One document matched: draft-ietf-rtcweb-rtp-usage-11.xml
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<rfc category="std" docName="draft-ietf-rtcweb-rtp-usage-11" ipr="trust200902">
<front>
<title abbrev="RTP for WebRTC">Web Real-Time Communication (WebRTC): Media
Transport and Use of RTP</title>
<author fullname="Colin Perkins" initials="C. S." surname="Perkins">
<organization>University of Glasgow</organization>
<address>
<postal>
<street>School of Computing Science</street>
<city>Glasgow</city>
<code>G12 8QQ</code>
<country>United Kingdom</country>
</postal>
<email>csp@csperkins.org</email>
<uri>http://csperkins.org/</uri>
</address>
</author>
<author fullname="Magnus Westerlund" initials="M." surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>SE-164 80 Kista</city>
<country>Sweden</country>
</postal>
<phone>+46 10 714 82 87</phone>
<email>magnus.westerlund@ericsson.com</email>
</address>
</author>
<author fullname="Joerg Ott" initials="J." surname="Ott">
<organization>Aalto University</organization>
<address>
<postal>
<street>School of Electrical Engineering</street>
<city>Espoo</city>
<code>02150</code>
<country>Finland</country>
</postal>
<email>jorg.ott@aalto.fi</email>
</address>
</author>
<date day="16" month="December" year="2013"/>
<workgroup>RTCWEB Working Group</workgroup>
<abstract>
<t>The Web Real-Time Communication (WebRTC) framework provides support
for direct interactive rich communication using audio, video, text,
collaboration, games, etc. between two peers' web-browsers. This memo
describes the media transport aspects of the WebRTC framework. It
specifies how the Real-time Transport Protocol (RTP) is used in the
WebRTC context, and gives requirements for which RTP features, profiles,
and extensions need to be supported.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>The <xref target="RFC3550">Real-time Transport Protocol (RTP)</xref>
provides a framework for delivery of audio and video teleconferencing
data and other real-time media applications. Previous work has defined
the RTP protocol, along with numerous profiles, payload formats, and
other extensions. When combined with appropriate signalling, these form
the basis for many teleconferencing systems.</t>
<t>The Web Real-Time communication (WebRTC) framework provides the
protocol building blocks to support direct, interactive, real-time
communication using audio, video, collaboration, games, etc., between
two peers' web-browsers. This memo describes how the RTP framework is to
be used in the WebRTC context. It proposes a baseline set of RTP
features that are to be implemented by all WebRTC-aware end-points,
along with suggested extensions for enhanced functionality.</t>
<t>This memo specifies a protocol intended for use within the WebRTC
framework, but is not restricted to that context. An overview of the
WebRTC framework is given in <xref
target="I-D.ietf-rtcweb-overview"/>.</t>
<t>The structure of this memo is as follows. <xref
target="sec-rationale"/> outlines our rationale in preparing this memo
and choosing these RTP features. <xref target="sec-terminology"/>
defines terminology. Requirements for core RTP protocols are described
in <xref target="sec-rtp-core"/> and suggested RTP extensions are
described in <xref target="sec-rtp-extn"/>. <xref
target="sec-rtp-robust"/> outlines mechanisms that can increase
robustness to network problems, while <xref target="sec-rate-control"/>
describes congestion control and rate adaptation mechanisms. The
discussion of mandated RTP mechanisms concludes in <xref
target="sec-perf"/> with a review of performance monitoring and network
management tools that can be used in the WebRTC context. <xref
target="sec-extn"/> gives some guidelines for future incorporation of
other RTP and RTP Control Protocol (RTCP) extensions into this
framework. <xref target="sec-signalling"/> describes requirements placed
on the signalling channel. <xref target="sec-webrtc-api"/> discusses the
relationship between features of the RTP framework and the WebRTC
application programming interface (API), and <xref
target="sec-rtp-func"/> discusses RTP implementation considerations. The
memo concludes with <xref target="sec-security">security
considerations</xref> and <xref target="sec-iana">IANA
considerations</xref>.</t>
</section>
<section anchor="sec-rationale" title="Rationale">
<t>The RTP framework comprises the RTP data transfer protocol, the RTP
control protocol, and numerous RTP payload formats, profiles, and
extensions. This range of add-ons has allowed RTP to meet various needs
that were not envisaged by the original protocol designers, and to
support many new media encodings, but raises the question of what
extensions are to 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 20 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.</t>
<t>Other RTP and RTCP extensions not discussed in this document can be
implemented by WebRTC end-points if they are beneficial for new use
cases. However, they are not necessary to address the WebRTC use cases
and requirements identified to date <xref
target="I-D.ietf-rtcweb-use-cases-and-requirements"/>.</t>
<t>While the baseline set of RTP features and extensions defined in this
memo is targeted 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 appropriate for other desktop or mobile video
conferencing systems, or for room-based high-quality telepresence
applications.</t>
</section>
<section anchor="sec-terminology" title="Terminology">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in <xref target="RFC2119"/>.
The RFC 2119 interpretation of these key words applies only when written
in ALL CAPS. Lower- or mixed-case uses of these key words are not to be
interpreted as carrying special significance in this memo.</t>
<t>We define the following terms:<list style="hanging">
<t hangText="RTP Media Stream:">A sequence of RTP packets, and
associated RTCP packets, using a single synchronisation source
(SSRC) that together carries part or all of the content of a
specific Media Type from a specific sender source within a given RTP
session.</t>
<t hangText="RTP Session:">As defined by <xref target="RFC3550"/>,
the endpoints belonging to the same RTP Session are those that share
a single SSRC space. That is, those endpoints can see an SSRC
identifier transmitted by any one of the other endpoints. An
endpoint can see an SSRC either directly in RTP and RTCP packets, or
as a contributing source (CSRC) in RTP packets from a mixer. The RTP
Session scope is hence decided by the endpoints' network
interconnection topology, in combination with RTP and RTCP
forwarding strategies deployed by endpoints and any interconnecting
middle nodes.</t>
<t hangText="WebRTC MediaStream:">The MediaStream concept defined by
the W3C in the API.</t>
</list></t>
<t>Other terms are used according to their definitions from the <xref
target="RFC3550">RTP Specification</xref>.</t>
</section>
<section anchor="sec-rtp-core" title="WebRTC Use of RTP: Core Protocols">
<t>The following sections describe the core features of RTP and RTCP
that need to be implemented, along with the mandated RTP profiles and
payload formats. Also described are the core extensions providing
essential features that all WebRTC implementations need to implement to
function effectively on today's networks.</t>
<section anchor="sec-rtp-rtcp" title="RTP and RTCP">
<t>The <xref target="RFC3550">Real-time Transport Protocol (RTP)
</xref> is REQUIRED to be 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 RTP, and MUST be implemented in all WebRTC
applications.</t>
<t>The following RTP and RTCP features are sometimes omitted in
limited functionality implementations of RTP, but are REQUIRED in all
WebRTC implementations: <list style="symbols">
<t>Support for use of multiple simultaneous SSRC values in a
single RTP session, including support for RTP end-points that send
many SSRC values simultaneously, following <xref
target="RFC3550"/> and <xref
target="I-D.ietf-avtcore-rtp-multi-stream"/>. Support for the RTCP
optimisations for multi-SSRC sessions defined in <xref
target="I-D.ietf-avtcore-rtp-multi-stream-optimisation"/> is
RECOMMENDED.</t>
<t>Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (see also <xref
target="sec-ssrc"/>).</t>
<t>Support for reception of RTP data packets containing CSRC
lists, as generated by RTP mixers, and RTCP packets relating to
CSRCs.</t>
<t>Sending correct synchronisation information in the RTCP Sender
Reports, to allow receivers to implement lip-sync, with support
for the rapid RTP synchronisation extensions (see <xref
target="rapid-sync"/>) being RECOMMENDED.</t>
<t>Support for multiple synchronisation contexts. Participants
that send multiple simultaneous RTP media streams MAY do so as
part of a single synchronisation context, using a single RTCP
CNAME for all streams and allowing receivers to play the streams
out in a synchronised manner, or they MAY use different
synchronisation contexts, and hence different RTCP CNAMEs, for
some or all of the streams. Receivers MUST support reception of
multiple RTCP CNAMEs from each participant in an RTP session. See
also <xref target="sec-cname"/>.</t>
<t>Support for sending and receiving RTCP SR, RR, SDES, and BYE
packet types, with OPTIONAL support for other RTCP packet types;
implementations MUST ignore unknown RTCP packet types. Note that
additional RTCP Packet types are needed by the <xref
target="sec-profile">RTP/SAVPF Profile</xref> and the other <xref
target="sec-rtp-extn">RTCP extensions</xref>.</t>
<t>Support for multiple end-points in a single RTP session, and
for scaling the RTCP transmission interval according to the number
of participants in the session; support for randomised RTCP
transmission intervals to avoid synchronisation of RTCP reports;
support for RTCP timer reconsideration.</t>
<t>Support for configuring the RTCP bandwidth as a fraction of the
media bandwidth, and for configuring the fraction of the RTCP
bandwidth allocated to senders, e.g., using the SDP "b=" line.</t>
</list></t>
<t>It is known that a significant number of legacy RTP
implementations, especially those targeted at VoIP-only systems, do
not support all of the above features, and in some cases do not
support RTCP at all. Implementers are advised to consider the
requirements for graceful degradation when interoperating with legacy
implementations.</t>
<t>Other implementation considerations are discussed in <xref
target="sec-rtp-func"/>.</t>
</section>
<section anchor="sec-profile" title="Choice of the RTP Profile">
<t>The complete specification of RTP for a particular application
domain requires the choice of an RTP Profile. For WebRTC use, the
<xref target="RFC5124">Extended Secure RTP Profile for RTCP-Based
Feedback (RTP/SAVPF)</xref>, as extended by <xref target="RFC7007"/>,
MUST be implemented. This builds on the basic <xref
target="RFC3551">RTP/AVP profile</xref>, the <xref
target="RFC4585">RTP profile for RTCP-based feedback
(RTP/AVPF)</xref>, and the <xref target="RFC3711">secure RTP profile
(RTP/SAVP)</xref>.</t>
<t>The RTCP-based feedback extensions <xref target="RFC4585"/> are
needed for the improved RTCP timer model, that allows more flexible
transmission of RTCP packets in response to events, rather than
strictly according to bandwidth. This is vital for being able to
report congestion events. These extensions also save RTCP bandwidth,
and will commonly only use the full RTCP bandwidth allocation if there
are many events that require feedback. They are also needed to make
use of the RTP conferencing extensions discussed in <xref
target="conf-ext"/>.</t>
<t><list style="empty">
<t>Note: The enhanced RTCP timer model defined in the RTP/AVPF
profile is backwards compatible with legacy systems that implement
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).</t>
</list></t>
<!--MW: Should we really define which transforms for SRTP to use, or does this belong
to draft-ietf-rtcweb-security-arch?-->
<t>The secure RTP profile <xref target="RFC3711"/> is needed to
provide media encryption, integrity protection, replay protection and
a limited form of source authentication. WebRTC implementations MUST
NOT send packets using the basic RTP/AVP profile or the RTP/AVPF
profile; they MUST employ the full RTP/SAVPF profile to protect all
RTP and RTCP packets that are generated. The default and mandatory to
implement transforms listed in Section 5 of <xref target="RFC3711"/>
SHALL apply.</t>
<t>The keying mechanism(s) to be used with the RTP/SAVPF profile are
defined in Section 5.5 of <xref
target="I-D.ietf-rtcweb-security-arch"/> or its replacement.</t>
</section>
<section anchor="sec.codecs" title="Choice of RTP Payload Formats">
<t>The set of mandatory to implement codecs and RTP payload formats
for WebRTC is not specified in this memo. Implementations can support
any codec for which an RTP payload format and associated signalling is
defined. Implementation cannot assume that the other participants in
an RTP session understand any RTP payload format, no matter how
common; the mapping between RTP payload type numbers and specific
configurations of particular RTP payload formats MUST be agreed before
those payload types/formats can be used. In an SDP context, this can
be done using the "a=rtpmap:" and "a=fmtp:" attributes associated with
an "m=" line.</t>
<t>Endpoints can signal support for multiple RTP payload formats, or
multiple configurations of a single RTP payload format, as long as
each unique RTP payload format configuration uses a different RTP
payload type number. As outlined in <xref target="sec-ssrc"/>, the RTP
payload type number is sometimes used to associate an RTP media stream
with a signalling context. This association is possible provided
unique RTP payload type numbers are used in each context. For example,
an RTP media stream can be associated with an SDP "m=" line by
comparing the RTP payload type numbers used by the media stream with
payload types signalled in the "a=rtpmap:" lines in the media sections
of the SDP. If RTP media streams are being associated with signalling
contexts based on the RTP payload type, then the assignment of RTP
payload type numbers MUST be unique across signalling contexts; if the
same RTP payload format configuration is used in multiple contexts,
then a different RTP payload type number has to be assigned in each
context to ensure uniqueness. If the RTP payload type number is not
being used to associated RTP media streams with a signalling context,
then the same RTP payload type number can be used to indicate the
exact same RTP payload format configuration in multiple contexts.</t>
<t>An endpoint that has signalled support for multiple RTP payload
formats SHOULD accept data in any of those payload formats at any
time, unless it has previously signalled limitations on its decoding
capability. This requirement is constrained if several types of media
(e.g., audio and video) are sent in the same RTP session. In such a
case, a source (SSRC) is restricted to switching only between the RTP
payload formats signalled for the type of media that is being sent by
that source; see <xref target="sec.session-mux"/>. To support rapid
rate adaptation by changing codec, RTP does not require advance
signalling for changes between RTP payload formats that were signalled
during session set-up.</t>
<t>An RTP sender that changes between two RTP payload types that use
different RTP clock rates MUST follow the recommendations in Section
4.1 of <xref target="I-D.ietf-avtext-multiple-clock-rates"/>. RTP
receivers MUST follow the recommendations in Section 4.3 of <xref
target="I-D.ietf-avtext-multiple-clock-rates"/>, in order to support
sources that switch between clock rates in an RTP session (these
recommendations for receivers are backwards compatible with the case
where senders use only a single clock rate).</t>
</section>
<section anchor="sec.session-mux" title="Use of RTP Sessions">
<t>An association amongst a set of participants communicating using
RTP is known as an RTP session. A participant can be involved in
several RTP sessions at the same time. In a multimedia session, each
type of media has typically been carried in a separate RTP session
(e.g., using one RTP session for the audio, and a separate RTP session
using different transport addresses for the video). WebRTC
implementations of RTP are REQUIRED to implement support for
multimedia sessions in this way, separating each session using
different transport-layer addresses (e.g., different UDP ports) for
compatibility with legacy systems.</t>
<t>In modern day networks, however, with the widespread use of network
address/port translators (NAT/NAPT) and firewalls, it is desirable to
reduce the number of transport-layer flows used by RTP applications.
This can be done by sending all the RTP media streams in a single RTP
session, which will comprise a single transport-layer flow (this will
prevent the use of some quality-of-service mechanisms, as discussed in
<xref target="sec-differentiated"/>). Implementations are REQUIRED to
support transport of all RTP media streams, independent of media type,
in a single RTP session according to <xref
target="I-D.ietf-avtcore-multi-media-rtp-session"/>. If multiple types
of media are to be used in a single RTP session, all participants in
that session MUST agree to this usage. In an SDP context, <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation"/> can be used to
signal this.</t>
<!--MW: What are we doing with the below paragraph?-->
<t>It is also possible to use a shim-based approach to run multiple
RTP sessions on a single transport-layer flow. This gives advantages
in some gateway scenarios, and makes it easy to distinguish groups of
RTP media streams that might need distinct processing. One way of
doing this is described in <xref
target="I-D.westerlund-avtcore-transport-multiplexing"/>. At the time
of this writing, there is no consensus to use a shim-based approach in
WebRTC implementations.</t>
<t>Further discussion about when different RTP session structures and
multiplexing methods are suitable can be found in <xref
target="I-D.ietf-avtcore-multiplex-guidelines"/>.</t>
</section>
<section anchor="sec.rtcp-mux" title="RTP and RTCP Multiplexing">
<t>Historically, RTP and RTCP have been run on separate transport
layer addresses (e.g., two UDP ports for each RTP session, one port
for RTP and one port for RTCP). With the increased use of Network
Address/Port Translation (NAPT) this has become problematic, since
maintaining multiple NAT bindings can be costly. It also complicates
firewall administration, since multiple ports need to be opened to
allow RTP traffic. To reduce these costs and session set-up times,
support for multiplexing RTP data packets and RTCP control packets on
a single port for each RTP session is REQUIRED, as specified in <xref
target="RFC5761"/>. For backwards compatibility, implementations are
also REQUIRED to support RTP and RTCP sent on separate transport-layer
addresses.</t>
<t>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 can be useful to
keep NAT bindings alive, and is the recommend method for application
level <xref target="RFC6263">keep-alives of RTP sessions</xref>.</t>
</section>
<section title="Reduced Size RTCP">
<t>RTCP packets are usually sent as compound RTCP packets, and <xref
target="RFC3550"/> requires that those compound packets start with an
Sender Report (SR) or Receiver Report (RR) packet. When using frequent
RTCP feedback messages under the RTP/AVPF Profile <xref
target="RFC4585"/> these statistics are not needed in every packet,
and unnecessarily increase the mean RTCP packet size. This can limit
the frequency at which RTCP packets can be sent within the RTCP
bandwidth share.</t>
<t>To avoid this problem, <xref target="RFC5506"/> specifies how to
reduce the mean RTCP message size and allow for more frequent
feedback. Frequent feedback, in turn, is essential to make real-time
applications quickly aware of changing network conditions, and to
allow them to adapt their transmission and encoding behaviour. Support
for non-compound RTCP feedback packets <xref target="RFC5506"/> is
REQUIRED, but MUST be negotiated using the signalling channel before
use. For backwards compatibility, implementations are also REQUIRED to
support the use of compound RTCP feedback packets if the remote
endpoint does not agree to the use of non-compound RTCP in the
signalling exchange.</t>
</section>
<section title="Symmetric RTP/RTCP">
<t>To ease traversal of NAT and firewall devices, implementations are
REQUIRED to implement and use <xref target="RFC4961">Symmetric
RTP</xref>. The reasons for using symmetric RTP is primarily to avoid
issues with NAT and Firewalls by ensuring that the flow is actually
bi-directional and thus kept alive and registered as flow the intended
recipient actually wants. In addition, it saves resources,
specifically ports at the end-points, but also in the network as NAT
mappings or firewall state is not unnecessary bloated. Also the amount
of QoS state is reduced.</t>
</section>
<section anchor="sec-ssrc"
title="Choice of RTP Synchronisation Source (SSRC)">
<t>Implementations are REQUIRED to support signalled RTP
synchronisation source (SSRC) identifiers, using the "a=ssrc:" SDP
attribute defined in Section 4.1 and Section 5 of <xref
target="RFC5576"/>. Implementations MUST also support the
"previous-ssrc" source attribute defined in Section 6.2 of <xref
target="RFC5576"/>. Other per-SSRC attributes defined in <xref
target="RFC5576"/> MAY be supported.</t>
<t>Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP
session is OPTIONAL. Implementations MUST be prepared to accept RTP
and RTCP packets using SSRCs that have not been explicitly signalled
ahead of time. Implementations MUST support random SSRC assignment,
and MUST support SSRC collision detection and resolution, according to
<xref target="RFC3550"/>. When using signalled SSRC values, collision
detection MUST be performed as described in Section 5 of <xref
target="RFC5576"/>.</t>
<t>It is often desirable to associate an RTP media stream with a
non-RTP context (e.g., to associate an RTP media stream with an "m="
line in a session description formatted using SDP). If SSRCs are
signalled this is straightforward (in SDP the "a=ssrc:" line will be
at the media level, allowing a direct association with an "m=" line).
If SSRCs are not signalled, the RTP payload type numbers used in an
RTP media stream are often sufficient to associate that media stream
with a signalling context (e.g., if RTP payload type numbers are
assigned as described in <xref target="sec.codecs"/> of this memo, the
RTP payload types used by an RTP media stream can be compared with
values in SDP "a=rtpmap:" lines, which are at the media level in SDP,
and so map to an "m=" line).</t>
</section>
<section anchor="sec-cname"
title="Generation of the RTCP Canonical Name (CNAME)">
<t>The RTCP Canonical Name (CNAME) provides a persistent
transport-level identifier for an RTP endpoint. While the
Synchronisation Source (SSRC) identifier for an RTP endpoint can
change if a collision is detected, or when the RTP application is
restarted, its RTCP CNAME is meant to stay unchanged, so that RTP
endpoints can be uniquely identified and associated with their RTP
media streams within a set of related RTP sessions. For proper
functionality, each RTP endpoint needs to have at least one unique
RTCP CNAME value. An endpoint MAY have multiple CNAMEs, as the CNAME
also identifies a particular synchronisation context, i.e. all SSRC
associated with a CNAME share a common reference clock, and if an
endpoint have SSRCs associated with different reference clocks it will
need to use multiple CNAMEs. This ought not be common, and if possible
reference clocks ought to be mapped to each other and one chosen to be
used with RTP and RTCP.</t>
<t>The <xref target="RFC3550">RTP specification</xref> includes
guidelines for choosing a unique RTP CNAME, but these are not
sufficient in the presence of NAT devices. In addition, long-term
persistent identifiers can be problematic from a privacy viewpoint.
Accordingly, support for generating a short-term persistent RTCP
CNAMEs following <xref target="RFC7022"/> is RECOMMENDED.</t>
<t>An WebRTC end-point MUST support reception of any CNAME that
matches the syntax limitations specified by the <xref
target="RFC3550">RTP specification</xref> and cannot assume that any
CNAME will be chosen according to the form suggested above.</t>
</section>
</section>
<section anchor="sec-rtp-extn" title="WebRTC Use of RTP: Extensions">
<t>There are a number of RTP extensions that are either needed to obtain
full functionality, or extremely useful to improve on the baseline
performance, in the WebRTC application context. One set of these
extensions is related to conferencing, while others are more generic in
nature. The following subsections describe the various RTP extensions
mandated or suggested for use within the WebRTC context.</t>
<section anchor="conf-ext" title="Conferencing Extensions">
<t>RTP is inherently a group communication protocol. Groups can be
implemented using a centralised server, multi-unicast, or using IP
multicast. While IP multicast is popular in IPTV systems,
overlay-based topologies dominate in interactive conferencing
environments. Such overlay-based topologies typically use one or more
central servers to connect end-points in a star or flat tree topology.
These central servers can be implemented in a number of ways as
discussed in the memo on <xref
target="I-D.ietf-avtcore-rtp-topologies-update"> RTP
Topologies</xref>.</t>
<t>Not all of the possible the overlay-based topologies are suitable
for use in the WebRTC environment. Specifically: <list style="symbols">
<t>The use of video switching MCUs makes the use of RTCP for
congestion control and quality of service reports problematic (see
Section 3.6.2 of <xref
target="I-D.ietf-avtcore-rtp-topologies-update"/>).</t>
<t>The use of content modifying MCUs with RTCP termination breaks
RTP loop detection, and prevents receivers from identifying active
senders (see section 3.8 of <xref
target="I-D.ietf-avtcore-rtp-topologies-update"/>).</t>
</list> Accordingly, only Point to Point (Topo-Point-to-Point),
Multiple concurrent Point to Point (Mesh) and RTP Mixers (Topo-Mixer)
topologies are needed to achieve the use-cases to be supported in
WebRTC initially. These RECOMMENDED topologies are expected to be
supported by all WebRTC end-points (these topologies require no
special RTP-layer support in the end-point if the RTP features
mandated in this memo are implemented).</t>
<t>The RTP extensions described in <xref target="sec-fir"/> to <xref
target="sec.tmmbr"/> are designed to be used with centralised
conferencing, where an RTP middlebox (e.g., a conference bridge)
receives a participant's RTP media streams and distributes them to the
other participants. These extensions are not necessary for
interoperability; an RTP endpoint that does not implement these
extensions will work correctly, but might offer poor performance.
Support for the listed extensions will greatly improve the quality of
experience and, to provide a reasonable baseline quality, some these
extensions are mandatory to be supported by WebRTC end-points.</t>
<t>The RTCP conferencing extensions are defined in <xref
target="RFC4585">Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)</xref> and the <xref
target="RFC5104">"Codec Control Messages in the RTP Audio-Visual
Profile with Feedback (AVPF)" (CCM)</xref> and are fully usable by the
<xref target="RFC5124">Secure variant of this profile
(RTP/SAVPF)</xref>.</t>
<section anchor="sec-fir" title="Full Intra Request (FIR)">
<t>The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of
the <xref target="RFC5104">Codec Control Messages</xref>. This
message is used to make the mixer request a new Intra picture from a
participant in the session. This is used when switching between
sources to ensure that the receivers can decode the video or other
predictive media encoding with long prediction chains. WebRTC
senders MUST understand and react to the FIR feedback message since
it greatly improves the user experience when using centralised
mixer-based conferencing; support for sending the FIR message is
OPTIONAL.</t>
</section>
<section title="Picture Loss Indication (PLI)">
<t>The Picture Loss Indication is defined in Section 6.3.1 of the
<xref target="RFC4585">RTP/AVPF profile</xref>. It is used by a
receiver to tell the sending encoder that it lost the decoder
context and would like to have it repaired somehow. This is
semantically different from the Full Intra Request above as there
could be multiple ways to fulfil the request. WebRTC senders MUST
understand and react to this feedback message as a loss tolerance
mechanism; receivers MAY send PLI messages.</t>
</section>
<section title="Slice Loss Indication (SLI)">
<t>The Slice Loss Indicator is defined in Section 6.3.2 of the <xref
target="RFC4585">RTP/AVPF profile</xref>. It is used by a receiver
to tell the encoder that it has detected the loss or corruption of
one or more consecutive macro blocks, and would like to have these
repaired somehow. Support for this feedback message is OPTIONAL as a
loss tolerance mechanism.</t>
</section>
<section title="Reference Picture Selection Indication (RPSI)">
<t>Reference Picture Selection Indication (RPSI) is defined in
Section 6.3.3 of the <xref target="RFC4585">RTP/AVPF profile
</xref>. Some video coding standards 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 encoder has learned about a
loss of encoder-decoder synchronisation, a known-as-correct
reference picture can be used for future coding. The RPSI message
allows this to be signalled. Support for RPSI messages is
OPTIONAL.</t>
</section>
<section title="Temporal-Spatial Trade-off Request (TSTR)">
<t>The temporal-spatial trade-off request and notification are
defined in Sections 3.5.2 and 4.3.2 of <xref target="RFC5104"/>.
This request can be used to ask the video encoder to change the
trade-off it makes between temporal and spatial resolution, for
example to prefer high spatial image quality but low frame rate.
Support for TSTR requests and notifications is OPTIONAL.</t>
</section>
<section anchor="sec.tmmbr"
title="Temporary Maximum Media Stream Bit Rate Request (TMMBR)">
<t>This feedback message is defined in Sections 3.5.4 and 4.2.1 of
the <xref target="RFC5104">Codec Control Messages</xref>. This
message and its notification message are used by a media receiver to
inform the sending party that there is a current limitation on the
amount of bandwidth available to this receiver. This can be various
reasons for this: for example, an RTP mixer can use this message to
limit the media rate of the sender being forwarded by the mixer
(without doing media transcoding) to fit the bottlenecks existing
towards the other session participants. WebRTC senders are REQUIRED
to implement support for TMMBR messages, and MUST follow bandwidth
limitations set by a TMMBR message received for their SSRC. The
sending of TMMBR requests is OPTIONAL.</t>
</section>
</section>
<section title="Header Extensions">
<t>The <xref target="RFC3550">RTP specification</xref> provides the
capability to include RTP header extensions containing in-band data,
but the format and semantics of the extensions are poorly specified.
The use of header extensions is OPTIONAL in the WebRTC context, but if
they are used, they MUST be formatted and signalled following the
general mechanism for RTP header extensions defined in <xref
target="RFC5285"/>, since this gives well-defined semantics to RTP
header extensions.</t>
<t>As noted in <xref target="RFC5285"/>, the requirement from the RTP
specification that header extensions are "designed so that the header
extension may be ignored" <xref target="RFC3550"/> stands. To be
specific, header extensions MUST only be used for data that can safely
be ignored by the recipient without affecting interoperability, and
MUST NOT be used when the 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 is signalled (e.g., as defined by
the payload type). Valid examples might include metadata that is
additional to the usual RTP information.</t>
<section anchor="rapid-sync" title="Rapid Synchronisation">
<t>Many RTP sessions require synchronisation between audio, video,
and other content. This synchronisation is performed by receivers,
using information contained in RTCP SR packets, as described in the
<xref target="RFC3550">RTP specification</xref>. This basic
mechanism can be slow, however, so it is RECOMMENDED that the rapid
RTP synchronisation extensions described in <xref target="RFC6051"/>
be implemented in addition to RTCP SR-based synchronisation. The
rapid synchronisation extensions use the general RTP header
extension mechanism <xref target="RFC5285"/>, which requires
signalling, but are otherwise backwards compatible.</t>
</section>
<section anchor="sec-client-to-mixer"
title="Client-to-Mixer Audio Level">
<t>The <xref target="RFC6464">Client to Mixer Audio Level
extension</xref> is an RTP header extension used by a client to
inform a mixer about the level of audio activity in the packet to
which the header is attached. This enables a central node to make
mixing or selection decisions without decoding or detailed
inspection of the payload, reducing the complexity in some types of
central RTP nodes. It can also save decoding resources in receivers,
which can choose to decode only the most relevant RTP media streams
based on audio activity levels.</t>
<t>The <xref target="RFC6464">Client-to-Mixer Audio Level</xref>
extension is RECOMMENDED to be implemented. If it is implemented, it
is REQUIRED that the header extensions are encrypted according to
<xref target="RFC6904"/> since the information contained in these
header extensions can be considered sensitive.</t>
</section>
<section anchor="sec-mixer-to-client"
title="Mixer-to-Client Audio Level">
<t>The <xref target="RFC6465">Mixer to Client Audio Level header
extension</xref> provides the client with the audio level of the
different sources mixed into a common mix by a RTP mixer. This
enables a user interface to indicate the relative activity level of
each session participant, rather than just being included or not
based on the CSRC field. This is a pure optimisations of non
critical functions, and is hence OPTIONAL to implement. If it is
implemented, it is REQUIRED that the header extensions are encrypted
according to <xref target="RFC6904"/> since the information
contained in these header extensions can be considered
sensitive.</t>
</section>
<section anchor="sec-mapping-to-signalling"
title="Associating RTP Media Streams and Signalling Contexts">
<t>(tbd: it seems likely that we need a mechanism to associate RTP
media streams with signalling contexts. The mechanism by which this
is done will likely be some combination of an RTP header extension,
periodic transmission of a new RTCP SDES item, and some signalling
extension. The semantics of those items are not yet settled; see
draft-westerlund-avtext-rtcp-sdes-srcname, draft-ietf-mmusic-msid,
and draft-even-mmusic-application-token for discussion).</t>
</section>
</section>
</section>
<section anchor="sec-rtp-robust"
title="WebRTC Use of RTP: Improving Transport Robustness">
<t>There are tools that can make RTP media streams robust against packet
loss and reduce the impact of loss on media quality. However, they all
add extra bits compared to a non-robust stream. The overhead of these
extra bits needs to be considered, and the aggregate bit-rate MUST be
rate controlled to avoid causing network congestion (see <xref
target="sec-rate-control"/>). As a result, improving robustness might
require a lower base encoding quality, but has the potential to deliver
that quality with fewer errors. The mechanisms described in the
following sub-sections can be used to improve tolerance to packet
loss.</t>
<section anchor="sec-rtx"
title="Negative Acknowledgements and RTP Retransmission">
<t>As a consequence of supporting the RTP/SAVPF profile,
implementations can support negative acknowledgements (NACKs) for RTP
data packets <xref target="RFC4585"/>. This feedback can be used to
inform a sender of the loss of particular RTP packets, subject to the
capacity limitations of the RTCP feedback channel. A sender can use
this information to optimise the user experience by adapting the media
encoding to compensate for known lost packets, for example.</t>
<t>Senders are REQUIRED to understand the Generic NACK message defined
in Section 6.2.1 of <xref target="RFC4585"/>, but MAY choose to ignore
this feedback (following Section 4.2 of <xref target="RFC4585"/>).
Receivers MAY send NACKs for missing RTP packets; <xref
target="RFC4585"/> provides some guidelines on when to send NACKs. It
is not expected that a receiver will send a NACK for every lost RTP
packet, rather it needs to consider the cost of sending NACK feedback,
and the importance of the lost packet, to make an informed decision on
whether it is worth telling the sender about a packet loss event.</t>
<t>The <xref target="RFC4588">RTP Retransmission Payload Format</xref>
offers the ability to retransmit lost packets based on NACK feedback.
Retransmission needs to be used with care in interactive real-time
applications to ensure that the retransmitted packet arrives in time
to be useful, but can be effective in environments with relatively low
network RTT (an RTP sender can estimate the RTT to the receivers using
the information in RTCP SR and RR packets, as described at the end of
Section 6.4.1 of <xref target="RFC3550"/>). The use of retransmissions
can also increase the forward RTP bandwidth, and can potentially
worsen the problem if the packet loss was caused by network
congestion. We note, however, that retransmission of an important lost
packet to repair decoder state can have lower cost than sending a full
intra frame. It is not appropriate to blindly retransmit RTP packets
in response to a NACK. The importance of lost packets and the
likelihood of them arriving in time to be useful needs to be
considered before RTP retransmission is used.</t>
<t>Receivers are REQUIRED to implement support for RTP retransmission
packets <xref target="RFC4588"/>. Senders MAY send RTP retransmission
packets in response to NACKs if the RTP retransmission payload format
has been negotiated for the session, and if the sender believes it is
useful to send a retransmission of the packet(s) referenced in the
NACK. An RTP sender does not need to retransmit every NACKed
packet.</t>
</section>
<section anchor="sec-FEC" title="Forward Error Correction (FEC)">
<t>The use of Forward Error Correction (FEC) can provide an effective
protection against some degree of packet loss, at the cost of steady
bandwidth overhead. There are several FEC schemes that are defined for
use with RTP. Some of these schemes are specific to a particular RTP
payload format, others operate across RTP packets and can be used with
any payload format. It needs to be noted that using redundant encoding
or FEC will lead to increased play out delay, which needs to be
considered when choosing the redundancy or FEC formats and their
respective parameters.</t>
<t>If an RTP payload format negotiated for use in a WebRTC session
supports redundant transmission or FEC as a standard feature of that
payload format, then that support MAY be used in the WebRTC session,
subject to any appropriate signalling.</t>
<t>There are several block-based FEC schemes that are designed for use
with RTP independent of the chosen RTP payload format. At the time of
this writing there is no consensus on which, if any, of these FEC
schemes is appropriate for use in the WebRTC context. Accordingly,
this memo makes no recommendation on the choice of block-based FEC for
WebRTC use.</t>
</section>
</section>
<section anchor="sec-rate-control"
title="WebRTC Use of RTP: Rate Control and Media Adaptation">
<t>WebRTC will be used in heterogeneous network environments using a
variety set of link technologies, including both wired and wireless
links, to interconnect potentially large groups of users around the
world. As a result, the network paths between users can have widely
varying one-way delays, available bit-rates, load levels, and traffic
mixtures. Individual end-points can send one or more RTP media streams
to each participant in a WebRTC conference, and there can be several
participants. Each of these RTP media streams can contain different
types of media, and the type of media, bit rate, and number of flows can
be highly asymmetric. Non-RTP traffic can share the network paths with
RTP flows. Since the network environment is not predictable or stable,
WebRTC endpoints MUST ensure that the RTP traffic they generate can
adapt to match changes in the available network capacity.</t>
<t>The quality of experience for users of WebRTC implementation is very
dependent on effective adaptation of the media to the limitations of the
network. End-points have to be designed so they do not transmit
significantly more data than the network path can support, except for
very short time periods, otherwise high levels of network packet loss or
delay spikes will occur, causing media quality degradation. The limiting
factor on the capacity of the network path might be the link bandwidth,
or it might be competition with other traffic on the link (this can be
non-WebRTC traffic, traffic due to other WebRTC flows, or even
competition with other WebRTC flows in the same session).</t>
<t>An effective media congestion control algorithm is therefore an
essential part of the WebRTC framework. However, at the time of this
writing, there is no standard congestion control algorithm that can be
used for interactive media applications such as WebRTC flows. Some
requirements for congestion control algorithms for WebRTC sessions are
discussed in <xref target="I-D.jesup-rtp-congestion-reqs"/>, and it is
expected that a future version of this memo will mandate the use of a
congestion control algorithm that satisfies these requirements.</t>
<section title="Boundary Conditions and Circuit Breakers">
<t>In the absence of a concrete congestion control algorithm, all
WebRTC implementations MUST implement the RTP circuit breaker
algorithm that is in described <xref
target="I-D.ietf-avtcore-rtp-circuit-breakers"/>. The RTP circuit
breaker is designed to enable applications to recognise and react to
situations of extreme network congestion. However, since the RTP
circuit breaker might not be triggered until congestion becomes
extreme, it cannot be considered a substitute for congestion control,
and applications MUST also implement congestion control to allow them
to adapt to changes in network capacity. Any future RTP congestion
control algorithms are expected to operate within the envelope allowed
by the circuit breaker.</t>
<t>The session establishment signalling will also necessarily
establish boundaries to which the media bit-rate will conform. The
choice of media codecs provides upper- and lower-bounds on the
supported bit-rates that the application can utilise to provide useful
quality, and the packetization choices that exist. In addition, the
signalling channel can establish maximum media bit-rate boundaries
using the SDP "b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary
Maximum Media Stream Bit Rate (TMMBR) Requests (see <xref
target="sec.tmmbr"/> of this memo). The combination of media codec
choice and signalled bandwidth limits SHOULD be used to limit traffic
based on known bandwidth limitations, for example the capacity of the
edge links, to the extent possible.</t>
</section>
<section title="RTCP Limitations for Congestion Control">
<t>Experience with the congestion control algorithms of TCP <xref
target="RFC5681"/>, TFRC <xref target="RFC5348"/>, and DCCP <xref
target="RFC4341"/>, <xref target="RFC4342"/>, <xref
target="RFC4828"/>, has shown that feedback on packet arrivals needs
to be sent roughly once per round trip time. We note that the
real-time media traffic might not have to adapt to changing path
conditions as rapidly as needed for the elastic applications TCP was
designed for, but frequent feedback is still needed to allow the
congestion control algorithm to track the path dynamics.</t>
<t>The total RTCP bandwidth is limited in its transmission rate to a
fraction of the RTP traffic (by default 5%). RTCP packets are larger
than, e.g., TCP ACKs (even when non-compound RTCP packets are used).
The RTP media stream bit rate thus limits the maximum feedback rate as
a function of the mean RTCP packet size.</t>
<t>Interactive communication might not be able to afford waiting for
packet losses to occur to indicate congestion, because an increase in
play out delay due to queuing (most prominent in wireless networks)
can easily lead to packets being dropped due to late arrival at the
receiver. Therefore, more sophisticated cues might need to be reported
-- to be defined in a suitable congestion control framework as noted
above -- which, in turn, increase the report size again. For example,
different RTCP XR report blocks (jointly) provide the necessary
details to implement a variety of congestion control algorithms, but
the (compound) report size grows quickly.</t>
<t>In group communication, the share of RTCP bandwidth needs to be
shared by all group members, reducing the capacity and thus the
reporting frequency per node.</t>
<t>Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP
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 for
every other frame in a 30 fps video.</t>
</section>
<section title="Congestion Control Interoperability and Legacy Systems">
<t>There are legacy implementations that do not implement RTCP, and
hence do not provide any congestion feedback. Congestion control
cannot be performed with these end-points. WebRTC implementations that
need to interwork with such end-points MUST limit their transmission
to a low rate, equivalent to a VoIP call using a low bandwidth codec,
that is unlikely to cause any significant congestion.</t>
<t>When interworking with legacy implementations that support RTCP
using the <xref target="RFC3551">RTP/AVP profile</xref>, congestion
feedback is provided in RTCP RR packets every few seconds.
Implementations that have to interwork with such end-points MUST
ensure that they keep within the <xref
target="I-D.ietf-avtcore-rtp-circuit-breakers"> RTP circuit
breaker</xref> constraints to limit the congestion they can cause.</t>
<t>If a legacy end-point supports RTP/AVPF, this enables negotiation
of important parameters for frequent reporting, such as the "trr-int"
parameter, and the possibility that the end-point supports some useful
feedback format for congestion control purpose such as <xref
target="RFC5104"> TMMBR</xref>. Implementations that have to interwork
with such end-points MUST ensure that they stay within the <xref
target="I-D.ietf-avtcore-rtp-circuit-breakers"> RTP circuit
breaker</xref> constraints to limit the congestion they can cause, but
might find that they can achieve better congestion response depending
on the amount of feedback that is available.</t>
<t>With proprietary congestion control algorithms issues can arise
when different algorithms and implementations interact in a
communication session. If the different implementations have made
different choices in regards to the type of adaptation, for example
one sender based, and one receiver based, then one could end up in
situation where one direction is dual controlled, when the other
direction is not controlled. This memo cannot mandate behaviour for
proprietary congestion control algorithms, but implementations that
use such algorithms ought to be aware of this issue, and try to ensure
that both effective congestion control is negotiated for media flowing
in both directions. If the IETF were to standardise both sender- and
receiver-based congestion control algorithms for WebRTC traffic in the
future, the issues of interoperability, control, and ensuring that
both directions of media flow are congestion controlled would also
need to be considered.</t>
</section>
</section>
<section anchor="sec-perf"
title="WebRTC Use of RTP: Performance Monitoring">
<t>As described in <xref target="sec-rtp-rtcp"/>, implementations are
REQUIRED to generate RTCP Sender Report (SR) and Reception Report (RR)
packets relating to the RTP media streams they send and receive. These
RTCP reports can be used for performance monitoring purposes, since they
include basic packet loss and jitter statistics.</t>
<t>A large number of additional performance metrics are supported by the
RTCP Extended Reports (XR) framework <xref target="RFC3611"/><xref
target="RFC6792"/>. It is not yet clear what extended metrics are
appropriate for use in the WebRTC context, so there is no requirement
that implementations generate RTCP XR packets. However, implementations
that can use detailed performance monitoring data MAY generate RTCP XR
packets as appropriate; the use of such packets SHOULD be signalled in
advance.</t>
<t>All WebRTC implementations MUST be prepared to receive RTP XR report
packets, whether or not they were signalled. There is no requirement
that the data contained in such reports be used, or exposed to the
Javascript application, however.</t>
</section>
<section anchor="sec-extn" title="WebRTC Use of RTP: Future Extensions">
<t>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 <xref target="RFC2736"> Guidelines for Writers of RTP
Payload Format Specifications </xref> and <xref target="RFC5968">
Guidelines for Extending the RTP Control Protocol</xref>, and SHOULD
take into account any future guidelines for extending RTP and related
protocols that have been developed.</t>
<t>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 will adopt RTP extensions
that are of general utility, to enable easy implementation of a gateway
to other applications using RTP, rather than adopt mechanisms that are
narrowly targeted at specific WebRTC use cases.</t>
</section>
<section anchor="sec-signalling" title="Signalling Considerations">
<t>RTP is built with the assumption that an external signalling channel
exists, and can be used to configure RTP sessions and their features.
The basic configuration of an RTP session consists of the following
parameters:</t>
<t><list style="hanging">
<t hangText="RTP Profile:">The name of the RTP profile to be used in
session. The <xref target="RFC3551">RTP/AVP</xref> and <xref
target="RFC4585">RTP/AVPF</xref> profiles can interoperate on basic
level, as can their secure variants <xref
target="RFC3711">RTP/SAVP</xref> and <xref
target="RFC5124">RTP/SAVPF</xref>. The secure variants of the
profiles do not directly interoperate with the non-secure variants,
due to the presence of additional header fields for authentication
in SRTP packets and cryptographic transformation of the payload.
WebRTC requires the use of the RTP/SAVPF profile, and this MUST be
signalled if SDP is used. Interworking functions might transform
this into the RTP/SAVP profile for a legacy use case, by indicating
to the WebRTC end-point that the RTP/SAVPF is used, and limiting the
usage of the "a=rtcp:" attribute to indicate a trr-int value of 4
seconds.</t>
<t hangText="Transport Information:">Source and destination IP
address(s) and ports for RTP and RTCP MUST be signalled for each RTP
session. In WebRTC these transport addresses will be provided by ICE
that signals candidates and arrives at nominated candidate address
pairs. If <xref target="RFC5761">RTP and RTCP multiplexing</xref> is
to be used, such that a single port is used for RTP and RTCP flows,
this MUST be signalled (see <xref target="sec.rtcp-mux"/>). If
several RTP sessions are to be multiplexed onto a single transport
layer flow, this MUST also be signalled (see <xref
target="sec.session-mux"/>).</t>
<t
hangText="RTP Payload Types, media formats, and 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). <xref target="sec.codecs"/> of this
memo discusses requirements for uniqueness of payload types.</t>
<t hangText="RTP Extensions:">The RTP extensions to be used SHOULD
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.</t>
<t hangText="RTCP Bandwidth:">Support for exchanging RTCP Bandwidth
values to the end-points will be necessary. This SHALL be done as
described in <xref target="RFC3556">"Session Description Protocol
(SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP)
Bandwidth"</xref>, 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
can lead to failure to interoperate.</t>
</list></t>
<t>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 upon, 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.</t>
</section>
<section anchor="sec-webrtc-api" title="WebRTC API Considerations">
<t>The <xref target="W3C.WD-webrtc-20130910">WebRTC API</xref> and the
<xref target="W3C.WD-mediacapture-streams-20130903">Media Capture and
Streams API</xref> defines and uses the concept of a MediaStream that
consists of zero or more MediaStreamTracks. A MediaStreamTrack is an
individual stream of media from any type of media source like a
microphone or a camera, but also conceptual sources, like a audio mix or
a video composition, are possible. The MediaStreamTracks within a
MediaStream need to be possible to play out synchronised.</t>
<t>A MediaStreamTrack's realisation in RTP in the context of an
RTCPeerConnection consists of a source packet stream identified with an
SSRC within an RTP session part of the RTCPeerConnection. The
MediaStreamTrack can also result in additional packet streams, and thus
SSRCs, in the same RTP session. These can be dependent packet streams
from scalable encoding of the source stream associated with the
MediaStreamTrack, if such a media encoder is used. They can also be
redundancy packet streams, these are created when applying <xref
target="sec-FEC">Forward Error Correction</xref> or <xref
target="sec-rtx">RTP retransmission</xref> to the source packet stream.
<list style="empty">
<t>Note: It is quite likely that a simulcast specification will
result in multiple source packet streams, and thus SSRCs, based on
the same source stream associated with the MediaStreamTrack being
simulcasted. Each such source packet stream can have dependent and
redundant packet streams associated with them. However, the final
conclusion on this awaits the specification of simulcast. Simulcast
will also require signalling to correctly separate and associate the
source packet streams with their sets of dependent and/or redundant
streams.</t>
</list></t>
<t>It is important to note that the same media source can be feeding
multiple MediaStreamTracks. As different sets of constraints or other
parameters can be applied to the MediaStreamTrack, each MediaStreamTrack
instance added to a RTCPeerConnection SHALL result in an independent
source packet stream, with its own set of associated packet streams, and
thus different SSRC(s). It will depend on applied constraints and
parameters if the source stream and the encoding configuration will be
identical between different MediaStreamTracks sharing the same media
source. Thus it is possible for multiple source packet streams to share
encoded streams (but not packet streams), but this is an implementation
choice to try to utilise such optimisations. Note that such
optimizations would need to take into account that the constraints for
one of the MediaStreamTracks can at any moment change, meaning that the
encoding configurations should no longer be identical.</t>
<t>The same MediaStreamTrack can also be included in multiple
MediaStreams, thus multiple sets of MediaStreams can implicitly need to
use the same synchronisation base. To ensure that this works in all
cases, and don't forces a endpoint to change synchronisation base and
CNAME in the middle of a ongoing delivery of any packet streams, which
would cause media disruption; all MediaStreamTracks and their associated
SSRCs originating from the same endpoint MUST be sent using the same
CNAME within one RTCPeerConnection as well as across all
RTCPeerConnections part of the same communication session context, which
for a browser are a single origin. <list style="empty">
<t>Note: It is important that the same CNAME is not used in
different communication session contexts or origins, as that could
enable tracking of a user and its device usage of different
services. See Section 4.4.1 of <xref
target="I-D.ietf-rtcweb-security">Security Considerations for
WebRTC</xref> for further discussion.</t>
<t>The reasons to require the same CNAME across multiple
RTCPeerConnections is to enable synchronisation of different
MediaStreamTracks originating from one endpoint despite them being
transported over different RTCPeerConnections.</t>
</list></t>
<t>The above will currently force a WebRTC endpoint that receives an
MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing on
any RTCPeerConnection to perform resynchronisation of the stream. This,
as the sending party needs to change the CNAME, which implies that it
has to use a locally available system clock as timebase for the
synchronisation. Thus, the relative relation between the timebase of the
incoming stream and the system sending out needs to defined. This
relation also needs monitoring for clock drift and likely adjustments of
the synchronisation. The sending entity is also responsible for
congestion control for its the sent streams. In cases of packet loss the
loss of incoming data also needs to be handled. This leads to the
observation that the method that is least likely to cause issues or
interruptions in the outgoing source packet stream is a model of full
decoding, including repair etc followed by encoding of the media again
into the outgoing packet stream. Optimisations of this method is clearly
possible and implementation specific.</t>
<t>A WebRTC endpoint MUST support receiving multiple MediaStreamTracks,
where each of different MediaStreamTracks (and their sets of associated
packet streams) uses different CNAMEs. However, MediaStreamTracks that
are received with different CNAMEs have no defined synchronisation.<list
style="empty">
<t>Note: The motivation for supporting reception of multiple CNAMEs
are to allow for forward compatibility with any future changes that
enables more efficient stream handling when endpoints relay/forward
streams. It also ensures that endpoints can interoperate with
certain types of multi-stream middleboxes or endpoints that are not
WebRTC.</t>
</list></t>
<t>The binding between the WebRTC MediaStreams, MediaStreamTracks and
the SSRC is done as specified in <xref
target="I-D.ietf-mmusic-msid">"Cross Session Stream Identification in
the Session Description Protocol"</xref>. <xref
target="I-D.ietf-mmusic-msid">This document</xref> also defines, in
section 4.1, how to map unknown source packet stream SSRCs to
MediaStreamTracks and MediaStreams. Commonly the RTP Payload Type of any
incoming packets will reveal if the packet stream is a source stream or
a redundancy or dependent packet stream. The association to the correct
source packet stream depends on the payload format in use for the packet
stream.</t>
</section>
<section anchor="sec-rtp-func" title="RTP Implementation Considerations">
<t>The following discussion provides some guidance on the implementation
of the RTP features described in this memo. The focus is on a WebRTC
end-point implementation perspective, and while some mention is made of
the behaviour of middleboxes, that is not the focus of this memo.</t>
<section title="Configuration and Use of RTP Sessions">
<t>A WebRTC end-point will be a simultaneous participant in one or
more RTP sessions. Each RTP session can convey multiple media flows,
and can include media data from multiple end-points. In the following,
we outline some ways in which WebRTC end-points can configure and use
RTP sessions.</t>
<section anchor="sec.multiple-flows"
title="Use of Multiple Media Flows Within an RTP Session">
<t>RTP is a group communication protocol, and in a WebRTC context
every RTP session can potentially contain multiple media flows.
There are several reasons why this might be desirable: <list
style="hanging">
<t hangText="Multiple media types:">Outside of WebRTC, it is
common to use one RTP session for each type of media (e.g., one
RTP session for audio and one for video, each sent on a
different UDP port). However, to reduce the number of UDP ports
used, the default in WebRTC is to send all types of media in a
single RTP session, as described in <xref
target="sec.session-mux"/>, using RTP and RTCP multiplexing
(<xref target="sec.rtcp-mux"/>) to further reduce the number of
UDP ports needed. This RTP session then uses only one UDP flow,
but will contain multiple RTP media streams, each containing a
different type of media. A common example might be an end-point
with a camera and microphone that sends two RTP streams, one
video and one audio, into a single RTP session.</t>
<t hangText="Multiple Capture Devices:">A WebRTC end-point might
have multiple cameras, microphones, or other media capture
devices, and so might want to generate several RTP media streams
of the same media type. Alternatively, it might want to send
media from a single capture device in several different formats
or quality settings at once. Both can result in a single
end-point sending multiple RTP media streams of the same media
type into a single RTP session at the same time.</t>
<t hangText="Associated Repair Data:">An end-point might send a
media stream that is somehow associated with another stream. For
example, it might send an RTP stream that contains FEC or
retransmission data relating to another stream. Some RTP payload
formats send this sort of associated repair data as part of the
original media stream, while others send it as a separate
stream.</t>
<t hangText="Layered or Multiple Description Coding:">An
end-point can use a layered media codec, for example H.264 SVC,
or a multiple description codec, that generates multiple media
flows, each with a distinct RTP SSRC, within a single RTP
session.</t>
<t hangText="RTP Mixers, Translators, and Other Middleboxes:">An
RTP session, in the WebRTC context, is a point-to-point
association between an end-point and some other peer device,
where those devices share a common SSRC space. The peer device
might be another WebRTC end-point, or it might be an RTP mixer,
translator, or some other form of media processing middlebox. In
the latter cases, the middlebox might send mixed or relayed RTP
streams from several participants, that the WebRTC end-point
will need to render. Thus, even though a WebRTC end-point might
only be a member of a single RTP session, the peer device might
be extending that RTP session to incorporate other end-points.
WebRTC is a group communication environment and end-points need
to be capable of receiving, decoding, and playing out multiple
RTP media streams at once, even in a single RTP session.</t>
</list></t>
</section>
<section anchor="sec.multiple-sessions"
title="Use of Multiple RTP Sessions">
<t>In addition to sending and receiving multiple media streams
within a single RTP session, a WebRTC end-point might participate in
multiple RTP sessions. There are several reasons why a WebRTC
end-point might choose to do this: <list style="hanging">
<t hangText="To interoperate with legacy devices:">The common
practice in the non-WebRTC world is to send different types of
media in separate RTP sessions, for example using one RTP
session for audio and another RTP session, on a different UDP
port, for video. All WebRTC end-points need to support the
option of sending different types of media on different RTP
sessions, so they can interwork with such legacy devices. This
is discussed further in <xref target="sec.session-mux"/>.</t>
<t hangText="To provide enhanced quality of service:">Some
network-based quality of service mechanisms operate on the
granularity of UDP 5-tuples. If it is desired to use these
mechanisms to provide differentiated quality of service for some
RTP flows, then those RTP flows need to be sent in a separate
RTP session using a different UDP port number, and with
appropriate quality of service marking. This is discussed
further in <xref target="sec-differentiated"/>.</t>
<!--MW: The below paragraph discusses simulcast. Based on developments the perceived
easiest way of doing it might be wrong. May need update after IETF 88.-->
<t hangText="To separate media with different purposes:">An
end-point might want to send media streams that have different
purposes on different RTP sessions, to make it easy for the peer
device to distinguish them. For example, some centralised
multiparty conferencing systems display the active speaker in
high resolution, but show low resolution "thumbnails" of other
participants. Such systems might configure the end-points to
send simulcast high- and low-resolution versions of their video
using separate RTP sessions, to simplify the operation of the
central mixer. In the WebRTC context this appears to be most
easily accomplished by establishing multiple RTCPeerConnection
all being feed the same set of WebRTC MediaStreams. Each
RTCPeerConnection is then configured to deliver a particular
media quality and thus media bit-rate, and will produce an
independently encoded version with the codec parameters agreed
specifically in the context of that RTCPeerConnection. The
central mixer can always distinguish packets corresponding to
the low- and high-resolution streams by inspecting their SSRC,
RTP payload type, or some other information contained in RTP
header extensions or RTCP packets, but it can be easier to
distinguish the flows if they arrive on separate RTP sessions on
separate UDP ports.</t>
<t hangText="To directly connect with multiple peers:">A
multi-party conference does not need to use a central mixer.
Rather, a multi-unicast mesh can be created, comprising several
distinct RTP sessions, with each participant sending RTP traffic
over a separate RTP session (that is, using an independent
RTCPeerConnection object) to every other participant, as shown
in <xref target="fig-mesh"/>. This topology has the benefit of
not requiring a central mixer node that is trusted to access and
manipulate the media data. The downside is that it increases the
used bandwidth at each sender by requiring one copy of the RTP
media streams for each participant that are part of the same
session beyond the sender itself.</t>
</list></t>
<figure align="center" anchor="fig-mesh"
title="Multi-unicast using several RTP sessions">
<artwork><![CDATA[
+---+ +---+
| A |<--->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
]]></artwork>
</figure>
<t><list style="hanging">
<t>The multi-unicast topology could also 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 RTCPeerConnection
objects we recommend that this is implemented as several
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, since it will not see RTCP reports for the RTP
session between B and C, whereas it would it all three
participants were part of a single RTP session. Experience with
the Mbone tools (experimental RTP-based multicast conferencing
tools from the late 1990s) has showed that RTCP reception
quality reports for third parties can usefully be presented to
the users in a way that helps them understand asymmetric network
problems, and the approach of using separate RTP sessions
prevents this. However, an advantage of using separate RTP
sessions is that it enables using different media bit-rates and
RTP session configurations between the different peers, thus not
forcing B to endure the same quality reductions if there are
limitations in the transport from A to C as C will. It it
believed that these advantages outweigh the limitations in
debugging power.</t>
<t hangText="To indirectly connect with multiple peers:">A
common scenario in multi-party conferencing is to create
indirect connections to multiple peers, using an RTP mixer,
translator, or some other type of RTP middlebox. <xref
target="fig-mixerFirst"/> outlines a simple topology that might
be used in a four-person centralised conference. The middlebox
acts to optimise the transmission of RTP media streams from
certain perspectives, either by only sending some of the
received RTP media stream to any given receiver, or by providing
a combined RTP media stream out of a set of contributing
streams.</t>
</list></t>
<figure align="center" anchor="fig-mixerFirst"
title="RTP mixer with only unicast paths">
<artwork><![CDATA[
+---+ +-------------+ +---+
| A |<---->| |<---->| B |
+---+ | RTP mixer, | +---+
| translator, |
| or other |
+---+ | middlebox | +---+
| C |<---->| |<---->| D |
+---+ +-------------+ +---+
]]></artwork>
</figure>
<t><list style="hanging">
<t>There are various methods of implementation for the
middlebox. If implemented as a standard RTP mixer or translator,
a single RTP session will extend across the middlebox and
encompass all the end-points in one multi-party session. Other
types of middlebox might use separate RTP sessions between each
end-point and the middlebox. A common aspect is that these
central nodes can use 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 controlling the most suitable
frame-rate, picture resolution, the trade-off between frame-rate
and spatial quality. The middlebox gets the significant
responsibility to correctly perform congestion control, source
identification, manage synchronisation while providing the
application with suitable media optimizations. The middlebox is
also has to be a trusted node when it comes to security, since
it manipulates either the RTP header or the media itself (or
both) received from one end-point, before sending it on towards
the end-point(s), thus they need to be able to decrypt and then
encrypt it before sending it out.</t>
<t>RTP Mixers can create a situation where an end-point
experiences a situation in-between a session with only two
end-points and multiple RTP sessions. Mixers are expected to not
forward RTCP reports regarding RTP media streams across
themselves. This is due to the difference in the RTP media
streams provided to the different end-points. The original media
source lacks information about a mixer's manipulations prior to
sending it the different receivers. This scenario also results
in that an end-point's feedback or requests goes to the mixer.
When the mixer can't act on this by itself, it is forced to go
to the original media source to fulfil the receivers request.
This will not necessarily be explicitly visible any RTP and RTCP
traffic, but the interactions and the time to complete them will
indicate such dependencies.</t>
<t>Providing source authentication in multi-party scenarios is a
challenge. In the mixer-based topologies, end-points source
authentication is based on, firstly, verifying that media comes
from the mixer by cryptographic verification and, secondly,
trust in 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 will 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 have SSRC
fields that came from other end-points before. For cryptographic
verification of the source SRTP would require additional
security mechanisms, for example <xref target="RFC4383">TESLA
for SRTP</xref>, that are not part of the base WebRTC
standards.</t>
<t hangText="To forward media between multiple peers:">It is
sometimes desirable for an end-point that receives an RTP media
stream to be able to forward that media stream to a third party.
The are some obvious security and privacy implications in
supporting this, but also potential uses. This is supported in
the W3C API by taking the received and decoded media and using
it as media source that is re-encoding and transmitted as a new
stream.</t>
<t>At the RTP layer, media forwarding acts as a back-to-back RTP
receiver and RTP sender. The receiving side terminates the RTP
session and decodes the media, while the sender side re-encodes
and transmits the media using an entirely separate RTP session.
The original sender will only see a single receiver of the
media, and will not be able to tell that forwarding is happening
based on RTP-layer information since the RTP session that is
used to send the forwarded media is not connected to the RTP
session on which the media was received by the node doing the
forwarding.</t>
<t>The end-point that is performing the forwarding is
responsible for producing an RTP media stream suitable for
onwards transmission. The outgoing RTP session that is used to
send the forwarded media is entirely separate to the RTP session
on which the media was received. This will require media
transcoding for congestion control purpose to produce a suitable
bit-rate for the outgoing RTP session, reducing media quality
and forcing the forwarding end-point to spend the resource on
the transcoding. The media transcoding does result in a
separation of the two different legs removing almost all
dependencies, and allowing the forwarding end-point to optimize
its media transcoding operation. The cost is greatly increased
computational complexity on the forwarding node. Receivers of
the forwarded stream will see the forwarding device as the
sender of the stream, and will not be able to tell from the RTP
layer that they are receiving a forwarded stream rather than an
entirely new media stream generated by the forwarding
device.</t>
</list></t>
</section>
<section anchor="sec-differentiated"
title="Differentiated Treatment of Flows">
<t>There are use cases for differentiated treatment of RTP media
streams. Such differentiation can happen at several places in the
system. First of all is the prioritization within the end-point
sending the media, which controls, both which RTP media streams that
will be sent, and their allocation of bit-rate out of the current
available aggregate as determined by the congestion control.</t>
<t>It is expected that the WebRTC API will allow the application to
indicate relative priorities for different MediaStreamTracks. These
priorities can then be used to influence the local RTP processing,
especially when it comes to congestion control response in how to
divide the available bandwidth between the RTP flows. Any changes in
relative priority will also need to be considered for RTP flows that
are associated with the main RTP flows, such as RTP retransmission
streams and FEC. The importance of such associated RTP traffic flows
is dependent on the media type and codec used, in regards to how
robust that codec is to packet loss. However, a default policy might
to be to use the same priority for associated RTP flows as for the
primary RTP flow.</t>
<t>Secondly, the network can prioritize packet flows, including RTP
media streams. Typically, differential treatment includes two steps,
the first being identifying whether an IP packet belongs to a class
that has to be treated differently, the second the actual mechanism
to prioritize packets. This is done according to three methods:
<list style="hanging">
<t hangText="DiffServ:">The end-point marks a packet with a
DiffServ code point to indicate to the network that the packet
belongs to a particular class.</t>
<t hangText="Flow based:">Packets that need to be given a
particular treatment are identified using a combination of IP
and port address.</t>
<t hangText="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.</t>
</list></t>
<t>Flow-based differentiation will provide the same treatment to all
packets within a flow, i.e., relative prioritization is not
possible. Moreover, if the resources are limited it might 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 it so that it can provide the separation of the RTP media
streams onto different UDP flows to enable a more granular usage of
flow based differentiation. That way at least providing different
prioritization of audio and video if desired by application.</t>
<t>DiffServ assumes that either the end-point or a classifier can
mark the packets with an appropriate DSCP so that the packets are
treated according to that marking. If the end-point is to mark the
traffic two requirements arise in the WebRTC context: 1) The WebRTC
application or browser has to know which DSCP to use and that it can
use them on some set of RTP media streams. 2) The information needs
to be propagated to the operating system when transmitting the
packet. Details of this process are outside the scope of this memo
and are further discussed in <xref
target="I-D.dhesikan-tsvwg-rtcweb-qos">"DSCP and other packet
markings for RTCWeb QoS"</xref>.</t>
<t>For packet based marking schemes it might be possible to mark
individual RTP packets differently based on the relative priority of
the RTP payload. For example video codecs that have I, P, and B
pictures could prioritise any payloads carrying only B frames less,
as these are less damaging to loose. As default policy all RTP
packets related to a media stream ought to be provided with the same
prioritization; per-packet prioritization is outside the scope of
this memo, but might be specified elsewhere in future.</t>
<t>It is also important to consider how RTCP packets associated with
a particular RTP media flow need to be marked. RTCP compound packets
with Sender Reports (SR), ought to be marked with the same priority
as the RTP media flow itself, so the RTCP-based round-trip time
(RTT) measurements are done using the same flow priority as the
media flow experiences. RTCP compound packets containing RR packet
ought to be sent with the priority used by the majority of the RTP
media flows reported on. RTCP packets containing time-critical
feedback packets can use higher priority to improve the timeliness
and likelihood of delivery of such feedback.</t>
</section>
</section>
<section title="Source, Flow, and Participant Identification">
<section title="Media Streams">
<t>Each RTP media stream is identified by a unique synchronisation
source (SSRC) identifier. The SSRC identifier is carried in the RTP
data packets comprising a media stream, and is also used to identify
that stream in the corresponding RTCP reports. The SSRC is chosen as
discussed in <xref target="sec-ssrc"/>. The first stage in
demultiplexing RTP and RTCP packets received at a WebRTC end-point
is to separate the media streams based on their SSRC value; once
that is done, additional demultiplexing steps can determine how and
where to render the media.</t>
<t>RTP allows a mixer, or other RTP-layer middlebox, to combine
media flows from multiple sources to form a new media flow. The RTP
data packets in that new flow can include a Contributing Source
(CSRC) list, indicating which original SSRCs contributed to the
combined packet. As described in <xref target="sec-rtp-rtcp"/>,
implementations need to support reception of RTP data packets
containing a CSRC list and RTCP packets that relate to sources
present in the CSRC list. The CSRC list can change on a
packet-by-packet basis, depending on the mixing operation being
performed. Knowledge of what sources contributed to a particular RTP
packet can be important if the user interface indicates which
participants are active in the session. Changes in the CSRC list
included in packets needs to be exposed to the WebRTC application
using some API, if the application is to be able to track changes in
session participation. It is desirable to map CSRC values back into
WebRTC MediaStream identities as they cross this API, to avoid
exposing the SSRC/CSRC name space to JavaScript applications.</t>
<t>If the mixer-to-client audio level extension <xref
target="RFC6465"/> is being used in the session (see <xref
target="sec-mixer-to-client"/>), the information in the CSRC list is
augmented by audio level information for each contributing source.
This information can usefully be exposed in the user interface.</t>
</section>
<section title="Media Streams: SSRC Collision Detection">
<t>The <xref target="RFC3550">RTP standard</xref> requires any RTP
implementation to have support for detecting and handling SSRC
collisions, i.e., resolve the conflict when two different end-points
use the same SSRC value. This requirement also applies to WebRTC
end-points. There are several scenarios where SSRC collisions can
occur.</t>
<t>In a point-to-point session where each SSRC is associated with
either of the two end-points and where the main media carrying SSRC
identifier will be announced in the signalling channel, a collision
is less likely to occur due to the information about used SSRCs
provided by <xref target="RFC5576">Source-Specific SDP
Attributes</xref>. Still if both end-points start 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 <xref target="RFC5576">Source-Specific SDP Attributes</xref>
contains no mechanism to resolve SSRC collisions or reject a
end-points usage of an SSRC.</t>
<t>There could also appear SSRC values that are not signalled. This
is more likely than it appears as certain RTP functions need extra
SSRCs to provide functionality related to another (the "main") SSRC,
for example, <xref target="RFC4588">SSRC multiplexed RTP
retransmission</xref>. In those cases, an end-point can create a new
SSRC that strictly doesn't need to be announced over the signalling
channel to function correctly on both RTP and RTCPeerConnection
level.</t>
<t>The more likely case for SSRC collision is that multiple
end-points in a multiparty conference create new sources and signals
those towards the central server. In cases where the SSRC/CSRC are
propagated between the different end-points from the central node
collisions can occur.</t>
<t>Another scenario is when the central node manages to connect an
end-point's RTCPeerConnection to another RTCPeerConnection the
end-point already has, thus forming a loop where the end-point will
receive its own traffic. While is is clearly considered a bug, it is
important that the end-point is able to recognise and handle the
case when it occurs. This case becomes even more problematic when
media mixers, and so on, are involved, where the stream received is
a different stream but still contains this client's input.</t>
<t>These SSRC/CSRC collisions can only be handled on RTP level as
long as the same RTP session is extended across multiple
RTCPeerConnections by a RTP middlebox. To resolve the more generic
case where multiple RTCPeerConnections are interconnected, then
identification of the media source(s) part of a MediaStreamTrack
being propagated across multiple interconnected RTCPeerConnection
needs to be preserved across these interconnections.</t>
</section>
<section title="Media Synchronisation Context">
<t>When an end-point sends media from more than one media source, it
needs to consider if (and which of) these media sources are to be
synchronized. In RTP/RTCP, synchronisation is provided by having a
set of RTP media streams be indicated as coming from the same
synchronisation context and logical end-point by using the same RTCP
CNAME identifier.</t>
<t>The next provision is that the internal clocks of all media
sources, i.e., what drives the RTP timestamp, can be correlated to a
system clock that is provided in RTCP Sender Reports encoded in an
NTP format. By correlating all RTP timestamps to a common system
clock for all sources, the timing relation of the different RTP
media streams, also across multiple RTP sessions can be derived at
the receiver and, if desired, the streams can be synchronized. The
requirement is for the media sender to provide the correlation
information; it is up to the receiver to use it or not.</t>
</section>
</section>
</section>
<section anchor="sec-security" title="Security Considerations">
<t>The overall security architecture for WebRTC is described in <xref
target="I-D.ietf-rtcweb-security-arch"/>, and security considerations
for the WebRTC framework are described in <xref
target="I-D.ietf-rtcweb-security"/>. These considerations apply to this
memo also.</t>
<t>The security considerations of the RTP specification, the RTP/SAVPF
profile, and the various RTP/RTCP extensions and RTP payload formats
that form the complete protocol suite described in this memo apply. We
do not believe there are any new security considerations resulting from
the combination of these various protocol extensions.</t>
<t>The <xref target="RFC5124">Extended Secure RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback</xref> (RTP/SAVPF)
provides handling of fundamental issues by offering confidentiality,
integrity and partial source authentication. A mandatory to implement
media security solution is created by combing this secured RTP profile
and <xref target="RFC5764">DTLS-SRTP keying</xref> as defined by <xref
target="I-D.ietf-rtcweb-security-arch">Section 5.5 of</xref>.</t>
<t>RTCP packets convey a Canonical Name (CNAME) identifier that is used
to associate media flows that need to be synchronised across related RTP
sessions. Inappropriate choice of CNAME values can be a privacy concern,
since long-term persistent CNAME identifiers can be used to track users
across multiple WebRTC calls. <xref target="sec-cname"/> of this memo
provides guidelines for generation of untraceable CNAME values that
alleviate this risk.</t>
<t>The guidelines in <xref target="RFC6562"/> apply when using variable
bit rate (VBR) audio codecs such as Opus (see <xref
target="sec.codecs"/> for discussion of mandated audio codecs). These
guidelines in <xref target="RFC6562"/> also apply, but are of lesser
importance, when using the client-to-mixer audio level header extensions
(<xref target="sec-client-to-mixer"/>) or the mixer-to-client audio
level header extensions (<xref target="sec-mixer-to-client"/>).</t>
</section>
<section anchor="sec-iana" title="IANA Considerations">
<t>This memo makes no request of IANA.</t>
<t>Note to RFC Editor: this section is to be removed on publication as
an RFC.</t>
</section>
<section title="Open Issues">
<t>This section contains a summary of the open issues or to be done
things noted in the document:<list style="numbers">
<t>tbd: The discussion at IETF 88 confirmed that there is broad
agreement to support simulcast, however the method for achieving
simulcast of a media source has to be decided.</t>
</list></t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>The authors would like to thank Bernard Aboba, Harald Alvestrand,
Cary Bran, Charles Eckel, Cullen Jennings, Dan Romascanu, and the other
members of the IETF RTCWEB working group for their valuable
feedback.</t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.3550"?>
<?rfc include='reference.RFC.2119'?>
<?rfc include='reference.RFC.2736'?>
<?rfc include='reference.RFC.3551'?>
<?rfc include='reference.RFC.3556'?>
<?rfc include='reference.RFC.3711'?>
<?rfc include='reference.RFC.4585'?>
<?rfc include='reference.RFC.4588'?>
<?rfc include='reference.RFC.4961'?>
<?rfc include='reference.RFC.5104'?>
<?rfc include='reference.RFC.5124'?>
<?rfc include='reference.RFC.5285'?>
<?rfc include='reference.RFC.5506'?>
<?rfc include='reference.RFC.5761'?>
<?rfc include='reference.RFC.5764'?>
<?rfc include='reference.RFC.6051'?>
<?rfc include='reference.RFC.6464'?>
<?rfc include='reference.RFC.6465'?>
<?rfc include='reference.RFC.6562'?>
<?rfc include='reference.RFC.6904'?>
<?rfc include='reference.RFC.7007'?>
<?rfc include='reference.RFC.7022'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
<?rfc include='reference.I-D.ietf-avtcore-multi-media-rtp-session'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-multi-stream'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-multi-stream-optimisation'?>
<?rfc include='reference.I-D.ietf-rtcweb-security'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-circuit-breakers'?>
<?rfc include='reference.I-D.ietf-avtext-multiple-clock-rates'?>
<?rfc include='reference.I-D.ietf-rtcweb-security-arch'?>
<?rfc include='reference.W3C.WD-webrtc-20130910'?>
<?rfc include='reference.W3C.WD-mediacapture-streams-20130903'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.3611'?>
<?rfc include='reference.RFC.4341'?>
<?rfc include='reference.RFC.4342'?>
<?rfc include='reference.RFC.4383'?>
<?rfc include='reference.RFC.4828'?>
<?rfc include='reference.RFC.5348'?>
<?rfc include='reference.RFC.5576'?>
<?rfc include='reference.RFC.5681'?>
<?rfc include='reference.RFC.5968'?>
<?rfc include='reference.RFC.6263'?>
<?rfc include='reference.RFC.6792'?>
<?rfc include='reference.I-D.ietf-mmusic-msid'?>
<?rfc include='reference.I-D.ietf-rtcweb-overview'?>
<?rfc include='reference.I-D.ietf-rtcweb-use-cases-and-requirements'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-topologies-update'?>
<?rfc include='reference.I-D.ietf-avtcore-multiplex-guidelines'?>
<?rfc include='reference.I-D.westerlund-avtcore-transport-multiplexing'?>
<?rfc include='reference.I-D.jesup-rtp-congestion-reqs'?>
<?rfc include='reference.I-D.dhesikan-tsvwg-rtcweb-qos'?>
</references>
</back>
</rfc>
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| PAFTECH AB 2003-2026 | 2026-04-23 21:40:21 |