draft-ietf-rtcweb-rtp-usage-24.txt   draft-ietf-rtcweb-rtp-usage-25.txt 
RTCWEB Working Group C. Perkins RTCWEB Working Group C. Perkins
Internet-Draft University of Glasgow Internet-Draft University of Glasgow
Intended status: Standards Track M. Westerlund Intended status: Standards Track M. Westerlund
Expires: November 30, 2015 Ericsson Expires: December 14, 2015 Ericsson
J. Ott J. Ott
Aalto University Aalto University
May 29, 2015 June 12, 2015
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-24 draft-ietf-rtcweb-rtp-usage-25
Abstract Abstract
The Web Real-Time Communication (WebRTC) framework provides support The Web Real-Time Communication (WebRTC) framework provides support
for direct interactive rich communication using audio, video, text, for direct interactive rich communication using audio, video, text,
collaboration, games, etc. between two peers' web-browsers. This collaboration, games, etc. between two peers' web-browsers. This
memo describes the media transport aspects of the WebRTC framework. memo describes the media transport aspects of the WebRTC framework.
It specifies how the Real-time Transport Protocol (RTP) is used in It specifies how the Real-time Transport Protocol (RTP) is used in
the WebRTC context, and gives requirements for which RTP features, the WebRTC context, and gives requirements for which RTP features,
profiles, and extensions need to be supported. profiles, and extensions need to be supported.
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on November 30, 2015. This Internet-Draft will expire on December 14, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5
4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8
4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 10
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 11 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 11
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 11 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 11
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 12
4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 12 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 12
4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 13 4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 13
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 13 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 13
5.1. Conferencing Extensions and Topologies . . . . . . . . . 13 5.1. Conferencing Extensions and Topologies . . . . . . . . . 14
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 15 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 15
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 15 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 15
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 15 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 16
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 16 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 16
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 16 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 16
5.1.6. Temporary Maximum Media Stream Bit Rate Request 5.1.6. Temporary Maximum Media Stream Bit Rate Request
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 16 (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 16
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 16 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 17
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 17 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 17
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 17 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 17
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 18 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 18
5.2.4. Media Stream Identification . . . . . . . . . . . . . 18 5.2.4. Media Stream Identification . . . . . . . . . . . . . 18
5.2.5. Coordination of Video Orientation . . . . . . . . . . 18 5.2.5. Coordination of Video Orientation . . . . . . . . . . 18
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 18 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 19
6.1. Negative Acknowledgements and RTP Retransmission . . . . 19 6.1. Negative Acknowledgements and RTP Retransmission . . . . 19
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 20 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 20
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 20 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 20
7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 21 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 21
7.2. Congestion Control Interoperability and Legacy Systems . 21 7.2. Congestion Control Interoperability and Legacy Systems . 22
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 22 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 23
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 23 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 23
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 24 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 25
12. RTP Implementation Considerations . . . . . . . . . . . . . . 27 12. RTP Implementation Considerations . . . . . . . . . . . . . . 27
12.1. Configuration and Use of RTP Sessions . . . . . . . . . 27 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 27
12.1.1. Use of Multiple Media Sources Within an RTP Session 27 12.1.1. Use of Multiple Media Sources Within an RTP Session 27
12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 28 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 28
12.1.3. Differentiated Treatment of RTP Packet Streams . . . 33 12.1.3. Differentiated Treatment of RTP Packet Streams . . . 33
12.2. Media Source, RTP Packet Streams, and Participant 12.2. Media Source, RTP Packet Streams, and Participant
Identification . . . . . . . . . . . . . . . . . . . . . 35 Identification . . . . . . . . . . . . . . . . . . . . . 35
12.2.1. Media Source Identification . . . . . . . . . . . . 35 12.2.1. Media Source Identification . . . . . . . . . . . . 35
12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36 12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36
12.2.3. Media Synchronisation Context . . . . . . . . . . . 37 12.2.3. Media Synchronisation Context . . . . . . . . . . . 37
13. Security Considerations . . . . . . . . . . . . . . . . . . . 37 13. Security Considerations . . . . . . . . . . . . . . . . . . . 37
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39
16. References . . . . . . . . . . . . . . . . . . . . . . . . . 39 16. References . . . . . . . . . . . . . . . . . . . . . . . . . 39
16.1. Normative References . . . . . . . . . . . . . . . . . . 39 16.1. Normative References . . . . . . . . . . . . . . . . . . 39
16.2. Informative References . . . . . . . . . . . . . . . . . 42 16.2. Informative References . . . . . . . . . . . . . . . . . 43
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] provides a framework The Real-time Transport Protocol (RTP) [RFC3550] provides a framework
for delivery of audio and video teleconferencing data and other real- for delivery of audio and video teleconferencing data and other real-
time media applications. Previous work has defined the RTP protocol, time media applications. Previous work has defined the RTP protocol,
along with numerous profiles, payload formats, and other extensions. along with numerous profiles, payload formats, and other extensions.
When combined with appropriate signalling, these form the basis for When combined with appropriate signalling, these form the basis for
many teleconferencing systems. many teleconferencing systems.
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The RTP framework comprises the RTP data transfer protocol, the RTP The RTP framework comprises the RTP data transfer protocol, the RTP
control protocol, and numerous RTP payload formats, profiles, and control protocol, and numerous RTP payload formats, profiles, and
extensions. This range of add-ons has allowed RTP to meet various extensions. This range of add-ons has allowed RTP to meet various
needs that were not envisaged by the original protocol designers, and needs that were not envisaged by the original protocol designers, and
to support many new media encodings, but raises the question of what to support many new media encodings, but raises the question of what
extensions are to be supported by new implementations. The extensions are to be supported by new implementations. The
development of the WebRTC framework provides an opportunity 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 to 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 development of RTP 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
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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", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. The RFC document are to be interpreted as described in [RFC2119]. The RFC
2119 interpretation of these key words applies only when written in 2119 interpretation of these key words applies only when written in
ALL CAPS. Lower- or mixed-case uses of these key words are not to be ALL CAPS. Lower- or mixed-case uses of these key words are not to be
interpreted as carrying special significance in this memo. 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]. the WebRTC API [W3C.WD-mediacapture-streams-20130903]. A
MediaStream consists of zero or more MediaStreamTracks.
MediaStreamTrack: Part of the MediaStream concept defined by the W3C
in the WebRTC API [W3C.WD-mediacapture-streams-20130903]. A
MediaStreamTrack is an individual stream of media from any type of
media source like a microphone or a camera, but also conceptual
sources, like a audio mix or a video composition, are possible.
Transport-layer Flow: A uni-directional flow of transport packets Transport-layer Flow: A uni-directional flow of transport packets
that are identified by having a particular 5-tuple of source IP that are identified by having a particular 5-tuple of source IP
address, source port, destination IP address, destination port, address, source port, destination IP address, destination port,
and transport protocol used. and transport protocol used.
Bi-directional Transport-layer Flow: A bi-directional transport- Bi-directional Transport-layer Flow: A bi-directional transport-
layer flow is a transport-layer flow that is symmetric. That is, layer flow is a transport-layer flow that is symmetric. That is,
the transport-layer flow in the reverse direction has a 5-tuple the transport-layer flow in the reverse direction has a 5-tuple
where the source and destination address and ports are swapped where the source and destination address and ports are swapped
compared to the forward path transport-layer flow, and the compared to the forward path transport-layer flow, and the
transport protocol is the same. transport protocol is the same.
This document uses the terminology from This document uses the terminology from
[I-D.ietf-avtext-rtp-grouping-taxonomy] and [I-D.ietf-avtext-rtp-grouping-taxonomy] and
[I-D.ietf-rtcweb-overview]. Other terms are used according to their [I-D.ietf-rtcweb-overview]. Other terms are used according to their
definitions from the RTP Specification [RFC3550]. Especially note definitions from the RTP Specification [RFC3550]. Especially note
the following frequently used terms: RTP Packet Stream, RTP Session, the following frequently used terms: RTP Packet Stream, RTP Session,
and End-point. 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
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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 are REQUIRED in all WebRTC
Endpoints: Endpoints:
o Support for use of multiple simultaneous SSRC values in a single o Support for use of multiple simultaneous SSRC values in a single
RTP session, including support for RTP end-points that send many RTP session, including support for RTP endpoints that send many
SSRC values simultaneously, following [RFC3550] and SSRC values simultaneously, following [RFC3550] and
[I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for [I-D.ietf-avtcore-rtp-multi-stream]. The RTCP optimisations for
multi-SSRC sessions defined in multi-SSRC sessions defined in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] MAY be supported; [I-D.ietf-avtcore-rtp-multi-stream-optimisation] MAY be supported;
if supported the usage MUST be signalled. if supported the usage MUST be signalled.
o Random choice of SSRC on joining a session; collision detection o Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (see also Section 4.8). and resolution for SSRC values (see also Section 4.8).
o Support for reception of RTP data packets containing CSRC lists, o Support for reception of RTP data packets containing CSRC lists,
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extensions. extensions.
o Support for multiple synchronisation contexts. Participants that o Support for multiple synchronisation 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 synchronisation 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 synchronised 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 synchronisation contexts, indicated by the use of multiple RTCP
CNAMEs in an RTP session. This specification requires the usage CNAMEs in an RTP session. This specification mandates the usage
of a single CNAME when sending RTP Packet Streams in some of a single CNAME when sending RTP Packet Streams in some
circumstances, see Section 4.9. circumstances, see Section 4.9.
o Support for sending and receiving RTCP SR, RR, SDES, and BYE o Support for sending and receiving RTCP SR, RR, SDES, and BYE
packet types, with OPTIONAL support for other RTCP packet types packet types. Note that support for other RTCP packet types is
unless mandated by other parts of this specification. Note that OPTIONAL, unless mandated by other parts of this specification.
additional RTCP Packet types are used by the RTP/SAVPF Profile Note that additional RTCP Packet types are used by the RTP/SAVPF
(Section 4.2) and the other RTCP extensions (Section 5). WebRTC Profile (Section 4.2) and the other RTCP extensions (Section 5).
endpoints that implement the SDP bundle negotiation extension will WebRTC endpoints that implement the SDP bundle negotiation
use the SDP grouping framework 'mid' attribute to identify media extension will use the SDP grouping framework 'mid' attribute to
streams. Such endpoints MUST implement the RTCP SDES MID item identify media streams. Such endpoints MUST implement the RTCP
described in [I-D.ietf-mmusic-sdp-bundle-negotiation]. SDES MID item described in
[I-D.ietf-mmusic-sdp-bundle-negotiation].
o Support for multiple end-points in a single RTP session, and for o 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 randomised RTCP
transmission intervals to avoid synchronisation of RTCP reports; transmission intervals to avoid synchronisation 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 o 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 o 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
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o Support for discontinuous transmission. RTP allows endpoints to o 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 is not audio specific, and can be used
with any RTP payload format. with any RTP payload format.
o Ignore unknown RTCP packet types and RTP header extensions. This o Ignore unknown RTCP packet types and RTP header extensions. This
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 behaviours, etc., that can result in not signalled RTCP packet
types or RTP header extensions being received. If a compound RTCP types or RTP header extensions being received. If a compound RTCP
packet is received that contains a mixture of known and unknown packet is received that contains a mixture of known and unknown
RTCP packet types, the known packets types need to be processed as RTCP packet types, the known packets types need to be processed as
usual, with only the unknown packet types being discarded. 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 VoIP-only systems, do not support all of
the above features, and in some cases do not support RTCP at all. the above features, and in some cases do not support RTCP at all.
Implementers are advised to consider the requirements for graceful Implementers are advised to consider the requirements for graceful
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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 basic RTP/AVP profile [RFC3551], the RTP profile
for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP
profile (RTP/SAVP) [RFC3711]. profile (RTP/SAVP) [RFC3711].
The 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 end-point 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 implement
only the RTP/AVP or RTP/SAVP profile, given some constraints on only the RTP/AVP or RTP/SAVP profile, given some constraints on
parameter configuration such as the RTCP bandwidth value and "trr- parameter configuration such as the RTCP bandwidth value and "trr-
int" (the most important factor for interworking with RTP/(S)AVP int" (the most important factor for interworking with RTP/(S)AVP
end-points via a gateway is to set the trr-int parameter to a endpoints via a gateway is to set the trr-int parameter to a value
value representing 4 seconds, see Section 6.1 in representing 4 seconds, see Section 6.1 in
[I-D.ietf-avtcore-rtp-multi-stream]). [I-D.ietf-avtcore-rtp-multi-stream]).
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 and
a limited form of source authentication. WebRTC Endpoints MUST NOT a limited form of source authentication. WebRTC Endpoints MUST NOT
send packets using the basic RTP/AVP profile or the RTP/AVPF profile; send packets using the basic RTP/AVP profile or the RTP/AVPF profile;
they MUST employ the full RTP/SAVPF profile to protect all RTP and they MUST employ the full RTP/SAVPF profile to protect all RTP and
RTCP packets that are generated (i.e., implementations MUST use SRTP RTCP packets that are generated (i.e., implementations MUST use SRTP
and SRTCP). The RTP/SAVPF profile MUST be configured using the and SRTCP). The RTP/SAVPF profile MUST be configured using the
cipher suites, DTLS-SRTP protection profiles, keying mechanisms, and cipher suites, DTLS-SRTP protection profiles, keying mechanisms, and
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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.
End-points 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 signalling 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 signalled 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:
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packet streams with a signalling context, then the same RTP packet streams with a signalling context, then the same RTP
payload type number can be used to indicate the exact same RTP payload type number can be used to indicate the exact same RTP
payload format configuration in multiple contexts. payload 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 [I-D.ietf-mmusic-sdp-bundle-negotiation] form
a single RTP session). a single RTP session).
An end-point that has signalled support for multiple RTP payload An endpoint that has signalled 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 signalled 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 signalled 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 codec, RTP does not require advance
signalling for changes between RTP payload formats used by a single signalling for changes between RTP payload formats used by a single
SSRC that were signalled during session set-up. SSRC that were signalled during session set-up.
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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 case where senders use only a single clock rate). the 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 end-points communicating using RTP is An association amongst a set of endpoints communicating using RTP is
known as an RTP session [RFC3550]. An end-point 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 using a different transport-layer flow for the video). session using a different transport-layer flow for the video).
WebRTC Endpoints are REQUIRED to implement support for multimedia WebRTC Endpoints are REQUIRED to implement support for multimedia
sessions in this way, separating each RTP session using different sessions in this way, separating each RTP session using different
transport-layer flows for compatibility with legacy systems (this is transport-layer flows for compatibility with legacy systems (this is
sometimes called session multiplexing). sometimes 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
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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 behaviour. Implementations
MUST support sending and receiving non-compound RTCP feedback packets MUST support sending and receiving non-compound RTCP feedback packets
[RFC5506]. Use of non-compound RTCP packets MUST be negotiated using [RFC5506]. Use of non-compound RTCP packets MUST be negotiated using
the signalling channel. If SDP is used for signalling, this the signalling channel. If SDP is used for signalling, 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 end-point the use of compound RTCP feedback packets if the remote endpoint does
does not agree to the use of non-compound RTCP in the signalling not agree to the use of non-compound RTCP in the signalling exchange.
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 bi-directional 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 end-points, but also in the network as NAT specifically ports at the endpoints, but also in the network as NAT
mappings or firewall state is not unnecessary bloated. The amount of mappings or firewall state is not unnecessary bloated. The amount of
per flow QoS state kept in the network is also reduced. per flow QoS state kept in the network is also reduced.
4.8. Choice of RTP Synchronisation Source (SSRC) 4.8. Choice of RTP Synchronisation Source (SSRC)
Implementations are REQUIRED to support signalled RTP synchronisation Implementations are REQUIRED to support signalled RTP synchronisation
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 Section 4.1 and Section 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
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RTP packet stream are often sufficient to associate that packet RTP packet stream are often sufficient to associate that packet
stream with a signalling context (e.g., if RTP payload type numbers stream with a signalling context (e.g., if RTP payload type numbers
are assigned as described in Section 4.3 of this memo, the RTP are assigned as described in Section 4.3 of this memo, the RTP
payload types used by an RTP packet stream can be compared with payload types used by an RTP packet stream can be compared with
values in SDP "a=rtpmap:" lines, which are at the media level in SDP, values in SDP "a=rtpmap:" lines, which are at the media level in SDP,
and so map to an "m=" line). 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 end-point. While the Synchronisation Source identifier for an RTP endpoint. While the Synchronisation Source
(SSRC) identifier for an RTP end-point can change if a collision is (SSRC) identifier for an RTP endpoint can change if a collision is
detected, or when the RTP application is restarted, its RTCP CNAME is detected, or when the RTP application is restarted, its RTCP CNAME is
meant to stay unchanged for the duration of a RTCPeerConnection meant to stay unchanged for the duration of a RTCPeerConnection
[W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely [W3C.WD-webrtc-20130910], so that RTP endpoints can be uniquely
identified and associated with their RTP packet streams within a set identified and associated with their RTP packet streams within a set
of related RTP sessions. of related RTP sessions.
Each RTP end-point 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 synchronisation 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 end-point has SSRCs that are associated with several If an endpoint has SSRCs that are associated with several
unsynchronised reference clocks, and hence different synchronisation unsynchronised reference clocks, and hence different synchronisation
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. synchronisation 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 single RTCPeerConnection (that is, an RTCPeerConnection forms a
synchronisation context). RTP middleboxes MAY generate RTP packet synchronisation 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 end- avoid having to resynchronize media from multiple different endpoints
points part of a multi-party RTP session. part of a multi-party 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 as an existing
RTCPeerConnection within their common same-origin context. RTCPeerConnection within their common same-origin context.
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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
[I-D.ietf-avtcore-rtp-topologies-update]). [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 Section 5.1.1 to Section 5.1.6 are
designed to be used with centralised conferencing, where an RTP designed to be used with centralised conferencing, where an RTP
middlebox (e.g., a conference bridge) receives a participant's RTP middlebox (e.g., a conference bridge) receives a participant's RTP
packet streams and distributes them to the other participants. These packet streams and distributes them to the other participants. These
extensions are not necessary for interoperability; an RTP end-point 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 and, to provide a
reasonable baseline quality, some of these extensions are mandatory reasonable baseline quality, some of these extensions are mandatory
to be supported by WebRTC Endpoints. to be 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 the memo on Codec Control Messages (CCM) in RTP/
AVPF [RFC5104]; they are fully usable by the Secure variant of this AVPF [RFC5104]; they are fully usable by the Secure variant of this
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encoding or FEC will lead to increased play out delay, which needs to encoding or FEC will lead to increased play out delay, which needs to
be considered when choosing FEC schemes and their parameters. be 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 [I-D.ietf-rtcweb-fec]. WebRTC endpoints MAY support other types of
FEC, but these MUST be negotiated before they are used. FEC, but 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 set 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 bit-rates, load levels, and traffic
mixtures. Individual end-points 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, bit rate, 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.
End-points 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
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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 [I-D.ietf-avtcore-rtp-circuit-breakers]. The
RTP circuit breaker is designed to enable applications to recognise RTP circuit breaker is designed to enable applications to recognise
and react to situations of extreme network congestion. However, and react to situations of extreme network congestion. However,
since the RTP circuit breaker might not be triggered until congestion since the RTP circuit breaker might not be triggered until congestion
becomes extreme, it cannot be considered a substitute for congestion becomes extreme, it cannot be considered a substitute for congestion
control, and applications MUST also implement congestion control to control, and applications MUST also implement congestion control to
allow them to adapt to changes in network capacity. Any future RTP allow them to adapt to changes in network capacity. The congestion
congestion control algorithms are expected to operate within the control algorithm will have to be proprietary until a standardized
envelope allowed by the circuit breaker. congestion control algorithm is available. Any future RTP congestion
control algorithms are expected to operate within the envelope
allowed by the circuit breaker.
The session establishment signalling will also necessarily establish The session establishment signalling will also necessarily establish
boundaries to which the media bit-rate will conform. The choice of boundaries to which the media bit-rate 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 bit-
rates that the application can utilise to provide useful quality, and rates that the application can utilise to provide useful quality, and
the packetisation choices that exist. In addition, the signalling the packetisation choices that exist. In addition, the signalling
channel can establish maximum media bit-rate boundaries using, for channel can establish maximum media bit-rate 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 Temporary
Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of
this memo). Signalled bandwidth limitations, such as SDP "b=AS:" or this memo). Signalled bandwidth limitations, such as SDP "b=AS:" or
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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 end-points 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 [I-D.ietf-avtcore-rtp-circuit-breakers]
constraints to limit the congestion they can cause. constraints to limit the congestion they can cause.
If a legacy end-point supports RTP/AVPF, this enables negotiation of If a legacy 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 end-point supports some parameter, and the possibility that the endpoint supports some useful
useful feedback format for congestion control purpose such as TMMBR feedback format for congestion control purpose such as TMMBR
[RFC5104]. Implementations that have to interwork with such end- [RFC5104]. Implementations that have to interwork with such
points 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 [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the
congestion they can cause, but might find that they can achieve congestion they can cause, but might find that they can achieve
better congestion response depending on the amount of feedback that better congestion response depending on the amount of feedback that
is available. is available.
With proprietary congestion control algorithms issues can arise when 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 situation
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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 Reception 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 [RFC3611][RFC6792]. At the time RTCP Extended Reports (XR) framework, see [RFC3611][RFC6792]. At the
of this writing, it is not clear what extended metrics are suitable time of this writing, it is not clear what extended metrics are
for use in WebRTC, so there is no requirement that implementations suitable for use in WebRTC, so there is no requirement that
generate RTCP XR packets. However, implementations that can use implementations generate RTCP XR packets. However, implementations
detailed performance monitoring data MAY generate RTCP XR packets as that can use detailed performance monitoring data MAY generate RTCP
appropriate; the use of such packets SHOULD be signalled in advance. XR packets as appropriate. The use of RTCP XR packets SHOULD be
signalled; implementations MUST ignore RTCP XR packets that are
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 MUST 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 the 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 [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP
Control Protocol [RFC5968], and SHOULD take into account any future Control Protocol [RFC5968], and SHOULD take into account any future
guidelines for extending RTP and related protocols that have been guidelines for extending RTP and related protocols that have been
developed. developed.
Authors of future extensions are urged to consider the wide range of Authors of future extensions are urged to consider the wide range of
environments in which RTP is used when recommending extensions, since environments in which RTP is used when recommending extensions, since
extensions that are applicable in some scenarios can be problematic extensions that are applicable in some scenarios can be problematic
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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 signalled. This signalling is to ensure that a WebRTC Endpoint's
behaviour, especially when sending, of any extensions is behaviour, especially when sending, of any extensions is
predictable and consistent. For robustness, and for compatibility predictable and consistent. For robustness, and for compatibility
with non-WebRTC systems that might be connected to a WebRTC with non-WebRTC systems that might be connected to a WebRTC
session via a gateway, implementations are REQUIRED to ignore session via a gateway, implementations are REQUIRED to ignore
unknown RTCP packets and RTP header extensions (see also unknown RTCP packets and 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 to the
end-points will be necessary. This SHALL be done as described in endpoints will be necessary. This SHALL be done as described in
"Session Description Protocol (SDP) Bandwidth Modifiers for RTP "Session Description Protocol (SDP) Bandwidth Modifiers for RTP
Control Protocol (RTCP) Bandwidth" [RFC3556] 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
end-points have a common view of the RTCP bandwidth. A common endpoints have a common view of the RTCP bandwidth. A common view
RTCP bandwidth is important as a too different view of the of the RTCP bandwidth among different endpoints is important, to
bandwidths can lead to failure to interoperate. prevent differences in RTCP packet timing and timeout intervals
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 on the
offer/answer model, but does require all the necessary parameters to offer/answer model, but does require all the necessary parameters to
be agreed upon, and provided to the RTP implementation. Note that in be 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 signalling 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 be need to either be
set in the API or explicitly signalled between the peers. set in the API or explicitly signalled 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.WD-webrtc-20130910] and the Media Capture and
Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses
the concept of a MediaStream that consists of zero or more the concept of a MediaStream that consists of zero or more
MediaStreamTracks. A MediaStreamTrack is an individual stream of MediaStreamTracks. A MediaStreamTrack is an individual stream of
media from any type of media source like a microphone or a camera, media from any type of media source like a microphone or a camera,
but also conceptual sources, like a audio mix or a video composition, but also conceptual sources, like a audio mix or a video composition,
are possible. The MediaStreamTracks within a MediaStream need to be are possible. The MediaStreamTracks within a MediaStream might need
possible to play out synchronised. to be synchronized during play back.
A MediaStreamTrack's realisation in RTP in the context of an A MediaStreamTrack's realisation in RTP in the context of an
RTCPeerConnection consists of a source packet stream identified with RTCPeerConnection consists of a source packet stream identified with
an SSRC within an RTP session part of the RTCPeerConnection. The an SSRC within an RTP session part of the RTCPeerConnection. The
MediaStreamTrack can also result in additional packet streams, and MediaStreamTrack can also result in additional packet streams, and
thus SSRCs, in the same RTP session. These can be dependent packet thus SSRCs, in the same RTP session. These can be dependent packet
streams from scalable encoding of the source stream associated with streams from scalable encoding of the source stream associated with
the MediaStreamTrack, if such a media encoder is used. They can also the MediaStreamTrack, if such a media encoder is used. They can also
be redundancy packet streams, these are created when applying Forward be redundancy packet streams, these are created when applying Forward
Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to
skipping to change at page 25, line 28 skipping to change at page 25, line 47
choose to use only one encoded stream to create the different RTP choose to use only one encoded stream to create the different RTP
packet streams. Note that such optimisations would need to take into packet streams. Note that such optimisations would need to take into
account that the constraints for one of the MediaStreamTracks can at account that the constraints for one of the MediaStreamTracks can at
any moment change, meaning that the encoding configurations might no any moment change, meaning that the encoding configurations might no
longer be identical and two different encoder instances would then be longer 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 synchronisation base. To ensure that this works in
all cases, and does not force an end-point to 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 synchronisation 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 end-point 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 same CNAME within one RTCPeerConnection. This is motivating the use
discussion in Section 4.9 to only use a single CNAME. of a single CNAME in Section 4.9.
The requirement on using the same CNAME for all SSRCs that The requirement on using the same CNAME for all SSRCs that
originate from the same end-point, does not require a middlebox originate from the same endpoint, does not require a middlebox
that forwards traffic from multiple end-points to only use a that forwards traffic from multiple endpoints to only use a single
single CNAME. 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 [I-D.ietf-rtcweb-security] for details). A web application can
request that the CNAMEs used in different RTCPeerConnections (within request that the CNAMEs used in different RTCPeerConnections (within
a same-orign context) be the same, this allows for synchronization of a same-orign context) be the same, this allows for synchronization of
the endpoint's RTP packet streams across the different the endpoint's RTP packet streams across the different
RTCPeerConnections. RTCPeerConnections.
skipping to change at page 26, line 24 skipping to change at page 26, line 39
synchronisation. Thus, the relative relation between the timebase of synchronisation. Thus, the relative relation between the timebase of
the incoming stream and the system sending out needs to be defined. the incoming stream and the system sending out needs to be defined.
This relation also needs monitoring for clock drift and likely This relation also needs monitoring for clock drift and likely
adjustments of the synchronisation. The sending entity is also adjustments of the synchronisation. The sending entity is also
responsible for congestion control for its sent streams. In cases of responsible for congestion control for its sent streams. In cases of
packet loss the loss of incoming data also needs to be handled. This packet loss the loss of incoming data also needs to be handled. This
leads to the observation that the method that is least likely to leads to the observation that the method that is least likely to
cause issues or interruptions in the outgoing source packet stream is cause issues or interruptions in the outgoing source packet stream is
a model of full decoding, including repair etc., followed by encoding a model of full decoding, including repair etc., followed by encoding
of the media again into the outgoing packet stream. Optimisations of of the media again into the outgoing packet stream. Optimisations of
this method is clearly possible and implementation specific. 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 different MediaStreamTracks (and their sets of where each of the different MediaStreamTracks (and their 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 synchronisation.
Note: The motivation for supporting reception of multiple CNAMEs Note: The motivation for supporting reception of multiple CNAMEs
is to allow for forward compatibility with any future changes that is to allow for forward compatibility with any future changes that
enable more efficient stream handling when end-points relay/ enable more efficient stream handling when endpoints relay/forward
forward streams. It also ensures that end-points can interoperate streams. It also ensures that endpoints can interoperate with
with certain types of multi-stream middleboxes or end-points that certain types of multi-stream middleboxes or endpoints that are
are not WebRTC. not WebRTC.
The binding between the WebRTC MediaStreams, MediaStreamTracks and Javascript Session Establishment Protocol [I-D.ietf-rtcweb-jsep]
the SSRC is done as specified in "Cross Session Stream Identification specifies that the binding between the WebRTC MediaStreams,
in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This MediaStreamTracks and the SSRC is done as specified in "Cross Session
document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to Stream Identification in the Session Description Protocol"
map unknown source packet stream SSRCs to MediaStreamTracks and [I-D.ietf-mmusic-msid]. The MSID document [I-D.ietf-mmusic-msid]
MediaStreams. This later is relevant to handle some cases of legacy also defines, in section 4.1, how to map unknown source packet stream
interop. Commonly the RTP Payload Type of any incoming packets will SSRCs to MediaStreamTracks and MediaStreams. This later is relevant
reveal if the packet stream is a source stream or a redundancy or to handle some cases of legacy interoperability. Commonly the RTP
dependent packet stream. The association to the correct source Payload Type of any incoming packets will reveal if the packet stream
packet stream depends on the payload format in use for the packet is a source stream or a redundancy or dependent packet stream. The
stream. association to the correct source packet stream depends 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 and 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 behaviour 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 can include media data from multiple end-points. In the and can include media data from multiple endpoints. In the
following, some ways in which WebRTC Endpoints can configure and use following, some ways in which WebRTC Endpoints can configure and use
RTP sessions is outlined. RTP sessions are 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: Outside of WebRTC, it is common to use one RTP
session for each type of media sources (e.g., one RTP session for session for each type of media source (e.g., one RTP session for
audio sources and one for video sources, each sent over different audio sources and one for video sources, each sent over different
transport layer flows). However, to reduce the number of UDP transport layer flows). However, to reduce the number of UDP
ports used, the default in WebRTC is to send all types of media in 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 a single RTP session, as described in Section 4.4, using RTP and
RTCP multiplexing (Section 4.5) to further reduce the number of RTCP multiplexing (Section 4.5) to further reduce the number of
UDP ports needed. This RTP session then uses only one bi- UDP ports needed. This RTP session then uses only one bi-
directional transport-layer flow, but will contain multiple RTP directional transport-layer flow, but will contain multiple RTP
packet streams, each containing a different type of media. A packet streams, each containing a different type of media. A
common example might be an end-point with a camera and microphone common example might be an endpoint with a camera and microphone
that sends two RTP packet streams, one video and one audio, into a that sends two RTP packet streams, one video and one audio, into a
single RTP session. single RTP session.
Multiple Capture Devices: A WebRTC Endpoint might have multiple Multiple Capture Devices: A WebRTC Endpoint might have multiple
cameras, microphones, or other media capture devices, and so might cameras, microphones, or other media capture devices, and so might
want to generate several RTP packet streams of the same media want to generate several RTP packet streams of the same media
type. Alternatively, it might want to send media from a single type. Alternatively, it might want to send media from a single
capture device in several different formats or quality settings at capture device in several different formats or quality settings at
once. Both can result in a single end-point sending multiple RTP once. Both can result in a single endpoint sending multiple RTP
packet streams of the same media type into a single RTP session at packet streams of the same media type into a single RTP session at
the same time. the same time.
Associated Repair Data: An end-point might send a RTP packet stream Associated Repair Data: An endpoint might send a RTP packet stream
that is somehow associated with another stream. For example, it that is somehow associated with another stream. For example, it
might send an RTP packet stream that contains FEC or might send an RTP packet stream that contains FEC or
retransmission data relating to another stream. Some RTP payload retransmission data relating to another stream. Some RTP payload
formats send this sort of associated repair data as part of the formats send this sort of associated repair data as part of the
source packet stream, while others send it as a separate packet source packet stream, while others send it as a separate packet
stream. stream.
Layered or Multiple Description Coding: An end-point can use a Layered or Multiple Description Coding: An endpoint can use a
layered media codec, for example H.264 SVC, or a multiple layered media codec, for example H.264 SVC, or a multiple
description codec, that generates multiple RTP packet streams, description codec, that generates multiple RTP packet streams,
each with a distinct RTP SSRC, within a single RTP session. each with a distinct RTP SSRC, within a single RTP session.
RTP Mixers, Translators, and Other Middleboxes: An RTP session, in RTP Mixers, Translators, and Other Middleboxes: An RTP session, in
the WebRTC context, is a point-to-point association between an the WebRTC context, is a point-to-point association between an
end-point and some other peer device, where those devices share a endpoint and some other peer device, where those devices share a
common SSRC space. The peer device might be another WebRTC common SSRC space. The peer device might be another WebRTC
Endpoint, or it might be an RTP mixer, translator, or some other Endpoint, or it might be an RTP mixer, translator, or some other
form of media processing middlebox. In the latter cases, the form of media processing middlebox. In the latter cases, the
middlebox might send mixed or relayed RTP streams from several middlebox might send mixed or relayed RTP streams from several
participants, that the WebRTC Endpoint will need to render. Thus, participants, that the WebRTC Endpoint will need to render. Thus,
even though a WebRTC Endpoint might only be a member of a single even though a WebRTC Endpoint might only be a member of a single
RTP session, the peer device might be extending that RTP session RTP session, the peer device might be extending that RTP session
to incorporate other end-points. WebRTC is a group communication to incorporate other endpoints. WebRTC is a group communication
environment and end-points need to be capable of receiving, environment and endpoints need to be capable of receiving,
decoding, and playing out multiple RTP packet streams at once, decoding, and playing out multiple RTP packet streams at once,
even in a single RTP session. 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:
skipping to change at page 29, line 9 skipping to change at page 29, line 25
To provide enhanced quality of service: Some network-based quality To provide enhanced quality of service: Some network-based quality
of service mechanisms operate on the granularity of transport of service mechanisms operate on the granularity of transport
layer flows. If it is desired to use these mechanisms to provide layer flows. If it is desired to use these mechanisms to provide
differentiated quality of service for some RTP packet streams, differentiated quality of service for some RTP packet streams,
then those RTP packet streams need to be sent in a separate RTP then those RTP packet streams need to be sent in a separate RTP
session using a different transport-layer flow, and with session using a different transport-layer flow, and with
appropriate quality of service marking. This is discussed further appropriate quality of service marking. This is discussed further
in Section 12.1.3. in Section 12.1.3.
To separate media with different purposes: An end-point might want To separate media with different purposes: An endpoint might want to
to send RTP packet streams that have different purposes on send RTP packet streams that have different purposes on different
different RTP sessions, to make it easy for the peer device to RTP sessions, to make it easy for the peer device to distinguish
distinguish them. For example, some centralised multiparty them. For example, some centralised multiparty conferencing
conferencing systems display the active speaker in high systems display the active speaker in high resolution, but show
resolution, but show low resolution "thumbnails" of other low resolution "thumbnails" of other participants. Such systems
participants. Such systems might configure the end-points to send might configure the endpoints to send simulcast high- and low-
simulcast high- and low-resolution versions of their video using resolution versions of their video using separate RTP sessions, to
separate RTP sessions, to simplify the operation of the RTP simplify the operation of the RTP middlebox. In the WebRTC
middlebox. In the WebRTC context this is currently possible by context this is currently possible by establishing multiple WebRTC
establishing multiple WebRTC MediaStreamTracks that have the same MediaStreamTracks that have the same media source in one (or more)
media source in one (or more) RTCPeerConnection. Each RTCPeerConnection. Each MediaStreamTrack is then configured to
MediaStreamTrack is then configured to deliver a particular media deliver a particular media quality and thus media bit-rate, and
quality and thus media bit-rate, and will produce an independently will produce an independently encoded version with the codec
encoded version with the codec parameters agreed specifically in parameters agreed specifically in the context of that
the context of that RTCPeerConnection. The RTP middlebox can RTCPeerConnection. The RTP middlebox can distinguish packets
distinguish packets corresponding to the low- and high-resolution corresponding to the low- and high-resolution streams by
streams by inspecting their SSRC, RTP payload type, or some other inspecting their SSRC, RTP payload type, or some other information
information contained in RTP payload, RTP header extension or RTCP contained in RTP payload, RTP header extension or RTCP packets,
packets, but it can be easier to distinguish the RTP packet but it can be easier to distinguish the RTP packet streams if they
streams if they arrive on separate RTP sessions on separate arrive on separate RTP sessions on separate transport-layer flows.
transport-layer flows.
To directly connect with multiple peers: A multi-party conference To directly connect with multiple peers: A multi-party conference
does not need to use an RTP middlebox. Rather, a multi-unicast does not need to use an RTP middlebox. Rather, a multi-unicast
mesh can be created, comprising several distinct RTP sessions, mesh can be created, comprising several distinct RTP sessions,
with each participant sending RTP traffic over a separate RTP with each participant sending RTP traffic over a separate RTP
session (that is, using an independent RTCPeerConnection object) session (that is, using an independent RTCPeerConnection object)
to every other participant, as shown in Figure 1. This topology to every other participant, as shown in Figure 1. This topology
has the benefit of not requiring an RTP middlebox node that is has the benefit of not requiring an RTP middlebox node that is
trusted to access and manipulate the media data. The downside is trusted to access and manipulate the media data. The downside is
that it increases the used bandwidth at each sender by requiring that it increases the used bandwidth at each sender by requiring
skipping to change at page 30, line 21 skipping to change at page 30, line 27
v v v v
+---+ +---+
| C | | C |
+---+ +---+
Figure 1: Multi-unicast using several RTP sessions 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 between the pair of peers. To maintain a coherent mapping of the relationship
relation between RTP sessions and RTCPeerConnection objects it is between RTP sessions and RTCPeerConnection objects it is recommend
recommend that this is implemented as several individual RTP that this is implemented as several individual RTP sessions. The
sessions. The only downside is that end-point A will not learn of only downside is that endpoint A will not learn of the quality of
the quality of any transmission happening between B and C, since any transmission happening between B and C, since it will not see
it will not see RTCP reports for the RTP session between B and C, RTCP reports for the RTP session between B and C, whereas it would
whereas it would it all three participants were part of a single if all three participants were part of a single RTP session.
RTP session. Experience with the Mbone tools (experimental RTP- Experience with the Mbone tools (experimental RTP-based multicast
based multicast conferencing tools from the late 1990s) has showed conferencing tools from the late 1990s) has showed that RTCP
that RTCP reception quality reports for third parties can be reception quality reports for third parties can be presented to
presented to users in a way that helps them understand asymmetric users in a way that helps them understand asymmetric network
network problems, and the approach of using separate RTP sessions problems, and the approach of using separate RTP sessions prevents
prevents this. However, an advantage of using separate RTP this. However, an advantage of using separate RTP sessions is
sessions is that it enables using different media bit-rates and that it enables using different media bit-rates and RTP session
RTP session configurations between the different peers, thus not configurations between the different peers, thus not forcing B to
forcing B to endure the same quality reductions if there are endure the same quality reductions if there are limitations in the
limitations in the transport from A to C as C will. It is transport from A to C as C will. It is believed that these
believed that these advantages outweigh the limitations in advantages outweigh the limitations in debugging power.
debugging power.
To indirectly connect with multiple peers: A common scenario in To indirectly connect with multiple peers: A common scenario in
multi-party conferencing is to create indirect connections to multi-party conferencing is to create indirect connections to
multiple peers, using an RTP mixer, translator, or some other type multiple peers, using an RTP mixer, translator, or some other type
of RTP middlebox. Figure 2 outlines a simple topology that might of RTP middlebox. Figure 2 outlines a simple topology that might
be used in a four-person centralised conference. The middlebox be used in a four-person centralised conference. The middlebox
acts to optimise the transmission of RTP packet streams from acts to optimise the transmission of RTP packet streams from
certain perspectives, either by only sending some of the received certain perspectives, either by only sending some of the received
RTP packet stream to any given receiver, or by providing a RTP packet stream to any given receiver, or by providing a
combined RTP packet stream out of a set of contributing streams. combined RTP packet stream out of a set of contributing streams.
skipping to change at page 31, line 19 skipping to change at page 31, line 23
| or other | | or other |
+---+ | middlebox | +---+ +---+ | middlebox | +---+
| C |<---->| |<---->| D | | C |<---->| |<---->| D |
+---+ +-------------+ +---+ +---+ +-------------+ +---+
Figure 2: RTP mixer with only unicast paths 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
end-points in one multi-party session. Other types of middlebox endpoints in one multi-party session. Other types of middlebox
might use separate RTP sessions between each end-point 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 have the encoder produce a so
called Intra frame. Another is limiting the bit-rate of a given called Intra frame. Another is limiting the bit-rate of a given
stream to better suit the mixer view of the multiple down-streams. stream to better suit the mixer view of the multiple down-streams.
Others are controlling the most suitable frame-rate, picture Others are controlling the most suitable frame-rate, picture
resolution, the trade-off between frame-rate and spatial quality. resolution, the trade-off between frame-rate and spatial quality.
The middlebox has the responsibility to correctly perform The middlebox has the responsibility to correctly perform
congestion control, source identification, manage synchronisation congestion control, source identification, manage synchronisation
while providing the application with suitable media optimisations. while providing the application with suitable media optimisations.
The middlebox also has to be a trusted node when it comes to The middlebox also has to be a trusted node when it comes to
security, since it manipulates either the RTP header or the media security, since it manipulates either the RTP header or the media
itself (or both) received from one end-point, before sending it on itself (or both) received from one endpoint, before sending it on
towards the end-point(s), thus they need to be able to decrypt and towards the endpoint(s), thus they need to be able to decrypt and
then re-encrypt the RTP packet stream before sending it out. then re-encrypt the RTP packet stream before sending it out.
RTP Mixers can create a situation where an end-point experiences a RTP Mixers can create a situation where an endpoint experiences a
situation in-between a session with only two end-points and situation in-between a session with only two endpoints and
multiple RTP sessions. Mixers are expected to not forward RTCP multiple RTP sessions. Mixers are expected to not forward RTCP
reports regarding RTP packet streams across themselves. This is reports regarding RTP packet streams across themselves. This is
due to the difference in the RTP packet streams provided to the due to the difference in the RTP packet streams provided to the
different end-points. The original media source lacks information different endpoints. The original media source lacks information
about a mixer's manipulations prior to sending it the different about a mixer's manipulations prior to sending it the different
receivers. This scenario also results in that an end-point's receivers. This scenario also results in that an endpoint's
feedback or requests goes to the mixer. When the mixer can't act feedback or requests go to the mixer. When the mixer can't act on
on this by itself, it is forced to go to the original media source this by itself, it is forced to go to the original media source to
to fulfil the receivers request. This will not necessarily be fulfil the receivers request. This will not necessarily be
explicitly visible any RTP and RTCP traffic, but the interactions explicitly visible to any RTP and RTCP traffic, but the
and the time to complete them will indicate such dependencies. interactions and the time to complete them will indicate such
dependencies.
Providing source authentication in multi-party scenarios is a Providing source authentication in multi-party scenarios is a
challenge. In the mixer-based topologies, end-points 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 end- in the mixer to correctly identify any source towards the
point. In RTP sessions where multiple end-points are directly endpoint. In RTP sessions where multiple endpoints are directly
visible to an end-point, all end-points 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 claimed to
come from another end-point 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 non-cryptographic 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
end-points 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 TESLA for SRTP [RFC4383], that are not part of the base WebRTC
standards. standards.
To forward media between multiple peers: It is sometimes desirable To forward media between multiple peers: It is sometimes desirable
for an end-point that receives an RTP packet stream to be able to 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 forward that RTP packet stream to a third party. The are some
obvious security and privacy implications in supporting this, but obvious security and privacy implications in supporting this, but
also potential uses. This is supported in the W3C API by taking also potential uses. This is supported in the W3C API by taking
the received and decoded media and using it as media source that the received and decoded media and using it as media source that
is re-encoding and transmitted as a new stream. is re-encoding 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 end-point 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 to 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 purpose to produce a suitable bit-rate for the outgoing
RTP session, reducing media quality and forcing the forwarding RTP session, reducing media quality and forcing the forwarding
end-point 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 end- removing almost all dependencies, and allowing the forwarding
point to optimise its media transcoding operation. The cost is endpoint to optimise 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 Packet Streams 12.1.3. Differentiated Treatment of RTP Packet 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 end-point 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 that
will be sent, and their allocation of bit-rate out of the current will be sent, and their allocation of bit-rate out of the current
available aggregate as determined by the congestion control. available aggregate as determined by the congestion control.
It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will
allow the application to indicate relative priorities for different allow the 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 congestion
control response in how to divide the available bandwidth between the control response in how to divide the available bandwidth between the
RTP packet streams. Any changes in relative priority will also need RTP packet streams. Any changes in relative priority will also need
skipping to change at page 33, line 38 skipping to change at page 33, line 45
might to be to use the same priority for redundant RTP packet stream might to be to use the same priority for redundant RTP packet stream
as for the source 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 sub-
flows, including RTP packet streams. Typically, differential flows, 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 to prioritize packets.
Three common methods for classifying IP packets are: Three common methods for classifying IP packets are:
DiffServ: The end-point 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.
skipping to change at page 34, line 15 skipping to change at page 34, line 23
for all the RTP packet streams used in a WebRTC session. The use of for all the RTP packet streams used in a WebRTC session. The use of
flow-based differentiation needs to be coordinated between the WebRTC flow-based differentiation needs to be coordinated between the WebRTC
system and the network(s). The WebRTC endpoint needs to know that system and the network(s). The WebRTC endpoint needs to know that
flow-based differentiation might be used to provide the separation of flow-based differentiation might be used to provide the separation of
the RTP packet streams onto different UDP flows to enable a more the RTP packet streams onto different UDP flows to enable a more
granular usage of flow based differentiation. The used flows, their granular usage of flow based differentiation. The used flows, their
5-tuples and prioritization will need to be communicated to the 5-tuples and prioritization will need to be communicated to the
network so that it can identify the flows correctly to enable network so that it can identify the flows correctly to enable
prioritization. No specific protocol support for this is specified. prioritization. No specific protocol support for this is specified.
DiffServ assumes that either the end-point 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 DSCP so that the packets are treated
according to that marking. If the end-point is to mark the traffic according to that marking. If the endpoint is to mark the traffic
two requirements arise in the WebRTC context: 1) The WebRTC Endpoint two requirements arise in the WebRTC context: 1) The WebRTC Endpoint
has to know which DSCP to use and that it can use them on some set of has to know which DSCP to use and that it can use them on some set of
RTP packet streams. 2) The information needs to be propagated to the RTP packet streams. 2) The information needs to be propagated to the
operating system when transmitting the packet. Details of this operating system when transmitting the packet. Details of this
process are outside the scope of this memo and are further discussed process are outside the scope of this memo and are further discussed
in "DSCP and other packet markings for RTCWeb QoS" in "DSCP and other packet markings for RTCWeb QoS"
[I-D.ietf-tsvwg-rtcweb-qos]. [I-D.ietf-tsvwg-rtcweb-qos].
Deep Packet Inspectors will, despite the SRTP media encryption, still Deep Packet Inspectors will, despite the SRTP media encryption, still
be fairly capable at classifying the RTP streams. The reason is that be fairly capable at classifying the RTP streams. The reason is that
skipping to change at page 34, line 42 skipping to change at page 34, line 50
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 prioritise 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 loose. However, depending on the QoS
mechanism and what markings that are applied, this can result in not mechanism and what markings that are applied, this can result in not
only different packet drop probabilities but also packet reordering, only different packet drop probabilities but also packet reordering,
see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As a default see [I-D.ietf-tsvwg-rtcweb-qos] and [I-D.ietf-dart-dscp-rtp] for
policy all RTP packets related to a RTP packet stream ought to be further discussion. As a default policy all RTP packets related to a
provided with the same prioritization; per-packet prioritization is RTP packet stream ought to be provided with the same prioritization;
outside the scope of this memo, but might be specified elsewhere in per-packet prioritization is outside the scope of this memo, but
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 (SR), 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 RR packet ought to be sent with the priority used
by the majority of the RTP packet streams reported on. RTCP packets by the majority of the RTP packet streams reported on. RTCP packets
containing time-critical feedback packets can use higher priority to containing time-critical feedback packets can use higher priority to
skipping to change at page 36, line 9 skipping to change at page 36, line 13
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., resolve the conflict
when two different end-points use the same SSRC value (see section when two different endpoints use the same SSRC value (see section 8.2
8.2 of [RFC3550]). This requirement also applies to WebRTC of [RFC3550]). This requirement also applies to WebRTC Endpoints.
Endpoints. There are several scenarios where SSRC collisions can There are several scenarios where SSRC collisions can occur:
occur:
o In a point-to-point session where each SSRC is associated with o In a point-to-point session where each SSRC is associated with
either of the two end-points and where the main media carrying either of the two endpoints and where the main media carrying SSRC
SSRC identifier will be announced in the signalling channel, a identifier will be announced in the signalling channel, a
collision is less likely to occur due to the information about collision is less likely to occur due to the information about
used SSRCs. If SDP is used, this information is provided by used SSRCs. If SDP is used, this information is provided by
Source-Specific SDP Attributes [RFC5576]. Still, collisions can Source-Specific SDP Attributes [RFC5576]. Still, collisions can
occur if both end-points start using a new SSRC identifier prior occur if both endpoints start using a new SSRC identifier prior to
to having signalled it to the peer and received acknowledgement on having signalled it to the peer and received acknowledgement on
the signalling message. The Source-Specific SDP Attributes the signalling message. The Source-Specific SDP Attributes
[RFC5576] contains a mechanism to signal how the end-point [RFC5576] contains a mechanism to signal how the endpoint resolved
resolved the SSRC collision. the SSRC collision.
o SSRC values that have not been signalled could also appear in an o SSRC values that have not been signalled 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 to the SSRC
of the source RTP packet stream [RFC4588]. In those cases, an of the source RTP packet stream [RFC4588]. In those cases, an
end-point can create a new SSRC that strictly doesn't need to be 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 signalling channel to function correctly on
both RTP and RTCPeerConnection level. both RTP and RTCPeerConnection level.
o Multiple end-points in a multiparty conference can create new o 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 end- where the SSRC/CSRC are propagated between the different endpoints
points from the RTP middlebox collisions can occur. from the RTP middlebox collisions can occur.
o An RTP middlebox could connect an end-point's RTCPeerConnection to o An RTP middlebox could connect an endpoint's RTCPeerConnection to
another RTCPeerConnection from the same end-point, thus forming a another RTCPeerConnection from the same endpoint, thus forming a
loop where the end-point will receive its own traffic. While it loop where the endpoint will receive its own traffic. While it is
is clearly considered a bug, it is important that the end-point is clearly considered a bug, it is important that the endpoint is
able to recognise and handle the case when it occurs. This case able to recognise 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 so on, 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 RTP level as long
as the same RTP session is extended across multiple as the same RTP session is extended across multiple
RTCPeerConnections by a RTP middlebox. To resolve the more generic RTCPeerConnections by a RTP middlebox. To resolve the more generic
case where multiple RTCPeerConnections are interconnected, case where multiple RTCPeerConnections are interconnected,
identification of the media source(s) part of a MediaStreamTrack identification of the media source(s) part of a MediaStreamTrack
being propagated across multiple interconnected RTCPeerConnection being propagated across multiple interconnected RTCPeerConnection
needs to be preserved across these interconnections. needs to be preserved across these interconnections.
12.2.3. Media Synchronisation Context 12.2.3. Media Synchronisation Context
When an end-point 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, synchronisation 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 end-point by using the same RTCP synchronisation 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 system
clock that is provided in RTCP Sender Reports encoded in an NTP clock that is provided in RTCP Sender Reports encoded in an NTP
format. By correlating all RTP timestamps to a common system clock format. By correlating all RTP timestamps to a common system clock
for all sources, the timing relation of the different RTP packet for all sources, the timing relation of the different RTP packet
streams, also across multiple RTP sessions can be derived at the streams, also across multiple RTP sessions can be derived at the
receiver and, if desired, the streams can be synchronized. The receiver and, if desired, the streams can be synchronized. The
requirement is for the media sender to provide the correlation requirement is for the media sender to provide the correlation
skipping to change at page 37, line 44 skipping to change at page 37, line 48
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 not believed there are any 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 The Extended Secure RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
handling of fundamental issues by offering confidentiality, integrity handling of fundamental issues by offering confidentiality, integrity
and partial source authentication. A mandatory to implement media and partial source authentication. A mandatory to implement and use
security solution is created by combing this secured RTP profile and media security solution is created by combining this secured RTP
DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of profile and DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of
[I-D.ietf-rtcweb-security-arch]. [I-D.ietf-rtcweb-security-arch].
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 synchronised 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 provides guidelines for generation of untraceable CNAME this memo mandates generation of short-term persistent RTCP CNAMES,
values that alleviate this risk. as specified in RFC7022, resulting in untraceable CNAME values that
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:
skipping to change at page 39, line 15 skipping to change at page 39, line 19
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 third
party monitoring that will greatly benefit from the information, and party monitoring that will greatly benefit from the information, and
this has been expressed using API or signalling. If further evidence this has been expressed using API or signalling. If further evidence
are produced to show that information leakage is significant from are produced to show that information leakage is significant from
audio level indications, then use of encryption needs to be mandated audio level indications, then use of encryption needs to be mandated
at that time. at that time.
In multi-party communication scenarios using RTP Middleboxes, a lot
of trust is placed on these middleboxes to preserve the sessions
security. The middlebox needs to maintain the confidentiality,
integrity and perform source authentication. As discussed in
Section 12.1.1 the middlebox can perform checks that prevents any
endpoint participating in a conference to impersonate another. Some
additional security considerations regarding multi-party topologies
can be found in [I-D.ietf-avtcore-rtp-topologies-update].
14. IANA Considerations 14. IANA Considerations
This memo makes no request of IANA. This memo makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as Note to RFC Editor: this section is to be removed on publication as
an RFC. an RFC.
15. Acknowledgements 15. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand, The authors would like to thank Bernard Aboba, Harald Alvestrand,
Cary Bran, Ben Campbell, Alissa Cooper, Charles Eckel, Alex Cary Bran, Ben Campbell, Alissa Cooper, Spencer Dawkins, Charles
Eleftheriadis, Christian Groves, Cullen Jennings, Olle Johansson, Eckel, Alex Eleftheriadis, Christian Groves, Chris Inacio, Cullen
Suhas Nandakumar, Dan Romascanu, Jim Spring, Martin Thomson, and the Jennings, Olle Johansson, Suhas Nandakumar, Dan Romascanu, Jim
other members of the IETF RTCWEB working group for their valuable Spring, Martin Thomson, and the other members of the IETF RTCWEB
feedback. working group for their valuable feedback.
16. References 16. References
16.1. Normative References 16.1. Normative References
[I-D.ietf-avtcore-multi-media-rtp-session] [I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Sending Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft- Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-07 (work in ietf-avtcore-multi-media-rtp-session-07 (work in
progress), March 2015. progress), March 2015.
skipping to change at page 40, line 12 skipping to change at page 40, line 24
draft-ietf-avtcore-rtp-multi-stream-07 (work in progress), draft-ietf-avtcore-rtp-multi-stream-07 (work in progress),
March 2015. March 2015.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins, Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session: "Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback", Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-05 (work draft-ietf-avtcore-rtp-multi-stream-optimisation-05 (work
in progress), February 2015. in progress), February 2015.
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-07 (work in progress),
April 2015.
[I-D.ietf-mmusic-sdp-bundle-negotiation] [I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings, Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session "Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-19 (work in progress), March 2015. negotiation-19 (work in progress), March 2015.
[I-D.ietf-rtcweb-audio] [I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-08 (work in Requirements", draft-ietf-rtcweb-audio-08 (work in
progress), April 2015. progress), April 2015.
[I-D.ietf-rtcweb-fec] [I-D.ietf-rtcweb-fec]
Uberti, J., "WebRTC Forward Error Correction Uberti, J., "WebRTC Forward Error Correction
Requirements", draft-ietf-rtcweb-fec-01 (work in Requirements", draft-ietf-rtcweb-fec-01 (work in
progress), March 2015. progress), March 2015.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-13
(work in progress), November 2014.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-08 (work in progress), February 2015. ietf-rtcweb-security-08 (work in progress), February 2015.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-11 (work in progress), March 2015. rtcweb-security-arch-11 (work in progress), March 2015.
[I-D.ietf-rtcweb-video] [I-D.ietf-rtcweb-video]
Roach, A., "WebRTC Video Processing and Codec Roach, A., "WebRTC Video Processing and Codec
skipping to change at page 42, line 39 skipping to change at page 43, line 15
[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, September 2013. Canonical Names (CNAMEs)", RFC 7022, September 2013.
[RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple [RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", RFC 7160, April 2014. Clock Rates in an RTP Session", RFC 7160, April 2014.
[RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC [RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC
7164, March 2014. 7164, March 2014.
[W3C.WD-mediacapture-streams-20130903]
Burnett, D., Bergkvist, A., Jennings, C., and A.
Narayanan, "Media Capture and Streams", World Wide Web
Consortium WD WD-mediacapture-streams-20130903, September
2013, <http://www.w3.org/TR/2013/
WD-mediacapture-streams-20130903>.
[W3C.WD-webrtc-20130910]
Bergkvist, A., Burnett, D., Jennings, C., and A.
Narayanan, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc-
20130910, September 2013,
<http://www.w3.org/TR/2013/WD-webrtc-20130910>.
16.2. Informative References 16.2. Informative References
[I-D.ietf-avtcore-multiplex-guidelines] [I-D.ietf-avtcore-multiplex-guidelines]
Westerlund, M., Perkins, C., and H. Alvestrand, Westerlund, M., Perkins, C., and H. Alvestrand,
"Guidelines for using the Multiplexing Features of RTP to "Guidelines for using the Multiplexing Features of RTP to
Support Multiple Media Streams", draft-ietf-avtcore- Support Multiple Media Streams", draft-ietf-avtcore-
multiplex-guidelines-03 (work in progress), October 2014. multiplex-guidelines-03 (work in progress), October 2014.
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-07 (work in progress),
April 2015.
[I-D.ietf-avtext-rtp-grouping-taxonomy] [I-D.ietf-avtext-rtp-grouping-taxonomy]
Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, "A Taxonomy of Grouping Semantics and B. Burman, "A Taxonomy of Grouping Semantics and
Mechanisms for Real-Time Transport Protocol (RTP) Mechanisms for Real-Time Transport Protocol (RTP)
Sources", draft-ietf-avtext-rtp-grouping-taxonomy-06 (work Sources", draft-ietf-avtext-rtp-grouping-taxonomy-06 (work
in progress), March 2015. in progress), March 2015.
[I-D.ietf-dart-dscp-rtp]
Black, D. and P. Jones, "Differentiated Services
(DiffServ) and Real-time Communication", draft-ietf-dart-
dscp-rtp-10 (work in progress), November 2014.
[I-D.ietf-mmusic-msid] [I-D.ietf-mmusic-msid]
Alvestrand, H., "WebRTC MediaStream Identification in the Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", draft-ietf-mmusic-msid-10 Session Description Protocol", draft-ietf-mmusic-msid-10
(work in progress), April 2015. (work in progress), April 2015.
[I-D.ietf-payload-rtp-howto] [I-D.ietf-payload-rtp-howto]
Westerlund, M., "How to Write an RTP Payload Format", Westerlund, M., "How to Write an RTP Payload Format",
draft-ietf-payload-rtp-howto-14 (work in progress), May draft-ietf-payload-rtp-howto-14 (work in progress), May
2015. 2015.
[I-D.ietf-rmcat-cc-requirements] [I-D.ietf-rmcat-cc-requirements]
Jesup, R. and Z. Sarker, "Congestion Control Requirements Jesup, R. and Z. Sarker, "Congestion Control Requirements
for Interactive Real-Time Media", draft-ietf-rmcat-cc- for Interactive Real-Time Media", draft-ietf-rmcat-cc-
requirements-09 (work in progress), December 2014. requirements-09 (work in progress), December 2014.
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-jsep]
Alvestrand, H., "Overview: Real Time Protocols for Uberti, J., Jennings, C., and E. Rescorla, "Javascript
Browser-based Applications", draft-ietf-rtcweb-overview-13 Session Establishment Protocol", draft-ietf-rtcweb-jsep-09
(work in progress), November 2014. (work in progress), March 2015.
[I-D.ietf-tsvwg-rtcweb-qos] [I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J. Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J.
Polk, "DSCP and other packet markings for RTCWeb QoS", Polk, "DSCP and other packet markings for RTCWeb QoS",
draft-ietf-tsvwg-rtcweb-qos-03 (work in progress), draft-ietf-tsvwg-rtcweb-qos-03 (work in progress),
November 2014. November 2014.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003. 2003.
skipping to change at page 44, line 23 skipping to change at page 45, line 9
Keeping Alive the NAT Mappings Associated with RTP / RTP Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, June 2011. Control Protocol (RTCP) Flows", RFC 6263, June 2011.
[RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the [RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the
RTP Monitoring Framework", RFC 6792, November 2012. RTP Monitoring Framework", RFC 6792, November 2012.
[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,
March 2015. March 2015.
[W3C.WD-mediacapture-streams-20130903]
Burnett, D., Bergkvist, A., Jennings, C., and A.
Narayanan, "Media Capture and Streams", World Wide Web
Consortium WD WD-mediacapture-streams-20130903, September
2013, <http://www.w3.org/TR/2013/
WD-mediacapture-streams-20130903>.
[W3C.WD-webrtc-20130910]
Bergkvist, A., Burnett, D., Jennings, C., and A.
Narayanan, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc-
20130910, September 2013,
<http://www.w3.org/TR/2013/WD-webrtc-20130910>.
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: http://csperkins.org/ URI: https://csperkins.org/
Magnus Westerlund Magnus Westerlund
Ericsson Ericsson
Farogatan 6 Farogatan 6
SE-164 80 Kista SE-164 80 Kista
Sweden Sweden
Phone: +46 10 714 82 87 Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com Email: magnus.westerlund@ericsson.com
Joerg Ott Joerg Ott
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