draft-ietf-rtcweb-rtp-usage-02.txt   draft-ietf-rtcweb-rtp-usage-03.txt 
Network Working Group C. Perkins Network Working Group C. Perkins
Internet-Draft University of Glasgow Internet-Draft University of Glasgow
Intended status: Standards Track M. Westerlund Intended status: Standards Track M. Westerlund
Expires: September 13, 2012 Ericsson Expires: December 6, 2012 Ericsson
J. Ott J. Ott
Aalto University Aalto University
March 12, 2012 June 4, 2012
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-02 draft-ietf-rtcweb-rtp-usage-03
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.
skipping to change at page 1, line 39 skipping to change at page 1, line 39
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 September 13, 2012. This Internet-Draft will expire on December 6, 2012.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Discussion and Rationale . . . . . . . . . . . . . . . . . . . 4 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Supported RTP Topologies . . . . . . . . . . . . . . . . . 4 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 6
3.2. Signalling Requirements . . . . . . . . . . . . . . . . . 7 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . . 6
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 8 4.2. Choice of RTP Profile . . . . . . . . . . . . . . . . . . 7
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . . 8 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7
4.2. Choice of RTP Profile . . . . . . . . . . . . . . . . . . 8 4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 8
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 9 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 8
4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 10 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 9
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 9
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 4.8. Generation of the RTCP Canonical Name (CNAME) . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 11 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 10
4.8. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11 5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 10
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12 5.1.1. Full Intra Request . . . . . . . . . . . . . . . . . . 11
5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12 5.1.2. Picture Loss Indication . . . . . . . . . . . . . . . 11
5.1.1. Full Intra Request . . . . . . . . . . . . . . . . . . 13 5.1.3. Slice Loss Indication . . . . . . . . . . . . . . . . 11
5.1.2. Picture Loss Indication . . . . . . . . . . . . . . . 13 5.1.4. Reference Picture Selection Indication . . . . . . . . 12
5.1.3. Slice Loss Indication . . . . . . . . . . . . . . . . 13 5.1.5. Temporary Maximum Media Stream Bit Rate Request . . . 12
5.1.4. Reference Picture Selection Indication . . . . . . . . 13 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 12
5.1.5. Temporary Maximum Media Stream Bit Rate Request . . . 13 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 12
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14 5.2.2. Client to Mixer Audio Level . . . . . . . . . . . . . 13
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 14 5.2.3. Mixer to Client Audio Level . . . . . . . . . . . . . 13
5.2.2. Client to Mixer Audio Level . . . . . . . . . . . . . 14 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 13
5.2.3. Mixer to Client Audio Level . . . . . . . . . . . . . 15 6.1. Retransmission . . . . . . . . . . . . . . . . . . . . . . 14
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 15 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 15
6.1. Retransmission . . . . . . . . . . . . . . . . . . . . . . 15 6.2.1. Basic Redundancy . . . . . . . . . . . . . . . . . . . 15
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 16 6.2.2. Block Based FEC . . . . . . . . . . . . . . . . . . . 16
6.2.1. Basic Redundancy . . . . . . . . . . . . . . . . . . . 17 6.2.3. Recommendations for FEC . . . . . . . . . . . . . . . 17
6.2.2. Block Based FEC . . . . . . . . . . . . . . . . . . . 18 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 17
6.2.3. Recommendations for FEC . . . . . . . . . . . . . . . 19 7.1. Congestion Control Requirements . . . . . . . . . . . . . 19
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 19 7.2. Rate Control Boundary Conditions . . . . . . . . . . . . . 19
7.1. Congestion Control Requirements . . . . . . . . . . . . . 20 7.3. RTCP Limiations . . . . . . . . . . . . . . . . . . . . . 19
7.2. RTCP Limiations . . . . . . . . . . . . . . . . . . . . . 20 7.4. Legacy Interop Limitations . . . . . . . . . . . . . . . . 20
7.3. Legacy Interop Limitations . . . . . . . . . . . . . . . . 21 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 21
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 21
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21
10. Security Considerations . . . . . . . . . . . . . . . . . . . 22 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 23
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23 11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 23
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23 12. RTP Implementation Considerations . . . . . . . . . . . . . . 23
12.1. Normative References . . . . . . . . . . . . . . . . . . . 23 12.1. RTP Sessions and PeerConnection . . . . . . . . . . . . . 24
12.2. Informative References . . . . . . . . . . . . . . . . . . 26 12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 25
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27 12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 25
12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 27
12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 28
12.6. Media Synchronization . . . . . . . . . . . . . . . . . . 29
12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . . 29
12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 30
12.9. Differentiated Treatment of Flows . . . . . . . . . . . . 30
13. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31
14. Security Considerations . . . . . . . . . . . . . . . . . . . 32
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 32
16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32
16.1. Normative References . . . . . . . . . . . . . . . . . . . 32
16.2. Informative References . . . . . . . . . . . . . . . . . . 35
Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 37
A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 37
A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 40
A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 43
A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 43
A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 46
A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . . 49
A.4. Translator Based . . . . . . . . . . . . . . . . . . . . . 52
A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . . 52
A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 53
A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 55
A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . . 59
A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 60
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61
1. Introduction 1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] provides a framework The Real-time Transport Protocol (RTP) [RFC3550] provides a framework
for delivery of audio and video teleconferencing data and other real- for delivery of audio and video teleconferencing data and other real-
time media applications. Previous work has defined the RTP protocol, time media applications. Previous work has defined the RTP protocol,
plus numerous profiles, payload formats, and other extensions. When along with numerous profiles, payload formats, and other extensions.
combined with appropriate signalling, these form the basis for many When combined with appropriate signalling, these form the basis for
teleconferencing systems. many teleconferencing systems.
The Web Real-Time communication (WebRTC) framework is a new activity The Web Real-Time communication (WebRTC) framework is a new protocol
that provides support for direct, interactive, and rich communication framework that provides support for direct, interactive, real-time
using audio, video, collaboration, games, etc., between two peers' communication using audio, video, collaboration, games, etc., between
web-browsers. This memo describes how the RTP framework is used in two peers' web-browsers. This memo describes how the RTP framework
the WebRTC context. It proposes a baseline set of RTP features that is to be used in the WebRTC context. It proposes a baseline set of
should be implemented by all WebRTC-aware browsers, along with some RTP features that must be implemented by all WebRTC-aware browsers,
suggested extensions for enhanced functionality. along with suggested extensions for enhanced functionality.
The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete
WebRTC framework, of which this memo is a part. WebRTC framework, of which this memo is a part.
2. Terminology The structure of this memo is as follows. Section 2 outlines our
rationale in preparing this memo and choosing these RTP features.
This memo is structured into different topics. For each topic, one Section 3 defines requirement terminology. Requirements for core RTP
or several recommendations from the authors are given. When it comes protocols are described in Section 4 and recommended RTP extensions
to the importance of extensions, or the need for implementation are described in Section 5. Section 6 outlines mechanisms that can
support, we use three requirement levels to indicate the importance increase robustness to network problems, while Section 7 describes
of the feature to the WebRTC specification: the required congestion control and rate adaptation mechanisms. The
discussion of required RTP mechanisms concludes in Section 8 with a
REQUIRED: Functionality that is absolutely needed to make the WebRTC review of performance monitoring and network management tools that
solution work well, or functionality of low complexity that can be used in the WebRTC context. Section 9 gives some guidelines
provides high value. for future incorporation of other RTP and RTP Control Protocol (RTCP)
extensions into this framework. Section 10 describes requirements
RECOMMENDED: Should be included as its brings significant benefit, placed on the signalling channel. Section 11 discusses the
but the solution can potentially work without it. relationship between features of the RTP framework and the WebRTC
application programming interface (API), and Section 12 discusses RTP
OPTIONAL: Something that is useful in some cases, but not always a implementation considerations. This memo concludes with an appendix
benefit. discussing several different RTP Topologies, and how they affect the
RTP session(s) and various implementation details of possible
3. Discussion and Rationale realization of central nodes.
3.1. Supported RTP Topologies
RTP supports both unicast and group communication, with participants
being connected using wide range of transport-layer topologies. Some
of these topologies involve only the end-points, while others use RTP
translators and mixers to provide in-network processing. Properties
of some RTP topologies are discussed in [RFC5117], and we further
describe those expected to be useful for WebRTC in the following.
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 1: Point to Point
The point to point topology (Figure 1) is going to be very common in
any single user to single user applications.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 2: Multi-unicast
For small multiparty sessions it is practical enough to create RTP
sessions by letting every participant send individual unicast RTP/UDP
flows to each of the other participants. This is called multi-
unicast (Figure 2), and is unfortunately not discussed in the RTP
Topologies [RFC5117]. This topology has the benefit of not requiring
central nodes. The downside is that it increases the used bandwidth
at each sender by requiring one copy of the media streams for each
participant that are part of the same session beyond the sender
itself. Thus this is limited to scenarios with few end-points unless
the media is very low bandwidth.
This topology may be implemented as a single RTP session, spanning 2. Rationale
multiple peer to peer transport layer connections, or as several
pairwise RTP sessions, one between each pair of peers. The later
approach simplifies rate adaptation, but reduces the effectiveness of
RTCP for debugging purposes, by limiting the ability to send third-
party RTCP reports.
+---+ +------------+ +---+ The RTP framework comprises the RTP data transfer protocol, the RTP
| A |<---->| |<---->| B | control protocol, and numerous RTP payload formats, profiles, and
+---+ | | +---+ extensions. This range of add-ons has allowed RTP to meet various
| Mixer | needs that were not envisaged by the original protocol designers, and
+---+ | | +---+ to support many new media encodings, but raises the question of what
| C |<---->| |<---->| D | features should be supported by new implementations? The development
+---+ +------------+ +---+ of the WebRTC framework provides an opportunity for us to review the
available RTP features and extensions, and to define a common
baseline feature set for all WebRTC implementations of RTP. This
builds on the past 15 years development of RTP to mandate the use of
extensions that have shown widespread utility, while still remaining
compatible with the wide installed base of RTP implementations where
possible.
Figure 3: RTP Mixer with Only Unicast Paths While the baseline set of RTP features and extensions defined in this
memo is targetted at the requirements of the WebRTC framework, it is
expected to be broadly useful for other conferencing-related uses of
RTP. In particular, it is likely that this set of RTP features and
extensions will be apppropriate for other desktop or mobile video
conferencing systems, or for room-based high-quality telepresence
applications.
An RTP mixer (Figure 3) is a centralised point that selects or mixes 3. Terminology
content in a conference to optimise the RTP session so that each end-
point only needs connect to one entity, the mixer. The mixer also
reduces the bit-rate needs as the media sent from the mixer to the
end-point can be optimised in different ways. These optimisations
include methods like only choosing media from the currently most
active speaker or mixing together audio so that only one audio stream
is required in stead of 3 in the depicted scenario. The downside of
the mixer is that someone is required to provide the actual mixer.
+---+ +------------+ +---+ This memo specifies various requirements levels for implementation or
| A |<---->| |<---->| B | use of RTP features and extensions. When we describe the importance
+---+ | | +---+ of RTP extensions, or the need for implementation support, we use the
| Translator | following requirement levels to specify the importance of the feature
+---+ | | +---+ in the WebRTC framework:
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 4: RTP Translator (Relay) with Only Unicast Paths MUST: This word, or the terms "REQUIRED" or "SHALL", mean that the
definition is an absolute requirement of the specification.
If one wants a less complex central node it is possible to use an SHOULD: This word, or the adjective "RECOMMENDED", mean that there
relay (called an Transport Translator) (Figure 4) that takes on the may exist valid reasons in particular circumstances to ignore a
role of forwarding the media to the other end-points but doesn't particular item, but the full implications must be understood and
perform any media processing. It simply forwards the media from all carefully weighed before choosing a different course.
other to all the other. Thus one endpoint A will only need to send a
media once to the relay, but it will still receive 3 RTP streams with
the media if B, C and D all currently transmits.
+------------+ MAY: This word, or the adjective "OPTIONAL", mean that an item is
| | truly optional. One vendor may choose to include the item because
+---+ | | +---+ a particular marketplace requires it or because the vendor feels
| A |<---->| Translator |<---->| B | that it enhances the product while another vendor may omit the
+---+ | | +---+ same item. An implementation which does not include a particular
| | option MUST be prepared to interoperate with another
+------------+ implementation which does include the option, though perhaps with
reduced functionality. In the same vein an implementation which
does include a particular option MUST be prepared to interoperate
with another implementation which does not include the option
(except, of course, for the feature the option provides.)
Figure 5: Translator towards Legacy end-point These key words are used in a manner consistent with their definition
To support legacy end-point (B) that don't fulfil the requirements of in [RFC2119].
WebRTC it is possible to insert a Translator (Figure 5) that takes on
the role to ensure that from A's perspective B looks like a fully
compliant end-point. Thus it is the combination of the Translator
and B that looks like the end-point B. The intention is that the
presence of the translator is transparent to A, however it is not
certain that is possible. Thus this case is include so that it can
be discussed if any mechanism specified to be used for WebRTC results
in such issues and how to handle them.
3.2. Signalling Requirements 4. WebRTC Use of RTP: Core Protocols
RTP is built with the assumption of an external signalling channel The following sections describe the core features of RTP and RTCP
that can be used to configure the RTP sessions and their features. that MUST be implemented, along with the mandated RTP profiles and
The basic configuration of an RTP session consists of the following payload formats. Also described are the core extensions providing
parameters: essential features that all WebRTC implementations MUST implement to
function effectively on today's networks.
RTP Profile: The name of the RTP profile to be used in session. The 4.1. RTP and RTCP
RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
on basic level, as can their secure variants RTP/SAVP [RFC3711]
and RTP/SAVPF [RFC5124]. The secure variants of the profiles do
not directly interoperate with the non-secure variants, due to the
presence of additional header fields in addition to any
cryptographic transformation of the packet content.
Transport Information: Source and destination address(s) and ports The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be
for RTP and RTCP MUST be signalled for each RTP session. If RTP implemented as the media transport protocol for WebRTC. RTP itself
and RTCP multiplexing [RFC5761] is to be used, such that a single comprises two parts: the RTP data transfer protocol, and the RTP
port is used for RTP and RTCP flows, this MUST be signalled (see control protocol (RTCP). RTCP is a fundamental and integral part of
Section 4.5). If several RTP sessions are to be multiplexed onto RTP, and MUST be implemented in all WebRTC applications.
a single transport layer flow, this MUST also be signalled (see
Section 4.4).
RTP Payload Types, media formats, and media format The following RTP and RTCP features are sometimes omitted in limited
parameters: The mapping between media type names (and hence the RTP functionality implementations of RTP, but are REQUIRED in all WebRTC
payload formats to be used) and the RTP payload type numbers must implementations:
be signalled. Each media type may also have a number of media
type parameters that must also be signalled to configure the codec
and RTP payload format (the "a=fmtp:" line from SDP).
RTP Extensions: The RTP extensions one intends to use need to be o Support for use of multiple simultaneous SSRC values in a single
agreed upon, including any parameters for each respective RTP session, including support for RTP end-points that send many
extension. At the very least, this will help avoiding using SSRC values simultaneously.
bandwidth for features that the other end-point will ignore. But
for certain mechanisms there is requirement for this to happen as
interoperability failure otherwise happens.
RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the o Random choice of SSRC on joining a session; collision detection
end-points will be necessary, as described in "Session Description and resolution for SSRC values.
Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP)
Bandwidth" [RFC3556], or something semantically equivalent. This
also ensures that the end-points have a common view of the RTCP
bandwidth, this is important as too different view of the
bandwidths may lead to failure to interoperate.
These parameters are often expressed in SDP messages conveyed within o Support reception of RTP data packets containing CSRC lists, as
an offer/answer exchange. RTP does not depend on SDP or on the generated by RTP mixers.
offer/answer model, but does require all the necessary parameters to
be agreed somehow, and provided to the RTP implementation. We note
that in RTCWEB context it will depend on the signalling model and API
how these parameters need to be configured but they will be need to
either set in the API or explicitly signalled between the peers.
4. WebRTC Use of RTP: Core Protocols o Support for sending correct synchronization information in the
RTCP Sender Reports, with RECOMMENDED support for the rapid RTP
synchronisation extensions (see Section 5.2.1).
RTP and RTCP are flexible and extensible protocols that allow, on the o Support for standard RTCP packet types, include SR, RR, SDES, and
one hand, choosing from a variety of building blocks and combining BYE packets.
those to meet application needs, but on the other hand, offer the
ability to create extensions where existing mechanisms are not
sufficient. This memo requires a number of RTP and RTCP extensions
that have been shown to be provide important functionality in the
WebRTC context be implemented. It is possible that future extensions
will be needed: several documents provide guidelines for the use and
extension of RTP and RTCP, including Guidelines for Writers of RTP
Payload Format Specifications [RFC2736] and Guidelines for Extending
the RTP Control Protocol [RFC5968], and should be consulted before
extending this memo.
The following describe core features of RTP, and core extensions, o Support for multiple end-points in a single RTP session, and for
that must be supported in all WebRTC implementations. scaling the RTCP transmission interval according to the number of
participants in the session; support randomised RTCP transmission
intervals to avoid synchronisation of RTCP reports.
4.1. RTP and RTCP It is known that a significant number of legacy RTP implementations,
especially those targetted for purely VoIP systems, do not support
all of the above features.
The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be Other implementation considerations are discussed in Section 12.
implemented as the media transport protocol for WebRTC. RTP itself
comprises two parts: the RTP data transfer protocol, and the RTP
control protocol (RTCP). RTCP is a fundamental and integral part of
the RTP protocol, and is REQUIRED to be implemented.
4.2. Choice of RTP Profile 4.2. Choice of RTP Profile
The complete specification of RTP for a particular application domain The complete specification of RTP for a particular application domain
requires the choice of an RTP Profile. For WebRTC use, the "Extended requires the choice of an RTP Profile. For WebRTC use, the "Extended
Secure RTP Profile for Real-time Transport Control Protocol (RTCP)- Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-
Based Feedback (RTP/SAVPF)" [RFC5124] is REQUIRED to be implemented. Based Feedback (RTP/SAVPF)" [RFC5124] is REQUIRED to be implemented.
This builds on the basic RTP/AVP profile [RFC3551], the RTP/AVPF This builds on the basic RTP/AVP profile [RFC3551], the RTP profile
feedback profile [RFC4585], and the secure RTP/SAVP profile for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP
[RFC3711]. profile (RTP/SAVP) [RFC3711].
The RTP/AVPF part of RTP/SAVPF is required to get the improved RTCP The RTP/AVPF part of RTP/SAVPF is required to get the improved RTCP
timer model, that allows more flexible transmission of RTCP packets timer model, that allows more flexible transmission of RTCP packets
in response to events, rather than strictly according to bandwidth. in response to events, rather than strictly according to bandwidth.
This also saves RTCP bandwidth and will commonly only use the full This is vital for being able to report congestion events. The RTP/
amount when there is a lot of events on which to send feedback. This AVPF profile also saves RTCP bandwidth, and will commonly only use
functionality is needed to make use of the RTP conferencing the full RTCP bandwidth allocation when there are many events that
extensions discussed in Section 5.1. The improved RTCP timer model require feedback. The RTP/AVPF functionality is also needed to make
defined by RTP/AVPF is backwards compatible with legacy systems that use of the RTP conferencing extensions discussed in Section 5.1.
implement only the RTP/AVP profile given some constraints on
parameter configuration such as RTCP banwidth and "trr-int".
The RTP/SAVP part of RTP/SAVPF is for support for Secure RTP (SRTP) Note: The enhanced RTCP timer model defined in the RTP/AVPF
[RFC3711]. This provides media encryption, integrity protection, profile is backwards compatible with legacy systems that implement
replay protection and a limited form of source authentication. only the base RTP/AVP profile, given some constraints on parameter
configuration such as the RTCP bandwidth value and "trr-int" (the
most important factor for interworking with RTP/AVP end-points via
a gateway is to set the trr-int parameter to a value representing
4 seconds).
(tbd: There is ongoing discussion on what keying mechanism is to be The RTP/SAVP part of the RTP/SAVPF profile is for support for Secure
required, what are the mandated cryptographic transforms, and whether RTP (SRTP) [RFC3711]. This provides media encryption, integrity
fallback to plain RTP should be supported. This section needs to be protection, replay protection and a limited form of source
updated based on the results of that discussion.) authentication.
WebRTC implementation MUST NOT send packets using the RTP/AVP profile
or the RTP/AVPF profile; they MUST use the RTP/SAVPF profile. WebRTC
implementations MUST support DTLS-SRTP [RFC5764] for key-management.
(tbd: There is ongoing discussion on what additional keying mechanism
is to be required, what are the mandated cryptographic transforms.
This section needs to be updated based on the results of that
discussion.)
4.3. Choice of RTP Payload Formats 4.3. Choice of RTP Payload Formats
(tbd: say something about the choice of RTP Payload Format for (tbd: say something about the choice of RTP Payload Format for
WebRTC. If there is a mandatory to implement set of codecs, this WebRTC. If there is a mandatory to implement set of codecs, this
should reference them. In any case, it should reference a discussion should reference them. In any case, it should reference a discussion
of signalling for the choice of codec, once that discussion reaches of signalling for the choice of codec, once that discussion reaches
closure.) closure.)
Endpoints may signal support for multiple media formats, or multiple Endpoints may signal support for multiple media formats, or multiple
configurations of a single format, provided each uses a different RTP configurations of a single format, provided each uses a different RTP
payload type number. An endpoint that has signalled it's support for payload type number. An endpoint that has signalled it's support for
multiple formats is REQUIRED to accept data in any of those formats multiple formats is REQUIRED to accept data in any of those formats
at any time, unless it has previously signalled limitations on it's at any time, unless it has previously signalled limitations on it's
decoding capability (this is modified if several RTP sessions are to decoding capability. This is modified if several media types are
be multiplexed onto one transport layer connection, such that an sent in the same RTP session, in that case a source (SSRC) is
endpoint must be prepared for a source to switch between formats of restricted to switch between any RTP payload format established for
the same media type at any time; see Section 4.4). To support rapid the media type that is being sent by that source; see Section 4.4.
rate adaptation, RTP does not require advance signalling for changes To support rapid rate adaptation, RTP does not require signalling in
between payload formats that were signalled during session setup. advance for changes between payload formats that were signalled
during session setup.
4.4. RTP Session Multiplexing 4.4. RTP Session Multiplexing
An association amongst a set of participants communicating with RTP An association amongst a set of participants communicating with RTP
is known as an RTP session. A participant may be involved in is known as an RTP session. A participant may be involved in
multiple RTP sessions at the same time. In a multimedia session, multiple RTP sessions at the same time. In a multimedia session,
each medium has typically been carried in a separate RTP session with each medium has typically been carried in a separate RTP session with
its own RTCP packets (i.e., one RTP session for the audio, with a its own RTCP packets (i.e., one RTP session for the audio, with a
separate RTP session running on a different transport connection for separate RTP session running on a different transport connection for
the video; if SDP is used, this corresponds to one RTP session for the video; if SDP is used, this corresponds to one RTP session for
each "m=" line in the SDP). WebRTC implementations of RTP are each "m=" line in the SDP). WebRTC implementations of RTP are
REQUIRED to implement support of multimedia sessions in this way, for REQUIRED to implement support for multimedia sessions in this way,
compatibility with legacy systems. for compatibility with legacy systems.
In today's networks, however, with the prolific use of Network In today's networks, however, with the widespread use of Network
Address Translators (NAT) and Firewalls (FW), there is a desire to Address/Port Translators (NAT/NAPT) and Firewalls (FW), it is
reduce the number of transport layer ports used by real-time media desirable to reduce the number of transport layer ports used by real-
applications using RTP by combining multimedia traffic in a single time media applications using RTP by combining multimedia traffic in
RTP session. (Details of how this is to be done are tbd, but see a single RTP session. (Details of how this is to be done are tbd,
[I-D.lennox-rtcweb-rtp-media-type-mux], but see [I-D.lennox-rtcweb-rtp-media-type-mux],
[I-D.holmberg-mmusic-sdp-bundle-negotiation] and [I-D.holmberg-mmusic-sdp-bundle-negotiation] and
[I-D.westerlund-avtcore-multiplex-architecture].) WebRTC [I-D.westerlund-avtcore-multiplex-architecture].) Using a single RTP
implementations of RTP are REQUIRED to support multiplexing of session also effects the possibility for differentiated treament of
multimedia sessions onto a single RTP session according to (tbd). If media flows. This is further discussed in Section 12.9.
such RTP session multiplexing is to be used, this MUST be negotiated
during the signalling phase. WebRTC implementations of RTP are REQUIRED to support multiplexing of
a multimedia session onto a single RTP session according to (tbd).
If such RTP session multiplexing is to be used, this MUST be
negotiated during the signalling phase. Support for multiple RTP
sessions over a single UDP flow as defined by
[I-D.westerlund-avtcore-transport-multiplexing] is RECOMMENDED.
4.5. RTP and RTCP Multiplexing 4.5. RTP and RTCP Multiplexing
Historically, RTP and RTCP have been run on separate UDP ports. With Historically, RTP and RTCP have been run on separate transport-layer
the increased use of Network Address/Port Translation (NAPT) this has ports (e.g., two UDP ports for each RTP session, one port for RTP and
become problematic, since maintaining multiple NAT bindings can be one port for RTCP). With the increased use of Network Address/Port
costly. It also complicates firewall administration, since multiple Translation (NAPT) this has become problematic, since maintaining
ports must be opened to allow RTP traffic. To reduce these costs and multiple NAT bindings can be costly. It also complicates firewall
session setup times, support for multiplexing RTP data packets and administration, since multiple ports must be opened to allow RTP
RTCP control packets on a single port [RFC5761] is REQUIRED. traffic. To reduce these costs and session setup times, support for
multiplexing RTP data packets and RTCP control packets on a single
port [RFC5761] for each RTP session is REQUIRED.
Note that the use of RTP and RTCP multiplexed on a single port (tbd: Are WebRTC implementations required to support the case where
ensures that there is occasional traffic sent on that port, even if the RTP and RTCP are run on separate UDP ports, for interoperability
there is no active media traffic. This may be useful to keep-alive with legacy systems?)
NAT bindings and is recommend method for application level keep-
alives of RTP sessions [RFC6263]. Note that the use of RTP and RTCP multiplexed onto a single transport
port ensures that there is occasional traffic sent on that port, even
if there is no active media traffic. This may be useful to keep-
alive NAT bindings, and is the recommend method for application level
keep-alives of RTP sessions [RFC6263].
4.6. Reduced Size RTCP 4.6. Reduced Size RTCP
RTCP packets are usually sent as compound RTCP packets, and RFC 3550 RTCP packets are usually sent as compound RTCP packets, and [RFC3550]
requires that those compound packets start with an SR or RR packet. requires that those compound packets start with an Sender Report (SR)
When using frequent feedback messages, these general statistics are or Receiver Report (RR) packet. When using frequent RTCP feedback
not needed in every packet and unnecessarily increase the mean RTCP messages, these general statistics are not needed in every packet and
packet size. This can limit the frequency at which RTCP packets can unnecessarily increase the mean RTCP packet size. This can limit the
be sent within the RTCP bandwidth share. frequency at which RTCP packets can be sent within the RTCP bandwidth
share.
To avoid this problem, [RFC5506] specifies how to reduce the mean To avoid this problem, [RFC5506] specifies how to reduce the mean
RTCP message and allow for more frequent feedback. Frequent RTCP message and allow for more frequent feedback. Frequent
feedback, in turn, is essential to make real-time application quickly feedback, in turn, is essential to make real-time application quickly
aware of changing network conditions and allow them to adapt their aware of changing network conditions and allow them to adapt their
transmission and encoding behaviour. Support for RFC5506 is transmission and encoding behaviour. Support for RFC5506 is
REQUIRED. REQUIRED.
4.7. Symmetric RTP/RTCP 4.7. Symmetric RTP/RTCP
RTP entities choose the RTP and RTCP transport addresses, i.e., IP To ease traversal of NAT and firewall devices, implementations are
addresses and port numbers, to receive packets on and bind their REQUIRED to implement Symmetric RTP [RFC4961]. This requires that
respective sockets to those. When sending RTP packets, however, they the IP address and port used for sending and receiving RTP and RTCP
may use a different IP address or port number for RTP, RTCP, or both; packets are identical. The reasons for using symmetric RTP is
e.g., when using a different socket instance for sending and for primarily to avoid issues with NAT and Firewalls by ensuring that the
receiving. Symmetric RTP/RTCP requires that the IP address and port flow is actually bi-directional and thus kept alive and registered as
number for sending and receiving RTP/RTCP packets are identical. flow the intended recipient actually wants. In addition it saves
resources in the form of ports at the end-points, but also in the
The reasons for using symmetric RTP is primarily to avoid issues with network as NAT mappings or firewall state is not unnecessary bloated.
NAT and Firewalls by ensuring that the flow is actually bi- Also the amount of QoS state is reduced.
directional and thus kept alive and registered as flow the intended
recipient actually wants. In addition it saves resources in the form
of ports at the end-points, but also in the network as NAT mappings
or firewall state is not unnecessary bloated. Also the number of QoS
state are reduced.
Using Symmetric RTP and RTCP [RFC4961] is REQUIRED.
4.8. Generation of the RTCP Canonical Name (CNAME) 4.8. Generation of the RTCP Canonical Name (CNAME)
The RTCP Canonical Name (CNAME) provides a persistent transport-level The RTCP Canonical Name (CNAME) provides a persistent transport-level
identifier for an RTP endpoint. While the Synchronisation Source identifier for an RTP endpoint. While the Synchronisation Source
(SSRC) identifier for an RTP endpoint may change if a collision is (SSRC) identifier for an RTP endpoint may change if a collision is
detected, or when the RTP application is restarted, it's RTCP CNAME detected, or when the RTP application is restarted, it's RTCP CNAME
is meant to stay unchanged, so that RTP endpoints can be uniquely is meant to stay unchanged, so that RTP endpoints can be uniquely
identified and associated with their RTP media streams. For proper identified and associated with their RTP media streams. For proper
functionality, RTCP CNAMEs should be unique among the participants of functionality, each RTP endpoint needs to have a unique RTCP CNAME
an RTP session. value.
The RTP specification [RFC3550] includes guidelines for choosing a The RTP specification [RFC3550] includes guidelines for choosing a
unique RTP CNAME, but these are not sufficient in the presence of NAT unique RTP CNAME, but these are not sufficient in the presence of NAT
devices. In addition, some may find long-term persistent identifiers devices. In addition, some may find long-term persistent identifiers
problematic from a privacy viewpoint. Accordingly, support for problematic from a privacy viewpoint. Accordingly, support for
generating a short-term persistent RTCP CNAMEs following method (b) generating a short-term persistent RTCP CNAMEs following method (b)
as specified in Section 4.2 of "Guidelines for Choosing RTP Control specified in Section 4.2 of "Guidelines for Choosing RTP Control
Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is REQUIRED, Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is REQUIRED,
since this addresses both concerns. since this addresses both concerns.
5. WebRTC Use of RTP: Extensions 5. WebRTC Use of RTP: Extensions
There are a number of RTP extensions that could be very useful in the There are a number of RTP extensions that are either required to
WebRTC context. One set is related to conferencing, others are more obtain full functionality, or extremely useful to improve on the
generic in nature. None of these are mandatory to implement, but baseline performance, in the WebRTC application context. One set of
they are expected to be generally useful. these extensions is related to conferencing, while others are more
generic in nature. The following subsections describe the various
RTP extensions mandated or strongly recommended within WebRTC.
5.1. Conferencing Extensions 5.1. Conferencing Extensions
RTP is inherently a group communication protocol. Groups can be RTP is inherently a group communication protocol. Groups can be
implemented using a centralised server, multi-unicast, or IP implemented using a centralised server, multi-unicast, or using IP
multicast. While IP multicast was popular in early deployments, in multicast. While IP multicast was popular in early deployments, in
today's practice, overlay-based conferencing dominates, typically today's practice, overlay-based conferencing dominates, typically
using one or more central servers to connect endpoints in a star or using one or more central servers to connect endpoints in a star or
flat tree topology. These central servers can be implemented in a flat tree topology. These central servers can be implemented in a
number of ways [RFC5117], of which the following are the most common: number of ways as discussed in Appendix A, and in the memo on RTP
Topologies [RFC5117].
1. RTP Translator (Relay) with Only Unicast Paths ([RFC5117],
section 3.3)
2. RTP Mixer with Only Unicast Paths ([RFC5117], section 3.4)
3. Point to Multipoint Using a Video Switching MCU ([RFC5117],
section 3.5)
4. Point to Multipoint Using Content Modifying MCUs ([RFC5117],
section 3.6)
As discussed in [RFC5117] section 3.5, the use of a video switching As discussed in Section 3.5 of [RFC5117], the use of a video
MCU makes the use of RTCP for congestion control, or any type of switching MCU makes the use of RTCP for congestion control, or any
quality reports, very problematic. Also, as discussed in [RFC5117] type of quality reports, very problematic. Also, as discussed in
section 3.6, the use of a content modifying MCU with RTCP termination section 3.6 of [RFC5117], the use of a content modifying MCU with
breaks RTP loop detection and removes the ability for receivers to RTCP termination breaks RTP loop detection and removes the ability
identify active senders. According only the first two options are for receivers to identify active senders. Accordingly, only RTP
recommended. Transport Translators (relays), RTP Mixers, and end-point based
forwarding topologies are supported in WebRTC. These RECOMMENDED
topologies are expected to be supported by all WebRTC end-points
(these three topologies require no special support in the end-point,
if the RTP features mandated in this memo are implemented).
RTP protocol extensions to be used with conferencing are included The RTP protocol extensions to be used with conferencing, described
because they are important in the context of centralised below, are not required for correctness; an RTP endpoint that does
not implement these extensions will work correctly, but offer poor
performance. Support for the listed extensions will greatly improve
the quality of experience, however, in the context of centralised
conferencing, where one RTP Mixer (Conference Focus) receives a conferencing, where one RTP Mixer (Conference Focus) receives a
participants media streams and distribute them to the other participants media streams and distribute them to the other
participants. These messages are defined in the Extended RTP Profile participants. These messages are defined in the Extended RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/ for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio- AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124]. usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
5.1.1. Full Intra Request 5.1.1. Full Intra Request
The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of CCM The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of the
[RFC5104]. It is used to have the mixer request from a session Codec Control Messages [RFC5104]. This message is used to have the
participants a new Intra picture. This is used when switching mixer request a new Intra picture from a participant in the session.
between sources to ensure that the receivers can decode the video or This is used when switching between sources to ensure that the
other predicted media encoding with long prediction chains. It is receivers can decode the video or other predicted media encoding with
RECOMMENDED that this feedback message is supported. long prediction chains. It is REQUIRED that this feedback message is
supported by RTP senders in WebRTC, since it greatly improves the
user experience when using centralised mixers-based conferencing.
5.1.2. Picture Loss Indication 5.1.2. Picture Loss Indication
The Picture Loss Indication is defined in Section 6.3.1 of the RTP/ The Picture Loss Indication is defined in Section 6.3.1 of the RTP/
AVPF profile [RFC4585]. It is used by a receiver to tell the encoder AVPF profile [RFC4585]. It is used by a receiver to tell the sending
that it lost the decoder context and would like to have it repaired encoder that it lost the decoder context and would like to have it
somehow. This is semantically different from the Full Intra Request repaired somehow. This is semantically different from the Full Intra
above. It is RECOMMENDED that this feedback message is supported as Request above as there can exist multiple methods to fulfil the
a loss tolerance mechanism. request. It is RECOMMENDED that this feedback message is supported
as a loss tolerance mechanism.
5.1.3. Slice Loss Indication 5.1.3. Slice Loss Indication
The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF
profile [RFC4585]. It is used by a receiver to tell the encoder that profile [RFC4585]. It is used by a receiver to tell the encoder that
it has detected the loss or corruption of one or more consecutive it has detected the loss or corruption of one or more consecutive
macroblocks, and would like to have these repaired somehow. The use macroblocks, and would like to have these repaired somehow. The use
of this feedback message is OPTIONAL as a loss tolerance mechanism. of this feedback message is OPTIONAL as a loss tolerance mechanism.
5.1.4. Reference Picture Selection Indication 5.1.4. Reference Picture Selection Indication
skipping to change at page 13, line 44 skipping to change at page 12, line 18
6.3.3 of the RTP/AVPF profile [RFC4585]. Some video coding standards 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video coding standards
allow the use of older reference pictures than the most recent one allow the use of older reference pictures than the most recent one
for predictive coding. If such a codec is in used, and if the for predictive coding. If such a codec is in used, and if the
encoder has learned about a loss of encoder-decoder synchronicity, a encoder has learned about a loss of encoder-decoder synchronicity, a
known-as-correct reference picture can be used for future coding. known-as-correct reference picture can be used for future coding.
The RPSI message allows this to be signalled. The use of this RTCP The RPSI message allows this to be signalled. The use of this RTCP
feedback message is OPTIONAL as a loss tolerance mechanism. feedback message is OPTIONAL as a loss tolerance mechanism.
5.1.5. Temporary Maximum Media Stream Bit Rate Request 5.1.5. Temporary Maximum Media Stream Bit Rate Request
This feedback message is defined in Section 3.5.4 and 4.2.1 in CCM This feedback message is defined in Sections 3.5.4 and 4.2.1 of the
[RFC5104]. This message and its notification message is used by a Codec Control Messages [RFC5104]. This message and its notification
media receiver, to inform the sending party that there is a current message is used by a media receiver, to inform the sending party that
limitation on the amount of bandwidth available to this receiver. there is a current limitation on the amount of bandwidth available to
This can be for various reasons, and can for example be used by an this receiver. This can be for various reasons, and can for example
RTP mixer to limit the media sender being forwarded by the mixer be used by an RTP mixer to limit the media sender being forwarded by
(without doing media transcoding) to fit the bottlenecks existing the mixer (without doing media transcoding) to fit the bottlenecks
towards the other session participants. It is RECOMMENDED that this existing towards the other session participants. It is REQUIRED that
feedback message is supported. this feedback message is supported.
5.2. Header Extensions 5.2. Header Extensions
The RTP specification [RFC3550] provides a capability to extend the The RTP specification [RFC3550] provides the capability to include
RTP header with in-band data, but the format and semantics of the RTP header extensions containing in-band data, but the format and
extensions are poorly specified. The use of header extensions is semantics of the extensions are poorly specified. The use of header
OPTIONAL, but if are used, it is REQUIRED that they be formatted and extensions is OPTIONAL in the WebRTC context, but if they are used,
signalled according to the general mechanism of RTP header extensions they MUST be formatted and signalled following the general mechanism
defined in [RFC5285]. for RTP header extensions defined in [RFC5285], since this gives
well-defined semantics to RTP header extensions.
As noted in [RFC5285], the requirement from the RTP specification As noted in [RFC5285], the requirement from the RTP specification
that header extensions are "designed so that the header extension may that header extensions are "designed so that the header extension may
be ignored" [RFC3550] stands. To be specific, header extensions must be ignored" [RFC3550] stands. To be specific, header extensions MUST
only be used for data that can safely be ignored by the recipient only be used for data that can safely be ignored by the recipient
without affecting interoperability, and must not be used when the without affecting interoperability, and MUST NOT be used when the
presence of the extension has changed the form or nature of the rest presence of the extension has changed the form or nature of the rest
of the packet in a way that is not compatible with the way the stream of the packet in a way that is not compatible with the way the stream
is signalled (e.g., as defined by the payload type). Valid examples is signalled (e.g., as defined by the payload type). Valid examples
might include metadata that is additional to the usual RTP might include metadata that is additional to the usual RTP
information. information.
The RTP rapid synchronisation header extension [RFC6051] is
recommended, as discussed in Section 5.2.1 we also recommend the
client to mixer audio level [RFC6464], and consider the mixer to
client audio level [RFC6465] as optional feature.
If the client-to-mixer or mixer-to-client audio level indications are
in use in SRTP encrypted sessions, it is REQUIRED that the extensions
are encrypted according to
[I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
contained in these header extensions may be considered sensitive.
5.2.1. Rapid Synchronisation 5.2.1. Rapid Synchronisation
Many RTP sessions require synchronisation between audio, video, and Many RTP sessions require synchronisation between audio, video, and
other content. This synchronisation is performed by receivers, using other content. This synchronisation is performed by receivers, using
information contained in RTCP SR packets, as described in the RTP information contained in RTCP SR packets, as described in the RTP
specification [RFC3550]. This basic mechanism can be slow, however, specification [RFC3550]. This basic mechanism can be slow, however,
so it is RECOMMENDED that the rapid RTP synchronisation extensions so it is RECOMMENDED that the rapid RTP synchronisation extensions
described in [RFC6051] be implemented. The rapid synchronisation described in [RFC6051] be implemented. The rapid synchronisation
extensions use the general RTP header extension mechanism [RFC5285], extensions use the general RTP header extension mechanism [RFC5285],
which requires signalling, but are otherwise backwards compatible. which requires signalling, but are otherwise backwards compatible.
5.2.2. Client to Mixer Audio Level 5.2.2. Client to Mixer Audio Level
The Client to Mixer Audio Level [RFC6464] is an RTP header extension The Client to Mixer Audio Level [RFC6464] is an RTP header extension
used by a client to inform a mixer about the level of audio activity used by a client to inform a mixer about the level of audio activity
in the packet the header is attached to. This enables a central node in the packet the header is attached to. This enables a central node
to make mixing or selection decisions without decoding or detailed to make mixing or selection decisions without decoding or detailed
inspection of the payload. Thus reducing the needed complexity in inspection of the payload. Thus reducing the needed complexity in
some types of central RTP nodes. some types of central RTP nodes. It can also be used to save
decoding resources in a WebRTC receiver in a mesh topology, which if
it has limited decoding resources, may select to decode only the most
relevant media streams based on audio activity levels.
The Client to Mixer Audio Level [RFC6464] is RECOMMENDED to be The Client-to-Mixer Audio Level [RFC6464] extension is RECOMMENDED to
implemented. be implemented. If it is implemented, it is REQUIRED that the header
extensions are encrypted according to
[I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
contained in these header extensions can be considered sensitive.
5.2.3. Mixer to Client Audio Level 5.2.3. Mixer to Client Audio Level
The Mixer to Client Audio Level header extension [RFC6465] provides The Mixer to Client Audio Level header extension [RFC6465] provides
the client with the audio level of the different sources mixed into a the client with the audio level of the different sources mixed into a
common mix from the RTP mixer. Thus enabling a user interface to common mix by a RTP mixer. This enables a user interface to indicate
indicate the relative activity level of a session participant, rather the relative activity level of each session participant, rather than
than just being included or not based on the CSRC field. This is a just being included or not based on the CSRC field. This is a pure
pure optimisations of non critical functions and thus optional optimisations of non critical functions, and is hence OPTIONAL to
functionality. implement. If it is implemented, it is REQUIRED that the header
extensions are encrypted according to
The Mixer to Client Audio Level [RFC6465] is OPTIONAL to implement. [I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
contained in these header extensions can be considered sensitive.
6. WebRTC Use of RTP: Improving Transport Robustness 6. WebRTC Use of RTP: Improving Transport Robustness
There are some tools that can make RTP flows robust against Packet There are some tools that can make RTP flows robust against Packet
loss and reduce the impact on media quality. However they all add loss and reduce the impact on media quality. However they all add
extra bits compared to a non-robust stream. These extra bits needs extra bits compared to a non-robust stream. These extra bits need to
to be considered and the aggregate bit-rate needs to be rate be considered, and the aggregate bit-rate must be rate controlled.
controlled. Thus improving robustness might require a lower base Thus improving robustness might require a lower base encoding
encoding quality but has the potential to give that quality with quality, but has the potential to give that quality with fewer
fewer errors in it. errors. The mechanisms described in the following sub-sections can
be used to improve tolerance to packet loss.
6.1. Retransmission 6.1. Retransmission
Support for RTP retransmission as defined by "RTP Retransmission Support for RTP retransmission as defined by "RTP Retransmission
Payload Format" [RFC4588] is RECOMMENDED. Payload Format" [RFC4588] is RECOMMENDED.
The retransmission scheme in RTP allows flexible application of The retransmission scheme in RTP allows flexible application of
retransmissions. Only selected missing packets can be requested by retransmissions. Only selected missing packets can be requested by
the receiver. It also allows for the sender to prioritise between the receiver. It also allows for the sender to prioritise between
missing packets based on senders knowledge about their content. missing packets based on senders knowledge about their content.
Compared to TCP, RTP retransmission also allows one to give up on a Compared to TCP, RTP retransmission also allows one to give up on a
packet that despite retransmission(s) still has not been received packet that despite retransmission(s) still has not been received
within a time window. within a time window.
"WebRTC Media Transport Requirements" [I-D.cbran-rtcweb-data] raises "WebRTC Media Transport Requirements" [I-D.cbran-rtcweb-data] raises
two issues that they think makes RTP Retransmission unsuitable for two issues that they think makes RTP Retransmission unsuitable for
RTCWEB. We here consider these issues and explain why they are in WebRTC. We here consider these issues and explain why they are in
fact not a reason to exclude RTP retransmission from the tool box fact not a reason to exclude RTP retransmission from the tool box
available to RTCWEB media sessions. available to WebRTC media sessions.
The additional latency added by [RFC4588] will exceed the latency The additional latency added by [RFC4588] will exceed the latency
threshold for interactive voice and video: RTP Retransmission will threshold for interactive voice and video: RTP Retransmission will
require at least one round trip time for a retransmission request require at least one round trip time for a retransmission request
and repair packet to arrive. Thus the general suitability of and repair packet to arrive. Thus the general suitability of
using retransmissions will depend on the actual network path using retransmissions will depend on the actual network path
latency between the end-points. In many of the actual usages the latency between the end-points. In many of the actual usages the
latency between two end-points will be low enough for RTP latency between two end-points will be low enough for RTP
retransmission to be effective. Interactive communication with retransmission to be effective. Interactive communication with
end-to-end delays of 400 ms still provide a fair quality. Even end-to-end delays of 400 ms still provide a fair quality. Even
skipping to change at page 16, line 30 skipping to change at page 14, line 50
The undesirable increase in packet transmission at the point when The undesirable increase in packet transmission at the point when
congestion occurs: Congestion loss will impact the rate controls congestion occurs: Congestion loss will impact the rate controls
view of available bit-rate for transmission. When using view of available bit-rate for transmission. When using
retransmission one will have to prioritise between performing retransmission one will have to prioritise between performing
retransmissions and the quality one can achieve with ones retransmissions and the quality one can achieve with ones
adaptable codecs. In many use cases one prefer error free or low adaptable codecs. In many use cases one prefer error free or low
rates of error with reduced base quality over high degrees of rates of error with reduced base quality over high degrees of
error at a higher base quality. error at a higher base quality.
The RTCWEB end-point implementations will need to both select when to The WebRTC end-point implementations will need to both select when to
enable RTP retransmissions based on API settings and measurements of enable RTP retransmissions based on API settings and measurements of
the actual round trip time. In addition for each NACK request that a the actual round trip time. In addition for each NACK request that a
media sender receives it will need to make a prioritisation based on media sender receives it will need to make a prioritisation based on
the importance of the requested media, the probability that the the importance of the requested media, the probability that the
packet will reach the receiver in time for being usable, the packet will reach the receiver in time for being usable, the
consumption of available bit-rate and the impact of the media quality consumption of available bit-rate and the impact of the media quality
for new encodings. for new encodings.
To conclude, the issues raised are implementation concerns that an To conclude, the issues raised are implementation concerns that an
implementation needs to take into consideration, they are not implementation needs to take into consideration, they are not
skipping to change at page 19, line 22 skipping to change at page 17, line 41
Forward Error Correction (FEC) Framework: This framework Forward Error Correction (FEC) Framework: This framework
[I-D.ietf-fecframe-framework] defines how not only RTP packets but [I-D.ietf-fecframe-framework] defines how not only RTP packets but
how arbitrary packet flows can be protected. Some solutions how arbitrary packet flows can be protected. Some solutions
produced or under development in FECFRAME WG are RTP specific. produced or under development in FECFRAME WG are RTP specific.
There exist alternatives supporting block codes such as Reed- There exist alternatives supporting block codes such as Reed-
Salomon and Raptor. Salomon and Raptor.
6.2.3. Recommendations for FEC 6.2.3. Recommendations for FEC
(tbd) Open Issue: Decision of need for FEC and if to be included in
recommendation which FEC scheme to be supported needs to be
documented.
7. WebRTC Use of RTP: Rate Control and Media Adaptation 7. WebRTC Use of RTP: Rate Control and Media Adaptation
WebRTC will be used in very varied network environment with a WebRTC will be used in very varied network environment with a
hetrogenous set of link technologies, including wired and wireless, hetrogenous set of link technologies, including wired and wireless,
interconnecting peers at different topological locations resulting in interconnecting peers at different topological locations resulting in
network paths with widely varying one way delays, bit-rate capacity, network paths with widely varying one way delays, bit-rate capacity,
load levels and traffic mixes. In addition individual end-points load levels and traffic mixes. In addition individual end-points
will open one or more WebRTC sessions between one or more peers. will open one or more WebRTC sessions between one or more peers.
Each of these session may contain different mixes of media and data Each of these session may contain different mixes of media and data
skipping to change at page 20, line 21 skipping to change at page 18, line 43
web-service capable of attracting large number of end-points and use web-service capable of attracting large number of end-points and use
the service, combined with BGP routing state a server could pick the service, combined with BGP routing state a server could pick
client pairs to drive traffic to specific paths. client pairs to drive traffic to specific paths.
The above estalish a clear need based on several reasons why there The above estalish a clear need based on several reasons why there
need to be a well working media adaptation mechanism. This mechanism need to be a well working media adaptation mechanism. This mechanism
also have a number of requirements on what services it should provide also have a number of requirements on what services it should provide
and what performance it needs to provide. and what performance it needs to provide.
The biggest issue is that there are no standardised and ready to use The biggest issue is that there are no standardised and ready to use
mechanism that can simply be included in WebRTC Thus there will be mechanism that can simply be included in WebRTC. Thus there will be
need for the IETF to produce such a specification. Therefore the need for the IETF to produce such a specification. Therefore the
suggested way forward is to specify requirements on any solution for suggested way forward is to specify requirements on any solution for
the media adaptation. These requirements is for now proposed to be the media adaptation. These requirements is for now proposed to be
documented in this specification. In addition a proposed detailed documented in this specification. In addition a proposed detailed
solution will be developed, but is expected to take longer time to solution will be developed, but is expected to take longer time to
finalize than this document. finalize than this document.
7.1. Congestion Control Requirements 7.1. Congestion Control Requirements
Requirements for congestion control of WebRTC sessions are discussed Requirements for congestion control of WebRTC sessions are discussed
in [I-D.jesup-rtp-congestion-reqs]. in [I-D.jesup-rtp-congestion-reqs].
Implementations are REQUIRED to implement the RTP circuit breakers Implementations are REQUIRED to implement the RTP circuit breakers
described in [I-D.perkins-avtcore-rtp-circuit-breakers]. described in [I-D.perkins-avtcore-rtp-circuit-breakers].
7.2. RTCP Limiations 7.2. Rate Control Boundary Conditions
The session establishment signalling will establish certain boundary
that the media bit-rate adaptation can act within. First of all the
set of media codecs provide practical limitations in the supported
bit-rate span where it can provide useful quality, which
packetization choices that exist. Next the signalling can establish
maximum media bit-rate boundaries using SDP b=AS or b=CT.
7.3. RTCP Limiations
Experience with the congestion control algorithms of TCP [RFC5681], Experience with the congestion control algorithms of TCP [RFC5681],
TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
that feedback on packet arrivals needs to be sent roughly once per that feedback on packet arrivals needs to be sent roughly once per
round trip time. We note that the capabilities of real-time media round trip time. We note that the capabilities of real-time media
traffic to adapt to changing path conditions may be less rapid than traffic to adapt to changing path conditions may be less rapid than
for the elastic applications TCP was designed for, but frequent for the elastic applications TCP was designed for, but frequent
feedback is still required to allow the congestion control algorithm feedback is still required to allow the congestion control algorithm
to track the path dynamics. to track the path dynamics.
skipping to change at page 21, line 25 skipping to change at page 20, line 10
In group communication, the share of RTCP bandwidth needs to be In group communication, the share of RTCP bandwidth needs to be
shared by all group members, reducing the capacity and thus the shared by all group members, reducing the capacity and thus the
reporting frequency per node. reporting frequency per node.
Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP
bandwidth, split across two entities in a point-to-point session. An bandwidth, split across two entities in a point-to-point session. An
endpoint could thus send a report of 100 bytes about every 70ms or endpoint could thus send a report of 100 bytes about every 70ms or
for every other frame in a 30 fps video. for every other frame in a 30 fps video.
7.3. Legacy Interop Limitations 7.4. Legacy Interop Limitations
Congestion control interoperability with most type of legacy devices, Congestion control interoperability with most type of legacy devices,
even using an translator could be difficult. There are numerous even using an translator could be difficult. There are numerous
reasons for this: reasons for this:
No RTCP Support: There exist legacy implementations that does not No RTCP Support: There exist legacy implementations that does not
even implement RTCP at all. Thus no feedback at all is provided. even implement RTCP at all. Thus no feedback at all is provided.
RTP/AVP Minimal RTCP Interval of 5s: RTP [RFC3550] under the RTP/AVP RTP/AVP Minimal RTCP Interval of 5s: RTP [RFC3550] under the RTP/AVP
profile specifies a recommended minimal fixed interval of 5 profile specifies a recommended minimal fixed interval of 5
skipping to change at page 22, line 11 skipping to change at page 20, line 43
high bandwidth sessions, getting frequent enough RTCP Report high bandwidth sessions, getting frequent enough RTCP Report
Blocks to report on the order of the RTT is very difficult as long Blocks to report on the order of the RTT is very difficult as long
as the legacy device uses the RTP/AVP profile. as the legacy device uses the RTP/AVP profile.
RTP/AVPF Supporting Legacy Device: If a legacy device supports RTP/ RTP/AVPF Supporting Legacy Device: If a legacy device supports RTP/
AVPF, then that enables negotation of important parameters for AVPF, then that enables negotation of important parameters for
frequent reporting, such as the "trr-int" parameter, and the frequent reporting, such as the "trr-int" parameter, and the
possibility that the end-point supports some useful feedback possibility that the end-point supports some useful feedback
format for congestion control purpose such as TMMBR [RFC5104]. format for congestion control purpose such as TMMBR [RFC5104].
It has been suggested on the RTCWEB mailing list that if It has been suggested on the WebRTC mailing list that if
interoperating with really limited legacy devices an WebRTC end-point interoperating with really limited legacy devices an WebRTC end-point
may not send more than 64 kbps of media streams, to avoid it causing may not send more than 64 kbps of media streams, to avoid it causing
massive congestion on most paths in the Internet when communicating massive congestion on most paths in the Internet when communicating
with a legacy node not providing sufficient feedback for effective with a legacy node not providing sufficient feedback for effective
congestion control. This warrants further discussion as there is congestion control. This warrants further discussion as there is
clearly a number of link layers that don't even provide that amount clearly a number of link layers that don't even provide that amount
of bit-rate consistently, and that assumes no competing traffic. of bit-rate consistently, and that assumes no competing traffic.
8. WebRTC Use of RTP: Performance Monitoring 8. WebRTC Use of RTP: Performance Monitoring
skipping to change at page 22, line 34 skipping to change at page 21, line 19
could be included in the set to be supported. However, in most cases could be included in the set to be supported. However, in most cases
which RTP monitoring that is needed depends on the application, which which RTP monitoring that is needed depends on the application, which
makes it difficult to select which to include when the set of makes it difficult to select which to include when the set of
applications is very large. applications is very large.
Exposing some metrics in the WebRTC API should be considered allowing Exposing some metrics in the WebRTC API should be considered allowing
the application to gather the measurements of interest. However, the application to gather the measurements of interest. However,
security implications for the different data sets exposed will need security implications for the different data sets exposed will need
to be considered in this. to be considered in this.
9. IANA Considerations 9. WebRTC Use of RTP: Future 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
of WebRTC applications. In this case, future updates to this memo
MUST be made following the Guidelines for Writers of RTP Payload
Format Specifications [RFC2736] and Guidelines for Extending the RTP
Control Protocol [RFC5968], and SHOULD take into account any future
guidelines for extending RTP and related protocols that have been
developed.
Authors of future extensions are urged to consider the wide range of
environments in which RTP is used when recommending extensions, since
extensions that are applicable in some scenarios can be problematic
in others. Where possible, the WebRTC framework should adopt RTP
extensions that are of general utility, to enable easy gatewaying to
other applications using RTP, rather than adopt mechanisms that are
narrowly targetted at specific WebRTC use cases.
10. Signalling Considerations
RTP is built with the assumption of an external signalling channel
that can be used to configure the RTP sessions and their features.
The basic configuration of an RTP session consists of the following
parameters:
RTP Profile: The name of the RTP profile to be used in session. The
RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
on basic level, as can their secure variants RTP/SAVP [RFC3711]
and RTP/SAVPF [RFC5124]. The secure variants of the profiles do
not directly interoperate with the non-secure variants, due to the
presence of additional header fields in addition to any
cryptographic transformation of the packet content. As WebRTC
requires the usage of the SAVPF profile only a single profile will
need to be signalled. Interworking functions may transform this
into SAVP for a legacy use case by indicating to the WebRTC end-
point a SAVPF end-point and limiting the usage of the a=rtcp
attribute to indicate a trr-int value of 4 seconds.
Transport Information: Source and destination address(s) and ports
for RTP and RTCP MUST be signalled for each RTP session. In
WebRTC these end-points will be provided by ICE that signalls
candidates and arrive at nominated candidate pairs. If RTP and
RTCP multiplexing [RFC5761] is to be used, such that a single port
is used for RTP and RTCP flows, this MUST be signalled (see
Section 4.5). If several RTP sessions are to be multiplexed onto
a single transport layer flow, this MUST also be signalled (see
Section 4.4).
RTP Payload Types, media formats, and media format
parameters: The mapping between media type names (and hence the RTP
payload formats to be used) and the RTP payload type numbers must
be signalled. Each media type may also have a number of media
type parameters that must also be signalled to configure the codec
and RTP payload format (the "a=fmtp:" line from SDP).
RTP Extensions: The RTP extensions one intends to use need to be
agreed upon, including any parameters for each respective
extension. At the very least, this will help avoiding using
bandwidth for features that the other end-point will ignore. But
for certain mechanisms there is requirement for this to happen as
interoperability failure otherwise happens.
RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the
end-points will be necessary, as described in "Session Description
Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP)
Bandwidth" [RFC3556], or something semantically equivalent. This
also ensures that the end-points have a common view of the RTCP
bandwidth, this is important as too different view of the
bandwidths may lead to failure to interoperate.
These parameters are often expressed in SDP messages conveyed within
an offer/answer exchange. RTP does not depend on SDP or on the
offer/answer model, but does require all the necessary parameters to
be agreed somehow, and provided to the RTP implementation. We note
that in the WebRTC context it will depend on the signalling model and
API how these parameters need to be configured but they will be need
to either set in the API or explicitly signalled between the peers.
11. WebRTC API Considerations
The following sections describe how the WebRTC API features map onto
the RTP mechanisms described in this memo.
11.1. API MediaStream to RTP Mapping
The WebRTC API and its media function have the concept of a
MediaStream that consists of zero or more tracks. Where a track is
an individual stream of media from any type of media source like a
microphone or a camera, but also coneptual sources, like a audio mix
or a video composition. The tracks within a MediaStream are expected
to be synchronized.
A track correspondes to the media received with one particular SSRC.
There might be additional SSRCs associated with that SSRC, like for
RTP retransmission or Forward Error Correction. However, one SSRC
will identify a media stream and its timing.
Thus a MediaStream is a collection of SSRCs carrying the different
media included in the synchornized aggregate. Thus also the
synchronization state associated with the included SSRCs are part of
concept. One important thing to consider is that there can be
multiple different MediaStreams containing a given Track (SSRC).
Thus to avoid unnecessary duplication of media at transport level one
need to do the binding of which MediaStreams a given SSRC is
associated with at signalling level.
A proposal for how the binding between MediaStreams and SSRC can be
done exist in "Cross Session Stream Identification in the Session
Description Protocol" [I-D.alvestrand-rtcweb-msid].
12. RTP Implementation Considerations
The following provide some guidance on the implementation of the RTP
features described in this memo.
This section discusses RTP functionality that is part of the RTP
standard, required by decisions made, or to enable use cases raised
and their motivations. This discussion is done from an WebRTC end-
point perspective. It will occassional go into central nodes, but as
the specification is for an end-point that is where the focus lies.
For more discussion on the central nodes and details about RTP
topologies please reveiw Appendix A.
The section will touch on the relation with certain RTP/RTCP
extensions, but will focus on the RTP core functionality. The
definition of what functionalities and the level of requirement on
implementing it is defined in Section 2.
12.1. RTP Sessions and PeerConnection
An RTP session is an association among RTP nodes, which have one
common SSRC space. An RTP session can include any number of end-
points and nodes sourcing, sinking, manipulating or reporting on the
media streams being sent within the RTP session. A PeerConnection
being a point to point association between an end-point and another
node. That peer node may be both an end-point or centralized
processing node of some type, thus the RTP session may terminate
immediately on the far end of the PeerConnection, but it may also
continue as further discused below in Multiparty (Section 12.3) and
Multiple RTP End-points (Section 12.7).
A PeerConnection can contain one or more RTP session depending on how
it is setup and how many UDP flows it uses. A common usage has been
to have one RTP session per media type, e.g. one for audio and one
for Video, each sent over different UDP flows. However, the default
usage in WebRTC will be to use one RTP session for all media types.
This usage then uses only one UDP flow, as also RTP and RTCP
multiplexing is mandated (Section 4.5). However, for legacy
interworking and network prioritization (Section 12.9) based on flows
a WebRTC end-point needs to support a mode of operation where one RTP
session per media type is used. Currently each RTP session must use
its own UDP flow. Discussion are ongoing if a solution enabling
multiple RTP sessions over a single UDP flow, see Section 4.4.
The multi-unicast or mesh based multi-party topology (Figure 1) is
best to raise in this section as it concers the relation between RTP
sessions and PeerConnections. In this topology, each participant
sends individual unicast RTP/UDP/IP flows to each of the other
participants using independent PeerConnections in a full mesh. This
topology has the benefit of not requiring central nodes. The
downside is that it increases the used bandwidth at each sender by
requiring one copy of the media streams for each participant that are
part of the same session beyond the sender itself. Hence, this
topology is limited to scenarios with few participants unless the
media is very low bandwidth.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 1: Multi-unicast
The multi-unicast topology could be implemented as a single RTP
session, spanning multiple peer-to-peer transport layer connections,
or as several pairwise RTP sessions, one between each pair of peers.
To maintain a coherent mapping between the relation between RTP
sessions and PeerConnections we recommend that one implements this as
individual RTP sessions. The only downside is that end-point A will
not learn of the quality of any transmission happening between B and
C based on RTCP. This has not been seen as a significant downside as
no one has yet seen a clear need for why A would need to know about
the B's and C's communication. An advantage of using separate RTP
sessions is that it enables using different media bit-rates to the
differnt peers, thus not forcing B to endure the same quality
reductions if there are limiations in the transport from A to C as C
will.
12.2. Multiple Sources
A WebRTC end-point may have multiple cameras, microphones or audio
inputs thus a single end-point can source multiple media streams
concurrently of the same media type. In addition the above discussed
criteria to support multiple media types in one single RTP session
results that also an end-point that has one audio and one video
source still need two transmit using two SSRCs concurrently. As
multi-party conferences are supported, as discussed below in
Section 12.3, a WebRTC end-point will need to be capable of
receiving, decoding and playout multiple media streams of the same
type concurrently.
Open Issue:Are any mechanism needed to signal limiations in the
number of SSRC that an end-point can handle?
12.3. Multiparty
There exist numerous situations and clear use cases for WebRTC
supporting sessions supoprting multi-party. This can be realized in
a number of ways using a number of different implementations
strategies. This focus on the different set of WebRTC end-point
requirements that arise from different sets of multi-party
topologies.
The multi-unicast mesh (Figure 1) based multi-party topoology
discussed above provides a non-centralized solution but can easily
tax the end-points outgoing paths. It may also consume large amount
of encoding resources if each outgoing stream is specifically
encoded. If an encoding is transmitted to multiple parties, either
as in the mesh case or when using relaying central nodes (see below)
a requirement on the end-point becomes to be able to create media
streams suitable to multiple destinations requirements. These
requirements may both be dependent on transport path and the
different end-points preferences related to playout of the media.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 2: RTP Mixer with Only Unicast Paths
A Mixer (Figure 2) is an RTP end-point that optimizes the
transmission of media streams from certain perspectives, either by
only sending some of the received media stream to any given receiver
or by providing a combined media stream out of a set of contributing
streams. There exist various methods of implementation as discussed
in Appendix A.3. A common aspect is that these central nodes a
number of tools to control the media encoding provided by a WebRTC
end-point. This includes functions like requesting breaking the
encoding chain and have the encoder produce a so called Intra frame.
Another is limiting the bit-rate of a given stream to better suit the
mixer view of the multiple down-streams. Others are controling the
most suitable frame-rate, picture resultion, the trade-off between
frame-rate and spatial quality.
A mixer gets a significant responsibility to correctly perform
congestion control, identity management, manage synchronization while
providing a for the application suitable media optimization.
Mixers also need to be a trusted node when it comes to security as it
manipulates either RTP or the media itself before sending it on
towards the end-point(s) thus must be able to decrypt and then
encrypt it before sending it out. There exist one type of central
node, the relay that one doesn't need to trust with the keys to the
media. The relay operates only on the IP/UDP level of the transport.
It is configured so that it would forward any RTP/RTCP packets from A
to the other participants B-D.
+---+ +---+
| | +-----------+ | |
| A |<------->| DTLS-SRTP |<------->| C |
| |<-- -->| HOST |<-- -->| |
+---+ \ / +-----------+ \ / +---+
X X
+---+ / \ +-----------+ / \ +---+
| |<-- -->| RTP |<-- -->| |
| B |<------->| RELAY |<------->| D |
| | +-----------+ | |
+---+ +---+
Figure 3: DTLS-SRTP host and RTP Relay Separated
To accomplish the security properties discussed above using a relay
one need to have a separate key handling server and also support for
distribute the different keys such as Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt]. The relay also creates a situation where
there is multiple end-points visible in the RTCP reporting and any
feedback events. Thus becoming yet another situation in addition to
Mesh where the end-point will have to have logic for merging
different requirements and preferences. This is more detail
discussed in Section 12.7.
+---+ +---+ +---+
| A |--->| B |--->| C |
+---+ +---+ +---+
Figure 4: MediaStream Forwarding
The above Figure 4 depicts a possible scenario where an WebRTC end-
point (A) sends a media stream to B. B decides to forward the media
stream to C. This can either be realized in B (WebRTC end-point)
using a simple relay functionality creating similar consideration and
implementation requirements. Another implmentation strategy in B
could be to select to transcode the media from A to C, thus breaking
most of the dependecies between A and C. In that case A is not
required to be aware of B forwarding the media to C.
12.4. SSRC Collision Detection
The RTP standard [RFC3550] requires any RTP implementation to have
support for detecting and handling SSRC collisions, i.e. when two
different end-points uses the same SSRC value. This requirement
applies also to WebRTC end-points. There exist several scenarios
where SSRC collisions may occur.
In a point to point session where each SSRC are associated with
either of the two end-points and where the main media carrying SSRC
identifier will be announced in the signalling there is less likely
to occur due to the information about used SSRCs provided by Source-
Specific SDP Attributes [RFC5576]. Still if both end-points starts
uses an new SSRC identifier prior to having signalled it to the peer
and received acknowledgement on the signalling message there can be
collisions. The Source-Specific SDP Attributes [RFC5576] contains no
mechanism to resolve SSRC collisions or reject a end-points usage of
an SSRC.
There could also appear unsignalled SSRCs, this may be considered a
bug. This is more likely than it appears as certain RTP
functionalities need extra SSRCs to provide functionality related to
another SSRC, for example SSRC multiplexed RTP retransmission
[RFC4588]. In those cases an end-point can create a new SSRC which
strictly don't need to be announced over the signalling channel to
function correctly on both RTP and PeerConnection level.
The more likely cases for SSRC collision is that multiple end-points
in an multiparty creates new soruces and signalls those towards the
central server. In cases where the SSRC/CSRC are propogated between
the different end-points from the central node collisions can occur.
Another scenario is when the central node manage to connect an end-
points PeerConnection to another PeerConnectio the end-point it has.
Thus forming a loop where the end-point will receive its own traffic.
This must be considered a bug, but still if it occurs it is important
that the end-point can handle the situation.
12.5. Contributing Sources
Contributing Sources (CSRC) is a functionality in RTP header that
enables a RTP node combing multiple sources into one to identify the
sources that has gone into the combination. For WebRTC end-point the
support of contributing sources are trivial. The set of CSRC are
provided for a given RTP packet. This information can then be
exposed towards the applications using some form of API, most likely
a mapping back into MediaStream identities to avoid having to expose
two namespaces and the handling of SSRC collision handling to the
JavaScript.
There are also at least one extension that is dependent on the CRSRC
list being used, that is the Mixer to client audio level [RFC6465],
that enhances the information provided by the CSRC to actual energy
levels for audio for each contributing source.
12.6. Media Synchronization
When an end-point has more than one media source being sent one need
to consider if these media source are to be synchronized. In RTP/
RTCP synchronziation is provided by having a set of media streams be
indicated as comming from the same synchroniztion context and logical
end-point by using the same CNAME identifier.
The next provision is that all media sources internal clock, i.e.
what drives the RTP timestamp can be correlated with a system clock
that is provided in RTCP Sender Reports encoded in an NTP format. By
having the RTP timestamp to system clock being provided for all
sources the relation of the different media stream, also across
multiple RTP sessions can if chosen to be synchronized. The
requirement is for the media sender to provide the information, the
receiver can chose to use it or not.
12.7. Multiple RTP End-points
A number of usages of RTP discussed here results in that an WebRTC
end-point sending media in an RTP session out over an PeerConnection
will receive receiver reports from multiple RTP receiving nodes.
Note that receiving multiple receiver reports are expected due to
that any RTP node that has multiple SSRCs are required to report on
the media sender. The difference here is that they are multiple
nodes, and thus will have different path characteristics.
The topologies relevant to WebRTC when this can occur are centralized
relay and a end-point forwarding a media stream. Mixers are expected
to not forward media stream reports across itself due to the
difference in the media stream provided to different end-points which
the original media source lacks information about the mixers
manipulation.
Having multiple RTP nodes receive ones RTP flow and send reports and
feedback about it has several impacts. As previously discussed
(Section 12.3) any codec control and rate control needs to be capable
of merging the requirements and preferences to provide a single best
according to the situation media stream. Specifically when it comes
to congestion control it needs to be capable of identifying the
different end-points to form independent congestion state information
for each different path.
Providing source authentication in multi-party is a challange. In
the mixer based topologies an end-points source authentication is
based on verifying that media comes from the mixer by cryptographic
verification and secondly trust the mixer to correctly identify any
source towards the end-point. In RTP sessions where multiple end-
points are directly visible to an end-point all end-points have
knowledge about each others master keys, and can thus inject packets
claimed to come from another end-point in the session. Any node
performing relay can perform non-cryptographic mitigation by
preventing forwarding of packets that has SSRC fields that has
previously come from other end-points. For cryptographic
verification of the source SRTP will require additional security
mechanisms, like TESLA for SRTP [RFC4383].
12.8. Simulcast
This section discusses simulcast in the meaning of providing a node,
for example a Mixer, with multiple different encoded version of the
same media source. In the WebRTC context that appears to be most
easily accomplished by establishing mutliple PeerConnection all being
feed the same set of MediaStreams. Each PeerConnection is then
configured to deliver a particular media quality and thus media bit-
rate. This will work well as long as the end-point implements
indepdentent media encoding for each PeerConnection and not share the
encoder. Simulcast will fail if the end-point uses a common encoder
instance to multiple PeerConnections.
Thus it should be considered to explicitly signal which of the two
implementation strategies that are desired and which will be done.
At least making the application and possible the central node
interested in receiving simulcast of an end-points media streams to
be aware if it will function or not.
12.9. Differentiated Treatment of Flows
There exist use cases for differentiated treatment of media streams.
Such differentiation can happen at several places in the system.
First of all is the prioritization within the end-point for which
media streams that should be sent, there allocation of bit-rate out
of the current available aggregate as determined by the congestion
control.
Secondly, the transport can prioritize a media streams. This is done
according to three methods;
Diffserv: The end-point could mark the packet with a diffserv code
point to indicate to the network how the WebRTC application and
browser would like this particular packet treated.
Flow based: Prioritization of all packets belonging to a particular
media flow or RTP session by keeping them in separated UDP flows.
Thus enabling either end-point initiated or network initiated
prioritization of the flow.
Deep Packet Inspection: A network classifier (DPI) inspects the
packet and tries to determine if the packet represents a
particular application and type that is to be prioritized.
With the exception of diffserv both flow based and DPI have issues
with running multiple media types and flows on a single UDP flow,
especially when combined with data transport (SCTP/DTLS). DPI has
issues due to that multiple different type of flows are aggregated
and thus becomes more difficult to apply analysis on. The flow based
differentiation will provide the same treatment to all packets within
the flow. Thus relative prioritization is not possible. In addition
if the resources are limited it may not be possible to provide
differential treatment compared to best-effort for all the flows in a
WebRTC application.
When flow based differentiation is available the WebRTC application
needs to know about so that it can provide the separation of the
media streams onto different UDP flows to enable a more granular
usage of flow based differentiation.
Diffserv is based on that either the end-point or a classifier can
mark the packets with an appropriate DSCP so the packets is treated
according to that marking. If the end-point is to mark the traffic
there exist two requirements in the WebRTC context. The first is
that the WebRTC application or browser knows which DSCP to use and
that it can use them on some set of media streams. Secondly the
information needs to be propagated to the operating system when
transmitting the packet.
Open Issue: How will the WebRTC application and/or browser know that
differentiated treatment is desired and available and ensure that it
gets the information required to correctly configure the WebRTC
multimedia conference.
13. IANA Considerations
This memo makes no request of IANA. This memo makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. RFC.
10. Security Considerations 14. Security Considerations
RTP and its various extensions each have their own security RTP and its various extensions each have their own security
considerations. These should be taken into account when considering considerations. These should be taken into account when considering
the security properties of the complete suite. We currently don't the security properties of the complete suite. We currently don't
think this suite creates any additional security issues or think this suite creates any additional security issues or
properties. The use of SRTP [RFC3711] will provide protection or properties. The use of SRTP [RFC3711] will provide protection or
mitigation against all the fundamental issues by offering mitigation against all the fundamental issues by offering
confidentiality, integrity and partial source authentication. A confidentiality, integrity and partial source authentication. A
mandatory to implement media security solution will be required to be mandatory to implement media security solution will be required to be
picked. We currently don't discuss the key-management aspect of SRTP picked. We currently don't discuss the key-management aspect of SRTP
in this memo, that needs to be done taking the WebRTC communication in this memo, that needs to be done taking the WebRTC communication
model into account. model into account.
The guidelines in [I-D.ietf-avtcore-srtp-vbr-audio] apply when using The guidelines in [I-D.ietf-avtcore-srtp-vbr-audio] apply when using
variable bit rate (VBR) audio codecs, for example Opus or the Mixer variable bit rate (VBR) audio codecs, for example Opus or the Mixer
audio level header extensions. audio level header extensions.
Security considerations for the WebRTC work are discussed in Security considerations for the WebRTC work are discussed in
[I-D.ietf-rtcweb-security]. [I-D.ietf-rtcweb-security].
11. Acknowledgements 15. Acknowledgements
The authors would like to thank Harald Alvestrand, Cary Bran, and The authors would like to thank Harald Alvestrand, Cary Bran, Charles
Cullen Jennings for valuable feedback. Eckel and Cullen Jennings for valuable feedback.
12. References 16. References
12.1. Normative References 16.1. Normative References
[I-D.holmberg-mmusic-sdp-bundle-negotiation] [I-D.holmberg-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers", Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011. progress), October 2011.
[I-D.ietf-avtcore-srtp-encrypted-header-ext] [I-D.ietf-avtcore-srtp-encrypted-header-ext]
Lennox, J., "Encryption of Header Extensions in the Secure Lennox, J., "Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)", Real-Time Transport Protocol (SRTP)",
draft-ietf-avtcore-srtp-encrypted-header-ext-00 (work in draft-ietf-avtcore-srtp-encrypted-header-ext-01 (work in
progress), June 2011. progress), October 2011.
[I-D.ietf-avtcore-srtp-vbr-audio] [I-D.ietf-avtcore-srtp-vbr-audio]
Perkins, C. and J. Valin, "Guidelines for the use of Perkins, C. and J. Valin, "Guidelines for the use of
Variable Bit Rate Audio with Secure RTP", Variable Bit Rate Audio with Secure RTP",
draft-ietf-avtcore-srtp-vbr-audio-03 (work in progress), draft-ietf-avtcore-srtp-vbr-audio-04 (work in progress),
July 2011. December 2011.
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower- Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-02 (work based Applications", draft-ietf-rtcweb-overview-03 (work
in progress), September 2011. in progress), March 2012.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web", Rescorla, E., "Security Considerations for RTC-Web",
draft-ietf-rtcweb-security-01 (work in progress), draft-ietf-rtcweb-security-02 (work in progress),
October 2011. March 2012.
[I-D.jesup-rtp-congestion-reqs] [I-D.jesup-rtp-congestion-reqs]
Jesup, R. and H. Alvestrand, "Congestion Control Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media", Requirements For Real Time Media",
draft-jesup-rtp-congestion-reqs-00 (work in progress), draft-jesup-rtp-congestion-reqs-00 (work in progress),
March 2012. March 2012.
[I-D.lennox-rtcweb-rtp-media-type-mux] [I-D.lennox-rtcweb-rtp-media-type-mux]
Lennox, J. and J. Rosenberg, "Multiplexing Multiple Media Rosenberg, J. and J. Lennox, "Multiplexing Multiple Media
Types In a Single Real-Time Transport Protocol (RTP) Types In a Single Real-Time Transport Protocol (RTP)
Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work
in progress), October 2011. in progress), October 2011.
[I-D.perkins-avtcore-rtp-circuit-breakers] [I-D.perkins-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "RTP Congestion Control: Circuit Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
Breakers for Unicast Sessions", Breakers for Unicast Sessions",
draft-perkins-avtcore-rtp-circuit-breakers-00 (work in draft-perkins-avtcore-rtp-circuit-breakers-00 (work in
progress), March 2012. progress), March 2012.
[I-D.westerlund-avtcore-multiplex-architecture] [I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., and C. Perkins, "RTP Westerlund, M., Burman, B., and C. Perkins, "RTP
Multiplexing Architecture", Multiplexing Architecture",
draft-westerlund-avtcore-multiplex-architecture-00 (work draft-westerlund-avtcore-multiplex-architecture-01 (work
in progress), October 2011. in progress), March 2012.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
Single Lower-Layer Transport",
draft-westerlund-avtcore-transport-multiplexing-02 (work
in progress), March 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736, Payload Format Specifications", BCP 36, RFC 2736,
December 1999. December 1999.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003. Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
skipping to change at page 25, line 36 skipping to change at page 35, line 5
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008. Header Extensions", RFC 5285, July 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, April 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010. Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010. Flows", RFC 6051, November 2010.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for [RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, April 2011. (CNAMEs)", RFC 6222, April 2011.
[RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time [RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to- Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, December 2011. Mixer Audio Level Indication", RFC 6464, December 2011.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to- Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 2011. Client Audio Level Indication", RFC 6465, December 2011.
12.2. Informative References 16.2. Informative References
[I-D.alvestrand-rtcweb-msid]
Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol",
draft-alvestrand-rtcweb-msid-02 (work in progress),
May 2012.
[I-D.begen-mmusic-redundancy-grouping] [I-D.begen-mmusic-redundancy-grouping]
Begen, A., Cai, Y., and H. Ou, "Duplication Grouping Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
Semantics in the Session Description Protocol", Semantics in the Session Description Protocol",
draft-begen-mmusic-redundancy-grouping-01 (work in draft-begen-mmusic-redundancy-grouping-03 (work in
progress), June 2011. progress), March 2012.
[I-D.cbran-rtcweb-data] [I-D.cbran-rtcweb-data]
Bran, C. and C. Jennings, "RTC-Web Non-Media Data Bran, C. and C. Jennings, "RTC-Web Non-Media Data
Transport Requirements", draft-cbran-rtcweb-data-00 (work Transport Requirements", draft-cbran-rtcweb-data-00 (work
in progress), July 2011. in progress), July 2011.
[I-D.ietf-avt-srtp-ekt]
Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
(work in progress), October 2011.
[I-D.ietf-fecframe-framework] [I-D.ietf-fecframe-framework]
Watson, M., Begen, A., and V. Roca, "Forward Error Watson, M., Begen, A., and V. Roca, "Forward Error
Correction (FEC) Framework", Correction (FEC) Framework",
draft-ietf-fecframe-framework-15 (work in progress), draft-ietf-fecframe-framework-15 (work in progress),
June 2011. June 2011.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997. September 1997.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006. Congestion Control", RFC 4341, March 2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
March 2006. March 2006.
[RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient
Stream Loss-Tolerant Authentication (TESLA) in the Secure
Real-time Transport Protocol (SRTP)", RFC 4383,
February 2006.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control [RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828, (TFRC): The Small-Packet (SP) Variant", RFC 4828,
April 2007. April 2007.
[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, [RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
"RTP Payload Format and File Storage Format for the "RTP Payload Format and File Storage Format for the
Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
(AMR-WB) Audio Codecs", RFC 4867, April 2007. (AMR-WB) Audio Codecs", RFC 4867, April 2007.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008. January 2008.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", Friendly Rate Control (TFRC): Protocol Specification",
RFC 5348, September 2008. RFC 5348, September 2008.
[RFC5404] Westerlund, M. and I. Johansson, "RTP Payload Format for [RFC5404] Westerlund, M. and I. Johansson, "RTP Payload Format for
G.719", RFC 5404, January 2009. G.719", RFC 5404, January 2009.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009. Control", RFC 5681, September 2009.
[RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP
Control Protocol (RTCP)", RFC 5968, September 2010. Control Protocol (RTCP)", RFC 5968, September 2010.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, June 2011. Control Protocol (RTCP) Flows", RFC 6263, June 2011.
Appendix A. Supported RTP Topologies
RTP supports both unicast and group communication, with participants
being connected using wide range of transport-layer topologies. Some
of these topologies involve only the end-points, while others use RTP
translators and mixers to provide in-network processing. Properties
of some RTP topologies are discussed in [RFC5117], and we further
describe those expected to be useful for WebRTC in the following. We
also goes into important RTP session aspects that the topology or
implementation variant can place on a WebRTC end-point.
A.1. Point to Point
The point-to-point RTP topology (Figure 5) is the simplest scenario
for WebRTC applications. This is going to be very common for user to
user calls.
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 5: Point to Point
This being the basic one lets use the topology to high-light a couple
of details that are common for all RTP usage in the WebRTC context.
First is the intention to multiplex RTP and RTCP over the same UDP-
flow. Secondly is the question of using only a single RTP session or
one per media type for legacy interoperability. Thirdly is the
question of using multiple sender sources (SSRCs) per end-point.
Historically, RTP and RTCP have been run on separate UDP ports. With
the increased use of Network Address/Port Translation (NAPT) this has
become problematic, since maintaining multiple NAT bindings can be
costly. It also complicates firewall administration, since multiple
ports must be opened to allow RTP traffic. To reduce these costs and
session setup times, support for multiplexing RTP data packets and
RTCP control packets on a single port [RFC5761] will be supported.
In cases where there is only one type of media (e.g., a voice-only
call) this topology will be implemented as a single RTP session, with
bidirectional flows of RTP and RTCP packets, all then multiplexed
onto a single 5-tuple. If multiple types of media are to be used
(e.g., audio and video), then each type media can be sent as a
separate RTP session using a different 5-tuple, allowing for separate
transport level treatment of each type of media. Alternatively, all
types of media can be multiplexed onto a single 5-tuple as a single
RTP session, or as several RTP sessions if using a demultiplexing
shim. Multiplexing different types of media onto a single 5-tuple
places some limitations on how RTP is used, as described in "RTP
Multiplexing Architecture"
[I-D.westerlund-avtcore-multiplex-architecture]. It is not expected
that these limitations will significantly affect the scenarios
targetted by WebRTC, but they may impact interoperability with legacy
systems.
An RTP session have good support for simultanously transport multiple
media sources. Each media source uses an unique SSRC identifier and
each SSRC has independent RTP sequence number and timestamp spaces.
This is being utilized in WebRTC for several cases. One is to enable
multiple media sources of the same type, an end-point that has two
video cameras can potentially transmitt video from both to its
peer(s). Another usage is when a single RTP session is being used
for both multiple media types, thus an end-point can transmit both
audio and video to the peer(s). Thirdly to support multi-party cases
as will be discussed below support for multiple SSRC of the same
media type are required.
Thus we can introduce a couple of different notiations in the below
two alternate figures of a single peer connection in a a point to
point setup. The first depicting a setup where the peer connection
established has two different RTP sessions, one for audio and one for
video. The second one using a single RTP session. In both cases A
has two video streams to send and one audio stream. B has only one
audio and video stream. These are used to illustrate the relation
between a peerConnection, the UDP flow(s), the RTP session(s) and the
SSRCs that will be used in the later cases also. In the below
figures RTCP flows are not included. They will flow bi-directionally
between any RTP session instances in the different nodes.
+-A-------------+ +-B-------------+
| +-PeerC1------| |-PeerC1------+ |
| | +-UDP1------| |-UDP1------+ | |
| | | +-RTP1----| |-RTP1----+ | | |
| | | | +-Audio-| |-Audio-+ | | | |
| | | | | AA1|---------------->| | | | | |
| | | | | |<----------------|BA1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| | | | | |
| | +-UDP2------| |-UDP2------+ | |
| | | +-RTP2----| |-RTP1----+ | | |
| | | | +-Video-| |-Video-+ | | | |
| | | | | AV1|---------------->| | | | | |
| | | | | AV2|---------------->| | | | | |
| | | | | |<----------------|BV1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| +-------------| |-------------+ |
+---------------+ +---------------+
Figure 6: Point to Point: Multiple RTP sessions
As can be seen above in the Point to Point: Multiple RTP sessions
(Figure 6) the single Peer Connection contains two RTP sessions over
different UDP flows UDP 1 and UDP 2, i.e. their 5-tuples will be
different, normally on source and destination ports. The first RTP
session (RTP1) carries audio, one stream in each direction AA1 and
BA1. The second RTP session contains two video streams from A (AV1
and AV2) and one from B to A (BV1).
+-A-------------+ +-B-------------+
| +-PeerC1------| |-PeerC1------+ |
| | +-UDP1------| |-UDP1------+ | |
| | | +-RTP1----| |-RTP1----+ | | |
| | | | +-Audio-| |-Audio-+ | | | |
| | | | | AA1|---------------->| | | | | |
| | | | | |<----------------|BA1 | | | | |
| | | | +-------| |-------+ | | | |
| | | | | | | | | |
| | | | +-Video-| |-Video-+ | | | |
| | | | | AV1|---------------->| | | | | |
| | | | | AV2|---------------->| | | | | |
| | | | | |<----------------|BV1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| +-------------| |-------------+ |
+---------------+ +---------------+
Figure 7: Point to Point: Single RTP session.
In (Figure 7) there is only a single UDP flow and RTP session (RTP1).
This RTP session carries a total of five (5) media streams (SSRCs).
From A to B there is Audio (AA1) and two video (AV1 and AV2). From B
to A there is Audio (BA1) and Video (BV1).
A.2. Multi-Unicast (Mesh)
For small multiparty calls, it is practical to set up a multi-unicast
topology (Figure 8); unfortunately not discussed in the RTP
Topologies RFC [RFC5117]. In this topology, each participant sends
individual unicast RTP/UDP/IP flows to each of the other participants
using independent PeerConnections in a full mesh.
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 8: Multi-unicast
This topology has the benefit of not requiring central nodes. The
downside is that it increases the used bandwidth at each sender by
requiring one copy of the media streams for each participant that are
part of the same session beyond the sender itself. Hence, this
topology is limited to scenarios with few participants unless the
media is very low bandwidth. The multi-unicast topology could be
implemented as a single RTP session, spanning multiple peer-to-peer
transport layer connections, or as several pairwise RTP sessions, one
between each pair of peers. To maintain a coherent mapping between
the relation between RTP sessions and PeerConnections we recommend
that one implements this as individual RTP sessions. The only
downside is that end-point A will not learn of the quality of any
transmission happening between B and C based on RTCP. This has not
been seen as a significant downside as now one has yet seen a need
for why A would need to know about the B's and C's communication. An
advantage of using separate RTP sessions is that it enables using
different media bit-rates to the differnt peers, thus not forcing B
to endure the same quality reductions if there are limiations in the
transport from A to C as C will.
+-A------------------------+ +-B-------------+
|+---+ +-PeerC1------| |-PeerC1------+ |
||MIC| | +-UDP1------| |-UDP1------+ | |
|+---+ | | +-RTP1----| |-RTP1----+ | | |
| | +----+ | | | +-Audio-| |-Audio-+ | | | |
| +->|ENC1|--+-+-+-+--->AA1|------------->| | | | | |
| | +----+ | | | | |<-------------|BA1 | | | | |
| | | | | +-------| |-------+ | | | |
| | | | +---------| |---------+ | | |
| | | +-----------| |-----------+ | |
| | +-------------| |-------------+ |
| | | |---------------+
| | |
| | | +-C-------------+
| | +-PeerC2------| |-PeerC2------+ |
| | | +-UDP2------| |-UDP2------+ | |
| | | | +-RTP2----| |-RTP2----+ | | |
| | +----+ | | | +-Audio-| |-Audio-+ | | | |
| +->|ENC2|--+-+-+-+--->AA2|------------->| | | | | |
| +----+ | | | | |<-------------|CA1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| +-------------| |-------------+ |
+--------------------------+ +---------------+
Figure 9: Session strcuture for Multi-Unicast Setup
Lets review how the RTP sessions looks from A's perspective by
considering both how the media is a handled and what PeerConnections
and RTP sessions that are setup in Figure 9. A's microphone is
captured and the digital audio can then be feed into two different
encoder instances each beeing associated with two different
PeerConnections (PeerC1 and PeerC2) each containing independent RTP
sessions (RTP1 and RTP2). The SSRCs in each RTP session will be
completely independent and the media bit-rate produced by the encoder
can also be tuned to address any congestion control requirements
between A and B differently then for the path A to C.
For media encodings which are more resource consuming, like video,
one could expect that it will be common that end-points that are
resource costrained will use a different implementation strategy
where the encoder is shared between the different PeerConnections as
shown below Figure 10.
+-A----------------------+ +-B-------------+
|+---+ | | |
||CAM| +-PeerC1------| |-PeerC1------+ |
|+---+ | +-UDP1------| |-UDP1------+ | |
| | | | +-RTP1----| |-RTP1----+ | | |
| V | | | +-Video-| |-Video-+ | | | |
|+----+ | | | | |<----------------|BV1 | | | | |
||ENC |----+-+-+-+--->AV1|---------------->| | | | | |
|+----+ | | | +-------| |-------+ | | | |
| | | | +---------| |---------+ | | |
| | | +-----------| |-----------+ | |
| | +-------------| |-------------+ |
| | | |---------------+
| | |
| | | +-C-------------+
| | +-PeerC2------| |-PeerC2------+ |
| | | +-UDP2------| |-UDP2------+ | |
| | | | +-RTP2----| |-RTP2----+ | | |
| | | | | +-Video-| |-Video-+ | | | |
| +-------+-+-+-+--->AV2|---------------->| | | | | |
| | | | | |<----------------|CV1 | | | | |
| | | | +-------| |-------+ | | | |
| | | +---------| |---------+ | | |
| | +-----------| |-----------+ | |
| +-------------| |-------------+ |
+------------------------+ +---------------+
Figure 10: Single Encoder Multi-Unicast Setup
This will clearly save resources consumed by encoding but does
introduce the need for the end-point A to make decisions on how it
encodes the media so it suites delivery to both B and C. This is not
limited to congestion control, also prefered resolution to receive
based on dispaly area available is another aspect requiring
consideration. The need for this type of descion logic does arise in
several different topologies and implementation.
A.3. Mixer Based
An mixer (Figure 11) is a centralised point that selects or mixes
content in a conference to optimise the RTP session so that each end-
point only needs connect to one entity, the mixer. The mixer can
also reduce the bit-rate needed from the mixer down to a conference
participants as the media sent from the mixer to the end-point can be
optimised in different ways. These optimisations include methods
like only choosing media from the currently most active speaker or
mixing together audio so that only one audio stream is required in
stead of 3 in the depicted scenario (Figure 11).
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 11: RTP Mixer with Only Unicast Paths
Mixers has two downsides, the first is that the mixer must be a
trusted node as they either performs media operations or at least
repacketize the media. Both type of operations requires when using
SRTP that the mixer verifies integrity, decrypts the content, perform
its operation and form new RTP packets, encrypts and integegrity
protect them. This applies to all types of mixers described below.
The second downside is that all these operations and optimization of
the session requires processing. How much depends on the
implementation as will become evident below.
The implementation of an mixer can take several different forms and
we will discuss the main themes available that doesn't break RTP.
Please note that a Mixer could also contain translator
functionalities, like a media transcoder to adjust the media bit-rate
or codec used on a particular media stream.
A.3.1. Media Mixing
This type of mixer is one which clearly can be called RTP mixer is
likely the one that most thinks of when they hear the term mixer.
Its basic patter of operation is that it will receive the different
participants media stream. Select which that are to be included in a
media domain mix of the incomming media streams. Then create a
single outgoing stream from this mix.
Audio mixing is straight forward and commonly possible to do for a
number of participants. Lets assume that you want to mix N number of
streams from different participants. Then the mixer need to perform
N decodings. Then it needs to produce N or N+1 mixes, the reasons
that different mixes are needed are so that each contributing source
get a mix which don't contain themselves, as this would result in an
echo. When N is lower than the number of all participants one may
produce a Mix of all N streams for the group that are curently not
included in the mix, thus N+1 mixes. These audio streams are then
encoded again, RTP packetized and sent out.
Video can't really be "mixed" and produce something particular useful
for the users, however creating an composition out of the contributed
video streams can be done. In fact it can be done in a number of
ways, tiling the different streams creating a chessboard, selecting
someone as more important and showing them large and a number of
other sources as smaller is another. Also here one commonly need to
produce a number of different compositions so that the contributing
part doesn't need to see themselves. Then the mixer re-encodes the
created video stream, RTP packetize it and send it out
The problem with media mixing is that it both consume large amount of
media processing and encoding resources. The second is the quality
degradation created by decoding and re-encoding the media stream.
Its advantage is that it is quite simplistic for the clients to
handle as they don't need to handle local mixing and composition.
+-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ |
| | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | AA1|------------>|---------+-+-+-+-|DEC|->| | |
| | | | | |<------------|MA1 <----+ | | | +---+ | | |
| | | | | | |(BA1+CA1)|\| | | +---+ | | |
| | | | +-------| |---------+ +-+-+-|ENC|<-| B+C | |
| | | +---------| |-----------+ | | +---+ | | |
| | +-----------| |-------------+ | | M | |
| +-------------| |---------------+ | E | |
+---------------+ | | D | |
| | I | |
+-B-------------+ | | A | |
| +-PeerC2------| |-PeerC2--------+ | | |
| | +-UDP2------| |-UDP2--------+ | | M | |
| | | +-RTP2----| |-RTP2------+ | | | I | |
| | | | +-Audio-| |-Audio---+ | | | +---+ | X | |
| | | | | BA1|------------>|---------+-+-+-+-|DEC|->| E | |
| | | | | |<------------|MA2 <----+ | | | +---+ | R | |
| | | | +-------| |(BA1+CA1)|\| | | +---+ | | |
| | | +---------| |---------+ +-+-+-|ENC|<-| A+C | |
| | +-----------| |-----------+ | | +---+ | | |
| +-------------| |-------------+ | | | |
+---------------+ |---------------+ | | |
| | | |
+-C-------------+ | | | |
| +-PeerC3------| |-PeerC3--------+ | | |
| | +-UDP3------| |-UDP3--------+ | | | |
| | | +-RTP3----| |-RTP3------+ | | | | |
| | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | CA1|------------>|---------+-+-+-+-|DEC|->| | |
| | | | | |<------------|MA3 <----+ | | | +---+ | | |
| | | | +-------| |(BA1+CA1)|\| | | +---+ | | |
| | | +---------| |---------+ +-+-+-|ENC|<-| A+B | |
| | +-----------| |-----------+ | | +---+ | | |
| +-------------| |-------------+ | +-----+ |
+---------------+ |---------------+ |
+--------------------------------+
Figure 12: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be very straight forward as
can be seen in Figure 12. The mixer present one SSRC towards the
peer client, e.g. MA1 to Peer A, which is the media mix of the other
particpants. As each peer receives a different version produced by
the mixer there are no actual relation between the different RTP
sessions in the actual media or the transport level information.
There is however one connection between RTP1-RTP3 in this figure. It
has to do with the SSRC space and the identity information. When A
receives the MA1 stream which is a combination of BA1 and CA1 streams
in the other PeerConnections RTP could enable the mixer to include
CSRC information in the MA1 stream to identify the contributing
source BA1 and CA1.
The CSRC has in its turn utility in RTP extensions, like the in
Section 5.2.3 discussed Mixer to Client audio levels RTP header
extension [RFC6465]. If the SSRC from one PeerConnection are used as
CSRC in another PeerConnection then RTP1, RTP2 and RTP3 becomes one
joint session as they have a common SSRC space. At this stage one
also need to consider which RTCP information one need to expose in
the different legs. For the above situation commonly nothing more
than the Source Description (SDES) information and RTCP BYE for CSRC
need to be exposed. The main goal would be to enable the correct
binding against the application logic and other information sources.
This also enables loop detection in the RTP session.
A.3.1.1. RTP Session Termination
There exist an possible implementation choice to have the RTP
sessions being separated between the different legs in the multi-
party communication session and only generate media streams in each
without carrying on RTP/RTCP level any identity information about the
contributing sources. This removes both the functionaltiy that CSRC
can provide and the possibility to use any extensions that build on
CSRC and the loop detection. It may appear a simplification if SSRC
collision would occur between two different end-points as they can be
avoide to be resolved and instead remapped between the independent
sessions if at all exposed. However, SSRC/CSRC remapping
requiresthat SSRC/CSRC are never exposed to the WebRTC javascript
client to use as reference. This as they only have local importance
if they are used on a multi-party session scope the result would be
missreferencing. Also SSRC collision handling will still be needed
as it may occur between the mixer and the end-point.
Session termination may appear to resolve some issues, it however
creates other issues that needs resolving, like loop detection,
identification of contributing sources and the need to handle mapped
identities and ensure that the right one is used towards the right
identities and never used directly between multiple end-points.
A.3.2. Media Switching
An RTP Mixer based on media switching avoids the media decoding and
encoding cycle in the mixer, but not the decryption and re-encryption
cycle as one rewrites RTP headers. This both reduces the amount of
computational resources needed in the mixer and increases the media
quality per transmitted bit. This is achieve by letting the mixer
have a number of SSRCs that represents conceptual or functional
streams the mixer produces. These streams are created by selecting
media from one of the by the mixer received media streams and forward
the media using the mixers own SSRCs. The mixer can then switch
between available sources if that is required by the concept for the
source, like currently active speaker.
To achieve a coherent RTP media stream from the mixer's SSRC the
mixer is forced to rewrite the incoming RTP packet's header. First
the SSRC field must be set to the value of the Mixer's SSRC.
Secondly, the sequence number must be the next in the sequence of
outgoing packets it sent. Thirdly the RTP timestamp value needs to
be adjusted using an offset that changes each time one switch media
source. Finally depending on the negotiation the RTP payload type
value representing this particular RTP payload configuration may have
to be changed if the different PeerConnections have not arrived on
the same numbering for a given configuration. This also requires
that the different end-points do support a common set of codecs,
otherwise media transcoding for codec compatibility is still
required.
Lets consider the operation of media switching mixer that supports a
video conference with six participants (A-F) where the two latest
speakers in the conference are shown to each participants. Thus the
mixer has two SSRCs sending video to each peer.
+-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ |
| | | | +-Video-| |-Video---+ | | | | | |
| | | | | AV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|MV1 <----+-+-+-+-BV1----| | |
| | | | | |<------------|MV2 <----+-+-+-+-EV1----| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | | S | |
| +-------------| |---------------+ | W | |
+---------------+ | | I | |
| | T | |
+-B-------------+ | | C | |
| +-PeerC2------| |-PeerC2--------+ | H | |
| | +-UDP2------| |-UDP2--------+ | | | |
| | | +-RTP2----| |-RTP2------+ | | | M | |
| | | | +-Video-| |-Video---+ | | | | A | |
| | | | | BV1|------------>|---------+-+-+-+------->| T | |
| | | | | |<------------|MV3 <----+-+-+-+-AV1----| R | |
| | | | | |<------------|MV4 <----+-+-+-+-EV1----| I | |
| | | | +-------| |---------+ | | | | X | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | | | |
| +-------------| |---------------+ | | |
+---------------+ | | | |
: : : :
: : : :
+-F-------------+ | | | |
| +-PeerC6------| |-PeerC6--------+ | | |
| | +-UDP6------| |-UDP6--------+ | | | |
| | | +-RTP6----| |-RTP6------+ | | | | |
| | | | +-Video-| |-Video---+ | | | | | |
| | | | | CV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|MV11 <---+-+-+-+-AV1----| | |
| | | | | |<------------|MV12 <---+-+-+-+-EV1----| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | +-----+ |
| +-------------| |---------------+ |
+---------------+ +--------------------------------+
Figure 13: Media Switching RTP Mixer
The Media Switching RTP mixer can similar to the Media Mixing one
reduce the bit-rate needed towards the different peers by selecting
and switching in a sub-set of media streams out of the ones it
receives from the conference participations.
To ensure that a media receiver can correctly decode the media stream
after a switch, it becomes necessary to ensure for state saving
codecs that they start from default state at the point of switching.
Thus one common tool for video is to request that the encoding
creates an intra picture, something that isn't dependent on earlier
state. This can be done using Full Intra Request RTCP codec control
message as discussed in Section 5.1.1.
Also in this type of mixer one could consider to terminate the RTP
sessions fully between the different PeerConnection. The same
arguments and conisderations as discussed in Appendix A.3.1.1 applies
here.
A.3.3. Media Projecting
Another method for handling media in the RTP mixer is to project all
potential sources (SSRCs) into a per end-point independent RTP
session. The mixer can then select which of the potential sources
that are currently actively transmitting media, despite that the
mixer in another RTP session recieves media from that end-point.
This is similar to the media switching Mixer but have some important
differences in RTP details.
+-A-------------+ +-MIXER--------------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | +-----+ |
| | | | +-Video-| |-Video---+ | | | | | |
| | | | | AV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|BV1 <----+-+-+-+--------| | |
| | | | | |<------------|CV1 <----+-+-+-+--------| | |
| | | | | |<------------|DV1 <----+-+-+-+--------| | |
| | | | | |<------------|EV1 <----+-+-+-+--------| | |
| | | | | |<------------|FV1 <----+-+-+-+--------| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | | S | |
| +-------------| |---------------+ | W | |
+---------------+ | | I | |
| | T | |
+-B-------------+ | | C | |
| +-PeerC2------| |-PeerC2--------+ | H | |
| | +-UDP2------| |-UDP2--------+ | | | |
| | | +-RTP2----| |-RTP2------+ | | | M | |
| | | | +-Video-| |-Video---+ | | | | A | |
| | | | | BV1|------------>|---------+-+-+-+------->| T | |
| | | | | |<------------|AV1 <----+-+-+-+--------| R | |
| | | | | |<------------|CV1 <----+-+-+-+--------| I | |
| | | | | | : : : |: : : : : : : : : : :| X | |
| | | | | |<------------|FV1 <----+-+-+-+--------| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | | | |
| +-------------| |---------------+ | | |
+---------------+ | | | |
: : : :
: : : :
+-F-------------+ | | | |
| +-PeerC6------| |-PeerC6--------+ | | |
| | +-UDP6------| |-UDP6--------+ | | | |
| | | +-RTP6----| |-RTP6------+ | | | | |
| | | | +-Video-| |-Video---+ | | | | | |
| | | | | CV1|------------>|---------+-+-+-+------->| | |
| | | | | |<------------|AV1 <----+-+-+-+--------| | |
| | | | | | : : : |: : : : : : : : : : :| | |
| | | | | |<------------|EV1 <----+-+-+-+--------| | |
| | | | +-------| |---------+ | | | | | |
| | | +---------| |-----------+ | | | | |
| | +-----------| |-------------+ | +-----+ |
| +-------------| |---------------+ |
+---------------+ +--------------------------------+
Figure 14: Media Projecting Mixer
So in this six participant conference depicted above in (Figure 14)
one can see that end-point A will in this case be aware of 5 incoming
SSRCs, BV1-FV1. If this mixer intend to have the same behavior as in
Appendix A.3.2 where the mixer provides the end-points with the two
latest speaking end-points, then only two out of these five SSRCs
will concurrently transmitt media to A. As the mixer selects which
source in the different RTP sessions that transmit media to the end-
points each media stream will require some rewriting when being
projected from one session into another. The main thing is that the
sequence number will need to be consequitvely incremented based on
the packet actually being transmitted in each RTP session. Thus the
RTP sequence number offset will change each time a source is turned
on in RTP session.
As the RTP sessions are independent the SSRC numbers used can be
handled indepdentently also thus working around any SSRC collisions
by having remapping tables between the RTP sessions. However the
related MediaStream signalling must be correspondlingly changed to
ensure consistent MediaStream to SSRC mappings between the different
PeerConnections and the same comment that higher functions must not
use SSRC as references to media streams applies also here.
The mixer will also be responsible to act on any RTCP codec control
requests comming from an end-point and decide if it can act on it
locally or needs to translate the request into the RTP session that
contains the media source. Both end-points and the mixer will need
to implement conference related codec control functionalities to
provide a good experience. Full Intra Request to request from the
media source to provide switching points between the sources,
Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
to aggregate congestion control response towards the media source and
have it adjust its bit-rate in case the limitation is not in the
source to mixer link.
This version of the mixer also puts different requirements on the
end-point when it comes to decoder instances and handling of the
media streams providing media. As each projected SSRC can at any
time provide media the end-point either needs to handle having thus
many allocated decoder instances or have efficient switching of
decoder contexts in a more limited set of actual decoder instances to
cope with the switches. The WebRTC application also gets more
responsibility to update how the media provides is to be presented to
the user.
A.4. Translator Based
There is also a variety of translators. The core commonality is that
they do not need to make themselves visible in the RTP level by
having an SSRC themselves. Instead they sit between one or more end-
point and perform translation at some level. It can be media
transcoding, protocol translation or covering missing functionality
for a legacy device or simply relay packets between transport domains
or to realize multi-party. We will go in details below.
A.4.1. Transcoder
A transcoder operates on media level and really used for two
purposes, the first is to allow two end-points that doesn't have a
common set of media codecs to communicate by translating from one
codec to another. The second is to change the bit-rate to a lower
one. For WebRTC end-points communicating with each other only the
first one should at all be relevant. In certain legacy deployment
media transcoder will be necessary to ensure both codecs and bit-rate
falls within the envelope the legacy device supports.
As transcoding requires access to the media the transcoder must
within the security context and access any media encryption and
integrity keys. On the RTP plane a media transcoder will in practice
fork the RTP session into two different domains that are highly
decoupled when it comes to media parameters and reporting, but not
identities. To maintain signalling bindings to SSRCs a transcoder is
likely needing to use the SSRC of one end-point to represent the
transcoded media stream to the other end-point(s). The congestion
control loop can be terminated in the transcoder as the media bit-
rate being sent by the transcoder can be adjusted independently of
the incoming bit-rate. However, for optimizing performance and
resource consumption the translator needs to consider what signals or
bit-rate reductions it should send towards the source end-point. For
example receving a 2.5 mbps video stream and then send out a 250 kbps
video stream after transcoding is a vaste of resources. In most
cases a 500 kbps video stream from the source in the right resolution
is likely to provide equal quality after transcoding as the 2.5 mbps
source stream. At the same time increasing media bit-rate futher
than what is needed to represent the incoming quality accurate is
also wasted resources.
+-A-------------+ +-Translator------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1------+ | | |
| | | | +-Audio-| |-Audio---+ | | | +---+ |
| | | | | AA1|------------>|---------+-+-+-+-|DEC|----+ |
| | | | | |<------------|BA1 <----+ | | | +---+ | |
| | | | | | | |\| | | +---+ | |
| | | | +-------| |---------+ +-+-+-|ENC|<-+ | |
| | | +---------| |-----------+ | | +---+ | | |
| | +-----------| |-------------+ | | | |
| +-------------| |---------------+ | | |
+---------------+ | | | |
| | | |
+-B-------------+ | | | |
| +-PeerC2------| |-PeerC2--------+ | | |
| | +-UDP2------| |-UDP2--------+ | | | |
| | | +-RTP1----| |-RTP1------+ | | | | |
| | | | +-Audio-| |-Audio---+ | | | +---+ | | |
| | | | | BA1|------------>|---------+-+-+-+-|DEC|--+ | |
| | | | | |<------------|AA1 <----+ | | | +---+ | |
| | | | | | | |\| | | +---+ | |
| | | | +-------| |---------+ +-+-+-|ENC|<---+ |
| | | +---------| |-----------+ | | +---+ |
| | +-----------| |-------------+ | |
| +-------------| |---------------+ |
+---------------+ +-----------------------------+
Figure 15: Media Transcoder
Figure 15 exposes some important details. First of all you can see
the SSRC identifiers used by the translator are the corresponding
end-points. Secondly, there is a relation between the RTP sessions
in the two different PeerConnections that are represtented by having
both parts be identified by the same level and they need to share
certain contexts. Also certain type of RTCP messages will need to be
bridged between the two parts. Certain RTCP feedback messages are
likely needed to be soruced by the translator in response to actions
by the translator and its media encoder.
A.4.2. Gateway / Protocol Translator
Gateways are used when some protocol feature that is required is not
supported by an end-point wants to participate in session. This RTP
translator in Figure 16 takes on the role of ensuring that from the
perspective of participant A, participant B appears as a fully
compliant WebRTC end-point (that is, it is the combination of the
Translator and participant B that looks like a WebRTC end point).
+------------+
| |
+---+ | Translator | +---+
| A |<---->| to legacy |<---->| B |
+---+ | end-point | +---+
WebRTC | | Legacy
+------------+
Figure 16: Gateway (RTP translator) towards legacy end-point
For WebRTC there are a number of requirements that could force the
need for a gateway if a WebRTC end-point is to communicate with a
legacy end-point, such as support of ICE and DTLS-SRTP for
keymanagement. On RTP level the main functions that may be missing
in a legacy implementation that otherswise support RTP are RTCP in
general, SRTP implementation, congestion control and feedback
messages required to make it work.
+-A-------------+ +-Translator------------------+
| +-PeerC1------| |-PeerC1------+ |
| | +-UDP1------| |-UDP1------+ | |
| | | +-RTP1----| |-RTP1-----------------------+|
| | | | +-Audio-| |-Audio---+ ||
| | | | | AA1|------------>|---------+----------------+ ||
| | | | | |<------------|BA1 <----+--------------+ | ||
| | | | | |<---RTCP---->|<--------+----------+ | | ||
| | | | +-------| |---------+ +---+-+ | | ||
| | | +---------| |---------------+| T | | | ||
| | +-----------| |-----------+ | || R | | | ||
| +-------------| |-------------+ || A | | | ||
+---------------+ | || N | | | ||
| || S | | | ||
+-B-(Legacy)----+ | || L | | | ||
| | | || A | | | ||
| +-UDP2------| |-UDP2------+ || T | | | ||
| | +-RTP1----| |-RTP1----------+| E | | | ||
| | | +-Audio-| |-Audio---+ +---+-+ | | ||
| | | | |<---RTCP---->|<--------+----------+ | | ||
| | | | BA1|------------>|---------+--------------+ | ||
| | | | |<------------|AA1 <----+----------------+ ||
| | | +-------| |---------+ ||
| | +---------| |----------------------------+|
| +-----------| |-----------+ |
| | | |
+---------------+ +-----------------------------+
Figure 17: RTP/RTCP Protocol Translator
The legacy gateway may be implemented in several ways and what it
need to change is higly dependent on what functions it need to proxy
for the legacy end-point. One possibility is depicted in Figure 17
where the RTP media streams are compatible and forward without
changes. However, their RTP header values are captured to enable the
RTCP translator to create RTCP reception information related to the
leg between the end-point and the translator. This can then be
combined with the more basic RTCP reports that the legacy endpoint
(B) provides to give compatible and expected RTCP reporting to A.
Thus enabling at least full congestion control on the path between A
and the translator. If B has limited possibilities for congestion
response for the media then the translator may need the capabilities
to perform media transcoding to address cases where it otherwise
would need to terminate media transmission.
As the translator are generating RTP/RTCP traffic on behalf of B to A
it will need to be able to correctly protect these packets that it
translates or generates. Thus security context information are
required in this type of translator if it operates on the RTP/RTCP
packet content or media. In fact one of the more likley scenario is
that the translator (gateway) will need to have two different
security contexts one towards A and one towards B and for each RTP/
RTCP packet do a authenticity verification, decryption followed by a
encryption and integirty protection operation to resolve missmatch in
security systems.
A.4.3. Relay
There exist a class of translators that operates on transport level
below RTP and thus do not effect RTP/RTCP packets directly. They
come in two distinct flavors, the one used to bridge between two
different transport or address domains to more function as a gateway
and the second one which is to to provide a group communication
feature as depicted below in Figure 18.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 18: RTP Translator (Relay) with Only Unicast Paths
The first kind is straight forward and is likely to exist in WebRTC
context when an legacy end-point is compatible with the exception for
ICE, and thus needs a gateway that terminates the ICE and then
forwards all the RTP/RTCP traffic and keymanagment to the end-point
only rewriting the IP/UDP to forward the packet to the legacy node.
The second type is useful if one wants a less complex central node or
a central node that is outside of the security context and thus do
not have access to the media. This relay takes on the role of
forwarding the media (RTP and RTCP) packets to the other end-points
but doesn't perform any RTP or media processing. Such a device
simply forwards the media from each sender to all of the other
particpants, and is sometimes called a transport-layer translator.
In Figure 18, participant A will only need to send a media once to
the relay, which will redistribute it by sending a copy of the stream
to participants B, C, and D. Participant A will still receive three
RTP streams with the media from B, C and D if they transmit
simultaneously. This is from an RTP perspective resulting in an RTP
session that behaves equivalent to one transporter over an IP Any
Source Multicast (ASM).
This results in one common RTP session between all participants
despite that there will be independent PeerConnections created to the
translator as depicted below Figure 19.
+-A-------------+ +-RELAY--------------------------+
| +-PeerC1------| |-PeerC1--------+ |
| | +-UDP1------| |-UDP1--------+ | |
| | | +-RTP1----| |-RTP1-------------------------+ |
| | | | +-Video-| |-Video---+ | |
| | | | | AV1|------------>|---------------------------+ | |
| | | | | |<------------|BV1 <--------------------+ | | |
| | | | | |<------------|CV1 <------------------+ | | | |
| | | | +-------| |---------+ | | | | |
| | | +---------| |-------------------+ ^ ^ V | |
| | +-----------| |-------------+ | | | | | | |
| +-------------| |---------------+ | | | | | |
+---------------+ | | | | | | |
| | | | | | |
+-B-------------+ | | | | | | |
| +-PeerC2------| |-PeerC2--------+ | | | | | |
| | +-UDP2------| |-UDP2--------+ | | | | | | |
| | | +-RTP2----| |-RTP1--------------+ | | | | |
| | | | +-Video-| |-Video---+ | | | | |
| | | | | BV1|------------>|-----------------------+ | | | |
| | | | | |<------------|AV1 <----------------------+ | |
| | | | | |<------------|CV1 <--------------------+ | | |
| | | | +-------| |---------+ | | | | |
| | | +---------| |-------------------+ | | | | |
| | +-----------| |-------------+ | | V ^ V | |
| +-------------| |---------------+ | | | | | |
+---------------+ | | | | | | |
: | | | | | |
: | | | | | |
+-C-------------+ | | | | | | |
| +-PeerC3------| |-PeerC3--------+ | | | | | |
| | +-UDP3------| |-UDP3--------+ | | | | | | |
| | | +-RTP3----| |-RTP1--------------+ | | | | |
| | | | +-Video-| |-Video---+ | | | | |
| | | | | CV1|------------>|-------------------------+ | | |
| | | | | |<------------|AV1 <----------------------+ | |
| | | | | |<------------|BV1 <------------------+ | |
| | | | +-------| |---------+ | |
| | | +---------| |------------------------------+ |
| | +-----------| |-------------+ | |
| +-------------| |---------------+ |
+---------------+ +--------------------------------+
Figure 19: Transport Multi-party Relay
As the Relay RTP and RTCP packets between the UDP flows as indicated
by the arrows for the media flow a given WebRTC end-point, like A
will see the remote sources BV1 and CV1. There will be also two
different network paths between A, and B or C. This results in that
the client A must be capable of handlilng that when determining
congestion state that there might exist multiple destinations on the
far side of a PeerConnection and that these paths shall be treated
differently. It also results in a requirement to combine the
different congestion states into a decision to transmit a particular
media stream suitable to all participants.
It is also important to note that the relay can not perform selective
relaying of some sources and not others. The reason is that the RTCP
reporting in that case becomes incosistent and without explicit
information about it being blocked must be interpret as severe
congestion.
In this usage it is also necessary that the session management has
configured a common set of RTP configuration including RTP payload
formats as when A sends a packet with pt=97 it will arrive at both B
and C carrying pt=97 and having the same packetization and encoding,
no entity will have manipulated the packet.
When it comes to security there exist some additional requirements to
ensure that the property that the relay can't read the media traffic
is enforced. First of all the key to be used must be agreed such so
that the relay doesn't get it, e.g. no DTLS-SRTP handshake with the
relay, instead some other method must be used. Secondly, the keying
structure must be capable of handling multiple end-points in the same
RTP session.
The second problem can basically be solved in two ways. Either a
common master key from which all derive their per source key for
SRTP. The second alternative which might be more practical is that
each end-point has its own key used to protects all RTP/RTCP packets
it sends. Each participants key are then distributed to the other
participants. This second method could be implemented using DTLS-
SRTP to a special key server and then use Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] to distribute the actual used key to the
other participants in the RTP session Figure 20. The first one could
be achieved using MIKEY messages in SDP.
+---+ +---+
| | +-----------+ | |
| A |<------->| DTLS-SRTP |<------->| C |
| |<-- -->| HOST |<-- -->| |
+---+ \ / +-----------+ \ / +---+
X X
+---+ / \ +-----------+ / \ +---+
| |<-- -->| RTP |<-- -->| |
| B |<------->| RELAY |<------->| D |
| | +-----------+ | |
+---+ +---+
Figure 20: DTLS-SRTP host and RTP Relay Separated
The relay can still verify that a given SSRC isn't used or spoofed by
another participant within the multi-party session by binding SSRCs
on their first usage to a given source address and port pair.
Packets carrying that source SSRC from other addresses can be
suppressed to prevent spoofing. This is possible as long as SRTP is
used which leaves the SSRC of the packet originator in RTP and RTCP
packets in the clear. If such packet level method for enforcing
source authentication within the group, then there exist
cryptographic methods such as TESLA [RFC4383] that could be used for
true source authentication.
A.5. End-point Forwarding
An WebRTC end-point (B in Figure 21) will receive a MediaStream (set
of SSRCs) over a PeerConnection (from A). For the moment is not
decided if the end-point is allowed or not to in its turn send that
MediaStream over another PeerConnection to C. This section discusses
the RTP and end-point implications of allowing such functionality,
which on the API level is extremely simplistic to perform.
+---+ +---+ +---+
| A |--->| B |--->| C |
+---+ +---+ +---+
Figure 21: MediaStream Forwarding
There exist two main approaches to how B forwards the media from A to
C. The first one is to simply relay the media stream. The second one
is for B to act as a transcoder. Lets consider both approaches.
A relay approache will result in that the WebRTC end-points will have
to have the same capabilities as being discussed in Relay
(Appendix A.4.3). Thus A will see an RTP session that is extended
beyond the PeerConnection and see two different receiving end-points
with different path characteristics (B and C). Thus A's congestion
control needs to be capable of handling this. The security solution
can either support mechanism that allows A to inform C about the key
A is using despite B and C having agreed on another set of keys.
Alternatively B will decrypt and then re-encrypt using a new key.
The relay based approach has the advantage that B does not need to
transcode the media thus both maintaining the quality of the encoding
and reducing B's complexity requirements. If the right security
solutions are supported then also C will be able to verify the
authenticity of the media comming from A. As downside A are forced to
take both B and C into consideration when delivering content.
The media transcoder approach is similar to having B act as Mixer
terminating the RTP session combined with the transcoder as discussed
in Appendix A.4.1. A will only see B as receiver of its media. B
will responsible to produce a media stream suitable for the B to C
PeerConnection. This may require media transcoding for congestion
control purpose to produce a suitable bit-rate. Thus loosing media
quality in the transcoding and forcing B to spend the resource on the
transcoding. The media transcoding does result in a separation of
the two different legs removing almost all dependencies. B could
choice to implement logic to optimize its media transcoding
operation, by for example requesting media properties that are
suitable for C also, thus trying to avoid it having to transcode the
content and only forward the media payloads between the two sides.
For that optimization to be practical WebRTC end-points must support
sufficiently good tools for codec control.
A.6. Simulcast
This section discusses simulcast in the meaning of providing a node,
for example a stream switching Mixer, with multiple different encoded
version of the same media source. In the WebRTC context that appears
to be most easily accomplished by establishing mutliple
PeerConnection all being feed the same set of MediaStreams. Each
PeerConnection is then configured to deliver a particular media
quality and thus media bit-rate. This will work well as long as the
end-point implements media encoding according to Figure 9. Then each
PeerConnection will receive an independently encoded version and the
codec parameters can be agreed specifically in the context of this
PeerConnection.
For simulcast to work one needs to prevent that the end-point deliver
content encoded as depicted in Figure 10. If a single encoder
instance is feed to multiple PeerConnections the intention of
performing simulcast will fail.
Thus it should be considered to explicitly signal which of the two
implementation strategies that are desired and which will be done.
At least making the application and possible the central node
interested in receiving simulcast of an end-points media streams to
be aware if it will function or not.
Authors' Addresses Authors' Addresses
Colin Perkins Colin Perkins
University of Glasgow University of Glasgow
School of Computing Science School of Computing Science
Glasgow G12 8QQ Glasgow G12 8QQ
United Kingdom United Kingdom
Email: csp@csperkins.org Email: csp@csperkins.org
 End of changes. 88 change blocks. 
422 lines changed or deleted 1944 lines changed or added

This html diff was produced by rfcdiff 1.33. The latest version is available from http://tools.ietf.org/tools/rfcdiff/