operation
being performed. Knowledge of what media sources contributed to a being performed. Knowledge of what media sources contributed to a
particular RTP packet can be important if the user interface particular RTP packet can be important if the user interface
indicates which participants are active in the session. Changes in indicates which participants are active in the session. Changes in
the CSRC list included in packets needs to be exposed to the WebRTC the CSRC list included in packets need to be exposed to the WebRTC
application using some API, if the application is to be able to track application using some API if the application is to be able to track
changes in session participation. It is desirable to map CSRC values changes in session participation. It is desirable to map CSRC values
back into WebRTC MediaStream identities as they cross this API, to back into WebRTC MediaStream identities as they cross this API, to
avoid exposing the SSRC/CSRC name space to WebRTC applications. avoid exposing the SSRC/CSRC namespace to WebRTC applications.
If the mixer-to-client audio level extension [RFC6465] is being used If the mixer-to-client audio level extension [RFC6465] is being used
in the session (see Section 5.2.3), the information in the CSRC list in the session (see Section 5.2.3), the information in the CSRC list
is augmented by audio level information for each contributing source. is augmented by audio-level information for each contributing source.
It is desirable to expose this information to the WebRTC application It is desirable to expose this information to the WebRTC application
using some API, after mapping the CSRC values to WebRTC MediaStream using some API, after mapping the CSRC values to WebRTC MediaStream
identities, so it can be exposed in the user interface. identities, so it can be exposed in the user interface.
12.2.2. SSRC Collision Detection 12.2.2. SSRC Collision Detection
The RTP standard requires RTP implementations to have support for The RTP standard requires RTP implementations to have support for
detecting and handling SSRC collisions, i.e., resolve the conflict detecting and handling SSRC collisions -- i.e., be able to resolve
when two different endpoints use the same SSRC value (see section 8.2 the conflict when two different endpoints use the same SSRC value
of [RFC3550]). This requirement also applies to WebRTC Endpoints. (see Section 8.2 of [RFC3550]). This requirement also applies to
There are several scenarios where SSRC collisions can occur: WebRTC endpoints. There are several scenarios where SSRC collisions
can occur:
o In a point-to-point session where each SSRC is associated with * In a point-to-point session where each SSRC is associated with
either of the two endpoints and where the main media carrying SSRC either of the two endpoints and the main media-carrying SSRC
identifier will be announced in the signalling channel, a identifier will be announced in the signaling channel, a collision
collision is less likely to occur due to the information about is less likely to occur due to the information about used SSRCs.
used SSRCs. If SDP is used, this information is provided by If SDP is used, this information is provided by source-specific
Source-Specific SDP Attributes [RFC5576]. Still, collisions can SDP attributes [RFC5576]. Still, collisions can occur if both
occur if both endpoints start using a new SSRC identifier prior to endpoints start using a new SSRC identifier prior to having
having signalled it to the peer and received acknowledgement on signaled it to the peer and received acknowledgement on the
the signalling message. The Source-Specific SDP Attributes signaling message. "Source-Specific Media Attributes in the
[RFC5576] contains a mechanism to signal how the endpoint resolved Session Description Protocol (SDP)" [RFC5576] contains a mechanism
the SSRC collision. to signal how the endpoint resolved the SSRC collision.
o SSRC values that have not been signalled could also appear in an * SSRC values that have not been signaled could also appear in an
RTP session. This is more likely than it appears, since some RTP RTP session. This is more likely than it appears, since some RTP
functions use extra SSRCs to provide their functionality. For functions use extra SSRCs to provide their functionality. For
example, retransmission data might be transmitted using a separate example, retransmission data might be transmitted using a separate
RTP packet stream that requires its own SSRC, separate to the SSRC RTP packet stream that requires its own SSRC, separate from the
of the source RTP packet stream [RFC4588]. In those cases, an SSRC of the source RTP packet stream [RFC4588]. In those cases,
endpoint can create a new SSRC that strictly doesn't need to be an endpoint can create a new SSRC that strictly doesn't need to be
announced over the signalling channel to function correctly on announced over the signaling channel to function correctly on both
both RTP and RTCPeerConnection level. RTP and RTCPeerConnection level.
o Multiple endpoints in a multiparty conference can create new * Multiple endpoints in a multiparty conference can create new
sources and signal those towards the RTP middlebox. In cases sources and signal those towards the RTP middlebox. In cases
where the SSRC/CSRC are propagated between the different endpoints where the SSRC/CSRC are propagated between the different endpoints
from the RTP middlebox collisions can occur. from the RTP middlebox, collisions can occur.
o An RTP middlebox could connect an endpoint's RTCPeerConnection to * An RTP middlebox could connect an endpoint's RTCPeerConnection to
another RTCPeerConnection from the same endpoint, thus forming a another RTCPeerConnection from the same endpoint, thus forming a
loop where the endpoint will receive its own traffic. While it is loop where the endpoint will receive its own traffic. While it is
clearly considered a bug, it is important that the endpoint is clearly considered a bug, it is important that the endpoint be
able to recognise and handle the case when it occurs. This case able to recognize and handle the case when it occurs. This case
becomes even more problematic when media mixers, and so on, are becomes even more problematic when media mixers and such are
involved, where the stream received is a different stream but involved, where the stream received is a different stream but
still contains this client's input. still contains this client's input.
These SSRC/CSRC collisions can only be handled on RTP level as long These SSRC/CSRC collisions can only be handled on the RTP level when
as the same RTP session is extended across multiple the same RTP session is extended across multiple RTCPeerConnections
RTCPeerConnections by a RTP middlebox. To resolve the more generic by an RTP middlebox. To resolve the more generic case where multiple
case where multiple RTCPeerConnections are interconnected, RTCPeerConnections are interconnected, identification of the media
identification of the media source(s) part of a MediaStreamTrack source or sources that are part of a MediaStreamTrack being
being propagated across multiple interconnected RTCPeerConnection propagated across multiple interconnected RTCPeerConnection needs to
needs to be preserved across these interconnections. be preserved across these interconnections.
12.2.3. Media Synchronisation Context 12.2.3. Media Synchronization Context
When an endpoint sends media from more than one media source, it When an endpoint 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, synchronization is provided by having a
set of RTP packet streams be indicated as coming from the same set of RTP packet streams be indicated as coming from the same
synchronisation context and logical endpoint by using the same RTCP synchronization context and logical endpoint by using the same RTCP
CNAME identifier. CNAME identifier.
The next provision is that the internal clocks of all media sources, The next provision is that the internal clocks of all media sources
i.e., what drives the RTP timestamp, can be correlated to a system -- i.e., what drives the RTP timestamp -- can be correlated to a
clock that is provided in RTCP Sender Reports encoded in an NTP system clock that is provided in RTCP Sender Reports encoded in an
format. By correlating all RTP timestamps to a common system clock NTP format. By correlating all RTP timestamps to a common system
for all sources, the timing relation of the different RTP packet clock for all sources, the timing relation of the different RTP
streams, also across multiple RTP sessions can be derived at the packet streams, also across multiple RTP sessions, can be derived at
receiver and, if desired, the streams can be synchronized. The 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; whether or not the information is used is up to the
receiver.
13. Security Considerations 13. Security Considerations
The overall security architecture for WebRTC is described in The overall security architecture for WebRTC is described in
[I-D.ietf-rtcweb-security-arch], and security considerations for the [RFC8827], and security considerations for the WebRTC framework are
WebRTC framework are described in [I-D.ietf-rtcweb-security]. These described in [RFC8826]. These considerations also apply to this
considerations also apply to this memo. memo.
The security considerations of the RTP specification, the RTP/SAVPF The security considerations of the RTP specification, the RTP/SAVPF
profile, and the various RTP/RTCP extensions and RTP payload formats profile, and the various RTP/RTCP extensions and RTP payload formats
that form the complete protocol suite described in this memo apply. that form the complete protocol suite described in this memo apply.
It is not believed there are any new security considerations It is believed that there are no new security considerations
resulting from the combination of these various protocol extensions. resulting from the combination of these various protocol extensions.
The Extended Secure RTP Profile for Real-time Transport Control "Extended Secure RTP Profile for Real-time Transport Control Protocol
Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124] provides handling of
handling of fundamental issues by offering confidentiality, integrity fundamental issues by offering confidentiality, integrity, and
and partial source authentication. A mandatory to implement and use partial source authentication. A media-security solution that is
media security solution is created by combining this secured RTP mandatory to implement and use is created by combining this secured
profile and DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of RTP profile and DTLS-SRTP keying [RFC5764], as defined by Section 5.5
[I-D.ietf-rtcweb-security-arch]. of [RFC8827].
RTCP packets convey a Canonical Name (CNAME) identifier that is used RTCP packets convey a Canonical Name (CNAME) identifier that is used
to associate RTP packet streams that need to be synchronised across to associate RTP packet streams that need to be synchronized across
related RTP sessions. Inappropriate choice of CNAME values can be a related RTP sessions. Inappropriate choice of CNAME values can be a
privacy concern, since long-term persistent CNAME identifiers can be privacy concern, since long-term persistent CNAME identifiers can be
used to track users across multiple WebRTC calls. Section 4.9 of used to track users across multiple WebRTC calls. Section 4.9 of
this memo mandates generation of short-term persistent RTCP CNAMES, this memo mandates generation of short-term persistent RTCP CNAMES,
as specified in RFC7022, resulting in untraceable CNAME values that as specified in RFC 7022, resulting in untraceable CNAME values that
alleviate this risk. alleviate this risk.
Some potential denial of service attacks exist if the RTCP reporting Some potential denial-of-service attacks exist if the RTCP reporting
interval is configured to an inappropriate value. This could be done interval is configured to an inappropriate value. This could be done
by configuring the RTCP bandwidth fraction to an excessively large or by configuring the RTCP bandwidth fraction to an excessively large or
small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556] or some
similar mechanism, or by choosing an excessively large or small value similar mechanism, or by choosing an excessively large or small value
for the RTP/AVPF minimal receiver report interval (if using SDP, this for the RTP/AVPF minimal receiver report interval (if using SDP, this
is the "a=rtcp-fb:... trr-int" parameter) [RFC4585]. The risks are is the "a=rtcp-fb:... trr-int" parameter) [RFC4585]. The risks are
as follows: as follows:
1. the RTCP bandwidth could be configured to make the regular 1. the RTCP bandwidth could be configured to make the regular
reporting interval so large that effective congestion control reporting interval so large that effective congestion control
cannot be maintained, potentially leading to denial of service cannot be maintained, potentially leading to denial of service
due to congestion caused by the media traffic; due to congestion caused by the media traffic;
2. the RTCP interval could be configured to a very small value, 2. the RTCP interval could be configured to a very small value,
causing endpoints to generate high rate RTCP traffic, potentially causing endpoints to generate high-rate RTCP traffic, potentially
leading to denial of service due to the non-congestion controlled leading to denial of service due to the RTCP traffic not being
RTCP traffic; and congestion controlled; and
3. RTCP parameters could be configured differently for each 3. RTCP parameters could be configured differently for each
endpoint, with some of the endpoints using a large reporting endpoint, with some of the endpoints using a large reporting
interval and some using a smaller interval, leading to denial of interval and some using a smaller interval, leading to denial of
service due to premature participant timeouts due to mismatched service due to premature participant timeouts due to mismatched
timeout periods which are based on the reporting interval (this timeout periods that are based on the reporting interval. This
is a particular concern if endpoints use a small but non-zero is a particular concern if endpoints use a small but nonzero
value for the RTP/AVPF minimal receiver report interval (trr-int) value for the RTP/AVPF minimal receiver report interval (trr-int)
[RFC4585], as discussed in Section 6.1 of [RFC4585], as discussed in Section 6.1 of [RFC8108].
[I-D.ietf-avtcore-rtp-multi-stream]).
Premature participant timeout can be avoided by using the fixed (non- Premature participant timeout can be avoided by using the fixed
reduced) minimum interval when calculating the participant timeout (nonreduced) minimum interval when calculating the participant
(see Section 4.1 of this memo and Section 6.1 of timeout (see Section 4.1 of this memo and Section 7.1.2 of
[I-D.ietf-avtcore-rtp-multi-stream]). To address the other concerns, [RFC8108]). To address the other concerns, endpoints SHOULD ignore
endpoints SHOULD ignore parameters that configure the RTCP reporting parameters that configure the RTCP reporting interval to be
interval to be significantly longer than the default five second significantly longer than the default five-second interval specified
interval specified in [RFC3550] (unless the media data rate is so low in [RFC3550] (unless the media data rate is so low that the longer
that the longer reporting interval roughly corresponds to 5% of the reporting interval roughly corresponds to 5% of the media data rate),
media data rate), or that configure the RTCP reporting interval small or that configure the RTCP reporting interval small enough that the
enough that the RTCP bandwidth would exceed the media bandwidth. RTCP bandwidth would exceed the media bandwidth.
The guidelines in [RFC6562] apply when using variable bit rate (VBR) The guidelines in [RFC6562] apply when using variable bitrate (VBR)
audio codecs such as Opus (see Section 4.3 for discussion of mandated audio codecs such as Opus (see Section 4.3 for discussion of mandated
audio codecs). The guidelines in [RFC6562] also apply, but are of audio codecs). The guidelines in [RFC6562] also apply, but are of
lesser importance, when using the client-to-mixer audio level header lesser importance, when using the client-to-mixer audio level header
extensions (Section 5.2.2) or the mixer-to-client audio level header extensions (Section 5.2.2) or the mixer-to-client audio level header
extensions (Section 5.2.3). The use of the encryption of the header extensions (Section 5.2.3). The use of the encryption of the header
extensions are RECOMMENDED, unless there are known reasons, like RTP extensions are RECOMMENDED, unless there are known reasons, like RTP
middleboxes performing voice activity based source selection or third middleboxes performing voice-activity-based source selection or
party monitoring that will greatly benefit from the information, and third-party monitoring that will greatly benefit from the
this has been expressed using API or signalling. If further evidence information, and this has been expressed using API or signaling. If
are produced to show that information leakage is significant from further evidence is produced to show that information leakage is
audio level indications, then use of encryption needs to be mandated significant from audio-level indications, then use of encryption
at that time. needs to be mandated at that time.
In multi-party communication scenarios using RTP Middleboxes, a lot In multiparty communication scenarios using RTP middleboxes, a lot of
of trust is placed on these middleboxes to preserve the sessions trust is placed on these middleboxes to preserve the session's
security. The middlebox needs to maintain the confidentiality, security. The middlebox needs to maintain confidentiality and
integrity and perform source authentication. As discussed in integrity and perform source authentication. As discussed in
Section 12.1.1 the middlebox can perform checks that prevents any Section 12.1.1, the middlebox can perform checks that prevent any
endpoint participating in a conference to impersonate another. Some endpoint participating in a conference from impersonating another.
additional security considerations regarding multi-party topologies Some additional security considerations regarding multiparty
can be found in [I-D.ietf-avtcore-rtp-topologies-update]. topologies can be found in [RFC7667].
14. IANA Considerations 14. IANA Considerations
This memo makes no request of IANA. This document has no IANA actions.
Note to RFC Editor: this section is to be removed on publication as
an RFC.
15. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand,
Cary Bran, Ben Campbell, Alissa Cooper, Spencer Dawkins, Charles
Eckel, Alex Eleftheriadis, Christian Groves, Chris Inacio, Cullen
Jennings, Olle Johansson, Suhas Nandakumar, Dan Romascanu, Jim
Spring, Martin Thomson, and the other members of the IETF RTCWEB
working group for their valuable feedback.
16. References
16.1. Normative References
[I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-13 (work in
progress), December 2015.
[I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Varun, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", draft-ietf-
avtcore-rtp-circuit-breakers-13 (work in progress),
February 2016.
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
December 2015.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-12 (work
in progress), March 2016.
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-10 (work in progress),
July 2015.
[I-D.ietf-mmusic-mux-exclusive]
Holmberg, C., "Indicating Exclusive Support of RTP/RTCP
Multiplexing using SDP", draft-ietf-mmusic-mux-
exclusive-03 (work in progress), February 2016.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-27 (work in progress), February 2016.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-10 (work in
progress), February 2016.
[I-D.ietf-rtcweb-fec]
Uberti, J., "WebRTC Forward Error Correction
Requirements", draft-ietf-rtcweb-fec-02 (work in
progress), October 2015.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-15
(work in progress), January 2016.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-08 (work in progress), February 2015.
[I-D.ietf-rtcweb-security-arch] 15. References
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-11 (work in progress), March 2015.
[I-D.ietf-rtcweb-video] 15.1. Normative References
Roach, A., "WebRTC Video Processing and Codec
Requirements", draft-ietf-rtcweb-video-06 (work in
progress), June 2015.
[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, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
[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,
DOI 10.17487/RFC2736, December 1999, DOI 10.17487/RFC2736, December 1999,
<http://www.rfc-editor.org/info/rfc2736>. <https://www.rfc-editor.org/info/rfc2736>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>. July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003, DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>. <https://www.rfc-editor.org/info/rfc3551>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, DOI 10.17487/RFC3556, July 2003, RFC 3556, DOI 10.17487/RFC3556, July 2003,
<http://www.rfc-editor.org/info/rfc3556>. <https://www.rfc-editor.org/info/rfc3556>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004, RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>. <https://www.rfc-editor.org/info/rfc3711>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566, Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>. July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006, DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>. <https://www.rfc-editor.org/info/rfc4585>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588, Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006, DOI 10.17487/RFC4588, July 2006,
<http://www.rfc-editor.org/info/rfc4588>. <https://www.rfc-editor.org/info/rfc4588>.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007, BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
<http://www.rfc-editor.org/info/rfc4961>. <https://www.rfc-editor.org/info/rfc4961>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile "Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>. February 2008, <https://www.rfc-editor.org/info/rfc5104>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>. 2008, <https://www.rfc-editor.org/info/rfc5124>.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
2008, <http://www.rfc-editor.org/info/rfc5285>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>. 2009, <https://www.rfc-editor.org/info/rfc5506>.
[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, Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010, DOI 10.17487/RFC5761, April 2010,
<http://www.rfc-editor.org/info/rfc5761>. <https://www.rfc-editor.org/info/rfc5761>.
[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, Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010, DOI 10.17487/RFC5764, May 2010,
<http://www.rfc-editor.org/info/rfc5764>. <https://www.rfc-editor.org/info/rfc5764>.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, DOI 10.17487/RFC6051, November 2010, Flows", RFC 6051, DOI 10.17487/RFC6051, November 2010,
<http://www.rfc-editor.org/info/rfc6051>. <https://www.rfc-editor.org/info/rfc6051>.
[RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time [RFC6464] Lennox, J., Ed., 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, Mixer Audio Level Indication", RFC 6464,
DOI 10.17487/RFC6464, December 2011, DOI 10.17487/RFC6464, December 2011,
<http://www.rfc-editor.org/info/rfc6464>. <https://www.rfc-editor.org/info/rfc6464>.
[RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real- [RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
time Transport Protocol (RTP) Header Extension for Mixer- time Transport Protocol (RTP) Header Extension for Mixer-
to-Client Audio Level Indication", RFC 6465, to-Client Audio Level Indication", RFC 6465,
DOI 10.17487/RFC6465, December 2011, DOI 10.17487/RFC6465, December 2011,
<http://www.rfc-editor.org/info/rfc6465>. <https://www.rfc-editor.org/info/rfc6465>.
[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,
DOI 10.17487/RFC6562, March 2012, DOI 10.17487/RFC6562, March 2012,
<http://www.rfc-editor.org/info/rfc6562>. <https://www.rfc-editor.org/info/rfc6562>.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904, Real-time Transport Protocol (SRTP)", RFC 6904,
DOI 10.17487/RFC6904, April 2013, DOI 10.17487/RFC6904, April 2013,
<http://www.rfc-editor.org/info/rfc6904>. <https://www.rfc-editor.org/info/rfc6904>.
[RFC7007] Terriberry, T., "Update to Remove DVI4 from the [RFC7007] Terriberry, T., "Update to Remove DVI4 from the
Recommended Codecs for the RTP Profile for Audio and Video Recommended Codecs for the RTP Profile for Audio and Video
Conferences with Minimal Control (RTP/AVP)", RFC 7007, Conferences with Minimal Control (RTP/AVP)", RFC 7007,
DOI 10.17487/RFC7007, August 2013, DOI 10.17487/RFC7007, August 2013,
<http://www.rfc-editor.org/info/rfc7007>. <https://www.rfc-editor.org/info/rfc7007>.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP) "Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022, Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
September 2013, <http://www.rfc-editor.org/info/rfc7022>. September 2013, <https://www.rfc-editor.org/info/rfc7022>.
[RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple [RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
Clock Rates in an RTP Session", RFC 7160, Clock Rates in an RTP Session", RFC 7160,
DOI 10.17487/RFC7160, April 2014, DOI 10.17487/RFC7160, April 2014,
<http://www.rfc-editor.org/info/rfc7160>. <https://www.rfc-editor.org/info/rfc7160>.
[RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", [RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds",
RFC 7164, DOI 10.17487/RFC7164, March 2014, RFC 7164, DOI 10.17487/RFC7164, March 2014,
<http://www.rfc-editor.org/info/rfc7164>. <https://www.rfc-editor.org/info/rfc7164>.
[W3C.WD-mediacapture-streams-20130903] [RFC7742] Roach, A.B., "WebRTC Video Processing and Codec
Burnett, D., Bergkvist, A., Jennings, C., and A. Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
Narayanan, "Media Capture and Streams", World Wide Web <https://www.rfc-editor.org/info/rfc7742>.
Consortium WD WD-mediacapture-streams-20130903, September
2013, <http://www.w3.org/TR/2013/
WD-mediacapture-streams-20130903>.
[W3C.WD-webrtc-20130910] [RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
Bergkvist, A., Burnett, D., Jennings, C., and A. Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
Narayanan, "WebRTC 1.0: Real-time Communication Between <https://www.rfc-editor.org/info/rfc7874>.
Browsers", World Wide Web Consortium WD WD-webrtc-
20130910, September 2013,
<http://www.w3.org/TR/2013/WD-webrtc-20130910>.
16.2. Informative References [RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", RFC 8083,
DOI 10.17487/RFC8083, March 2017,
<https://www.rfc-editor.org/info/rfc8083>.
[I-D.ietf-avtcore-multiplex-guidelines] [RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
Westerlund, M., Perkins, C., and H. Alvestrand, "Sending Multiple RTP Streams in a Single RTP Session",
"Guidelines for using the Multiplexing Features of RTP to RFC 8108, DOI 10.17487/RFC8108, March 2017,
Support Multiple Media Streams", draft-ietf-avtcore- <https://www.rfc-editor.org/info/rfc8108>.
multiplex-guidelines-03 (work in progress), October 2014.
[I-D.ietf-avtext-rtp-grouping-taxonomy] [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
B. Burman, "A Taxonomy of Semantics and Mechanisms for May 2017, <https://www.rfc-editor.org/info/rfc8174>.
Real-Time Transport Protocol (RTP) Sources", draft-ietf-
avtext-rtp-grouping-taxonomy-08 (work in progress), July
2015.
[I-D.ietf-dart-dscp-rtp] [RFC8285] Singer, D., Desineni, H., and R. Even, Ed., "A General
Black, D. and P. Jones, "Differentiated Services Mechanism for RTP Header Extensions", RFC 8285,
(DiffServ) and Real-time Communication", draft-ietf-dart- DOI 10.17487/RFC8285, October 2017,
dscp-rtp-10 (work in progress), November 2014. <https://www.rfc-editor.org/info/rfc8285>.
[I-D.ietf-mmusic-msid] [RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Alvestrand, H., "WebRTC MediaStream Identification in the Browser-Based Applications", RFC 8825,
Session Description Protocol", draft-ietf-mmusic-msid-11 DOI 10.17487/RFC8825, January 2021,
(work in progress), October 2015. <https://www.rfc-editor.org/info/rfc8825>.
[I-D.ietf-payload-rtp-howto] [RFC8826] Rescorla, E., "Security Considerations for WebRTC",
Westerlund, M., "How to Write an RTP Payload Format", RFC 8826, DOI 10.17487/RFC8826, January 2021,
draft-ietf-payload-rtp-howto-14 (work in progress), May <https://www.rfc-editor.org/info/rfc8826>.
2015.
[I-D.ietf-rmcat-cc-requirements] [RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
Jesup, R. and Z. Sarker, "Congestion Control Requirements DOI 10.17487/RFC8827, January 2021,
for Interactive Real-Time Media", draft-ietf-rmcat-cc- <https://www.rfc-editor.org/info/rfc8827>.
requirements-09 (work in progress), December 2014.
[I-D.ietf-rtcweb-jsep] [RFC8843] Holmberg, C., Alvestrand, H., and C. Jennings,
Uberti, J., Jennings, C., and E. Rescorla, "Javascript "Negotiating Media Multiplexing Using the Session
Session Establishment Protocol", draft-ietf-rtcweb-jsep-13 Description Protocol (SDP)", RFC 8843,
(work in progress), March 2016. DOI 10.17487/RFC8843, January 2021,
<https://www.rfc-editor.org/info/rfc8843>.
[I-D.ietf-tsvwg-rtcweb-qos] [RFC8854] Uberti, J., "WebRTC Forward Error Correction
Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP Requirements", RFC 8854, DOI 10.17487/RFC8854, January
and other packet markings for WebRTC QoS", draft-ietf- 2021, <https://www.rfc-editor.org/info/rfc8854>.
tsvwg-rtcweb-qos-14 (work in progress), March 2016.
[RFC8858] Holmberg, C., "Indicating Exclusive Support of RTP and RTP
Control Protocol (RTCP) Multiplexing Using the Session
Description Protocol (SDP)", RFC 8858,
DOI 10.17487/RFC8858, January 2021,
<https://www.rfc-editor.org/info/rfc8858>.
[RFC8860] Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session",
RFC 8860, DOI 10.17487/RFC8860, January 2021,
<https://www.rfc-editor.org/info/rfc8860>.
[RFC8861] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session:
Grouping RTP Control Protocol (RTCP) Reception Statistics
and Other Feedback", RFC 8861, DOI 10.17487/RFC8861,
January 2021, <https://www.rfc-editor.org/info/rfc8861>.
[W3C.WD-mediacapture-streams]
Jennings, C., Aboba, B., Bruaroey, J-I., and H. Boström,
"Media Capture and Streams", W3C Candidate Recommendation,
<https://www.w3.org/TR/mediacapture-streams/>.
[W3C.WebRTC]
Jennings, C., Boström, H., and J-I. Bruaroey, "WebRTC 1.0:
Real-time Communication Between Browsers", W3C Proposed
Recommendation, <https://www.w3.org/TR/webrtc/>.
15.2. Informative References
[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed., [RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
"RTP Control Protocol Extended Reports (RTCP XR)", "RTP Control Protocol Extended Reports (RTCP XR)",
RFC 3611, DOI 10.17487/RFC3611, November 2003, RFC 3611, DOI 10.17487/RFC3611, November 2003,
<http://www.rfc-editor.org/info/rfc3611>. <https://www.rfc-editor.org/info/rfc3611>.
[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,
DOI 10.17487/RFC4383, February 2006, DOI 10.17487/RFC4383, February 2006,
<http://www.rfc-editor.org/info/rfc4383>. <https://www.rfc-editor.org/info/rfc4383>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010,
<http://www.rfc-editor.org/info/rfc5245>.
[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, DOI 10.17487/RFC5576, June 2009, (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<http://www.rfc-editor.org/info/rfc5576>. <https://www.rfc-editor.org/info/rfc5576>.
[RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP
Control Protocol (RTCP)", RFC 5968, DOI 10.17487/RFC5968, Control Protocol (RTCP)", RFC 5968, DOI 10.17487/RFC5968,
September 2010, <http://www.rfc-editor.org/info/rfc5968>. September 2010, <https://www.rfc-editor.org/info/rfc5968>.
[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, Control Protocol (RTCP) Flows", RFC 6263,
DOI 10.17487/RFC6263, June 2011, DOI 10.17487/RFC6263, June 2011,
<http://www.rfc-editor.org/info/rfc6263>. <https://www.rfc-editor.org/info/rfc6263>.
[RFC6792] Wu, Q., Ed., Hunt, G., and P. Arden, "Guidelines for Use [RFC6792] Wu, Q., Ed., Hunt, G., and P. Arden, "Guidelines for Use
of the RTP Monitoring Framework", RFC 6792, of the RTP Monitoring Framework", RFC 6792,
DOI 10.17487/RFC6792, November 2012, DOI 10.17487/RFC6792, November 2012,
<http://www.rfc-editor.org/info/rfc6792>. <https://www.rfc-editor.org/info/rfc6792>.
[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478, Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015, DOI 10.17487/RFC7478, March 2015,
<http://www.rfc-editor.org/info/rfc7478>. <https://www.rfc-editor.org/info/rfc7478>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<https://www.rfc-editor.org/info/rfc7656>.
[RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services
(Diffserv) and Real-Time Communication", RFC 7657,
DOI 10.17487/RFC7657, November 2015,
<https://www.rfc-editor.org/info/rfc7657>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
[RFC8088] Westerlund, M., "How to Write an RTP Payload Format",
RFC 8088, DOI 10.17487/RFC8088, May 2017,
<https://www.rfc-editor.org/info/rfc8088>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
[RFC8829] Uberti, J., Jennings, C., and E. Rescorla, Ed.,
"JavaScript Session Establishment Protocol (JSEP)",
RFC 8829, DOI 10.17487/RFC8829, January 2021,
<https://www.rfc-editor.org/info/rfc8829>.
[RFC8830] Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", RFC 8830,
DOI 10.17487/RFC8830, January 2021,
<https://www.rfc-editor.org/info/rfc8830>.
[RFC8836] Jesup, R. and Z. Sarker, Ed., "Congestion Control
Requirements for Interactive Real-Time Media", RFC 8836,
DOI 10.17487/RFC8836, January 2021,
<https://www.rfc-editor.org/info/rfc8836>.
[RFC8837] Jones, P., Dhesikan, S., Jennings, C., and D. Druta,
"Differentiated Services Code Point (DSCP) Packet Markings
for WebRTC QoS", RFC 8837, DOI 10.17487/RFC8837, January
2021, <https://www.rfc-editor.org/info/rfc8837>.
[RFC8872] Westerlund, M., Burman, B., Perkins, C., Alvestrand, H.,
and R. Even, "Guidelines for Using the Multiplexing
Features of RTP to Support Multiple Media Streams",
RFC 8872, DOI 10.17487/RFC8872, January 2021,
<https://www.rfc-editor.org/info/rfc8872>.
Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand,
Cary Bran, Ben Campbell, Alissa Cooper, Spencer Dawkins, Charles
Eckel, Alex Eleftheriadis, Christian Groves, Chris Inacio, Cullen
Jennings, Olle Johansson, Suhas Nandakumar, Dan Romascanu, Jim
Spring, Martin Thomson, and the other members of the IETF RTCWEB
working group for their valuable feedback.
Authors' Addresses Authors' Addresses
Colin Perkins Colin Perkins
University of Glasgow University of Glasgow
School of Computing Science School of Computing Science
Glasgow G12 8QQ Glasgow
G12 8QQ
United Kingdom United Kingdom
Email: csp@csperkins.org Email: csp@csperkins.org
URI: https://csperkins.org/ URI: https://csperkins.org/
Magnus Westerlund Magnus Westerlund
Ericsson Ericsson
Farogatan 6 Torshamnsgatan 23
SE-164 80 Kista SE-164 80 Kista
Sweden Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com Email: magnus.westerlund@ericsson.com
Joerg Ott Jörg Ott
Aalto University Technical University Munich
School of Electrical Engineering Department of Informatics
Espoo 02150 Chair of Connected Mobility
Finland Boltzmannstrasse 3
85748 Garching
Germany
Email: jorg.ott@aalto.fi Email: ott@in.tum.de
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