One document matched: draft-ietf-rtcweb-rtp-usage-25.xml
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<rfc category="std" docName="draft-ietf-rtcweb-rtp-usage-25" 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>https://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="12" month="June" year="2015" />
<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 Endpoints, 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"></xref>.</t>
<t>The structure of this memo is as follows. <xref
target="sec-rationale"></xref> outlines our rationale in preparing this
memo and choosing these RTP features. <xref
target="sec-terminology"></xref> defines terminology. Requirements for
core RTP protocols are described in <xref target="sec-rtp-core"></xref>
and suggested RTP extensions are described in <xref
target="sec-rtp-extn"></xref>. <xref target="sec-rtp-robust"></xref>
outlines mechanisms that can increase robustness to network problems,
while <xref target="sec-rate-control"></xref> describes congestion
control and rate adaptation mechanisms. The discussion of mandated RTP
mechanisms concludes in <xref target="sec-perf"></xref> with a review of
performance monitoring and network management tools. <xref
target="sec-extn"></xref> gives some guidelines for future incorporation
of other RTP and RTP Control Protocol (RTCP) extensions into this
framework. <xref target="sec-signalling"></xref> describes requirements
placed on the signalling channel. <xref target="sec-webrtc-api"></xref>
discusses the relationship between features of the RTP framework and the
WebRTC application programming interface (API), and <xref
target="sec-rtp-func"></xref> 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 to review the available
RTP features and extensions, and to define a common baseline RTP feature
set for all WebRTC Endpoints. This builds on the past 20 years of RTP
development 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>RTP and RTCP extensions that are not discussed in this document can
be implemented by WebRTC Endpoints if they are beneficial for new use
cases. However, they are not necessary to address the WebRTC use cases
and requirements identified in <xref target="RFC7478"></xref>.</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"></xref>. 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 additional terms:<list style="hanging">
<t hangText="WebRTC MediaStream:">The MediaStream concept defined by
the W3C in the <xref
target="W3C.WD-mediacapture-streams-20130903">WebRTC API</xref>. A
MediaStream consists of zero or more MediaStreamTracks.</t>
<t hangText="MediaStreamTrack:">Part of the MediaStream concept
defined by the W3C in the <xref
target="W3C.WD-mediacapture-streams-20130903">WebRTC API</xref>. 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.</t>
<t hangText="Transport-layer Flow:">A uni-directional flow of
transport packets that are identified by having a particular 5-tuple
of source IP address, source port, destination IP address,
destination port, and transport protocol used.</t>
<t hangText="Bi-directional Transport-layer Flow:">A bi-directional
transport-layer flow is a transport-layer flow that is symmetric.
That is, the transport-layer flow in the reverse direction has a
5-tuple where the source and destination address and ports are
swapped compared to the forward path transport-layer flow, and the
transport protocol is the same.</t>
</list></t>
<t>This document uses the terminology from <xref
target="I-D.ietf-avtext-rtp-grouping-taxonomy"></xref> and <xref
target="I-D.ietf-rtcweb-overview"></xref>. Other terms are used
according to their definitions from the <xref target="RFC3550">RTP
Specification</xref>. Especially note the following frequently used
terms: RTP Packet Stream, RTP Session, and Endpoint.</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. Also
described are the core extensions providing essential features that all
WebRTC Endpoints 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 and used in all
WebRTC Endpoints.</t>
<t>The following RTP and RTCP features are sometimes omitted in
limited functionality implementations of RTP, but are REQUIRED in all
WebRTC Endpoints: <list style="symbols">
<t>Support for use of multiple simultaneous SSRC values in a
single RTP session, including support for RTP endpoints that send
many SSRC values simultaneously, following <xref
target="RFC3550"></xref> and <xref
target="I-D.ietf-avtcore-rtp-multi-stream"></xref>. The RTCP
optimisations for multi-SSRC sessions defined in <xref
target="I-D.ietf-avtcore-rtp-multi-stream-optimisation"></xref>
MAY be supported; if supported the usage MUST be signalled.</t>
<t>Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (see also <xref
target="sec-ssrc"></xref>).</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-synchronisation; see
<xref target="rapid-sync"></xref> regarding support for the rapid
RTP synchronisation extensions.</t>
<t>Support for multiple synchronisation contexts. Participants
that send multiple simultaneous RTP packet streams SHOULD 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. For compatibility with potential
future versions of this specification, or for interoperability
with non-WebRTC devices through a gateway, receivers MUST support
multiple synchronisation contexts, indicated by the use of
multiple RTCP CNAMEs in an RTP session. This specification
mandates the usage of a single CNAME when sending RTP Packet
Streams in some circumstances, see <xref
target="sec-cname"></xref>.</t>
<t>Support for sending and receiving RTCP SR, RR, SDES, and BYE
packet types. Note that support for other RTCP packet types is
OPTIONAL, unless mandated by other parts of this specification.
Note that additional RTCP Packet types are used by the <xref
target="sec-profile">RTP/SAVPF Profile</xref> and the other <xref
target="sec-rtp-extn">RTCP extensions</xref>. WebRTC endpoints
that implement the SDP bundle negotiation extension will use the
SDP grouping framework 'mid' attribute to identify media streams.
Such endpoints MUST implement the RTCP SDES MID item described in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation"></xref>.</t>
<t>Support for multiple endpoints 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 (Section 6.3.6 of <xref
target="RFC3550"></xref>) and reverse reconsideration (Section
6.3.4 of <xref target="RFC3550"></xref>).</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
<xref target="RFC4566"></xref><xref target="RFC3556"></xref>.</t>
<t>Support for the reduced minimum RTCP reporting interval
described in Section 6.2 of <xref target="RFC3550"></xref>. When
using the reduced minimum RTCP reporting interval, the fixed
(non-reduced) minimum interval MUST be used when calculating the
participant timeout interval (see Sections 6.2 and 6.3.5 of <xref
target="RFC3550"></xref>). The delay before sending the initial
compound RTCP packet can be set to zero (see Section 6.2 of <xref
target="RFC3550"></xref> as updated by <xref
target="I-D.ietf-avtcore-rtp-multi-stream"></xref>).</t>
<t>Support for discontinuous transmission. RTP allows endpoints to
pause and resume transmission at any time. When resuming, the RTP
sequence number will increase by one, as usual, while the increase
in the RTP timestamp value will depend on the duration of the
pause. Discontinuous transmission is most commonly used with some
audio payload formats, but is not audio specific, and can be used
with any RTP payload format.</t>
<t>Ignore unknown RTCP packet types and RTP header extensions.
This is to ensure robust handling of future extensions, middlebox
behaviours, etc., that can result in not signalled RTCP packet
types or RTP header extensions being received. If a compound RTCP
packet is received that contains a mixture of known and unknown
RTCP packet types, the known packets types need to be processed as
usual, with only the unknown packet types being discarded.</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"></xref>.</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"></xref>, MUST be implemented. The RTP/SAVPF profile
is the combination of 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"></xref>
are needed for the improved RTCP timer model. This allows more
flexible transmission of RTCP packets in response to events, rather
than strictly according to bandwidth, and is vital for being able to
report congestion signals as well as media events. These extensions
also allow saving RTCP bandwidth, and an endpoint will commonly only
use the full RTCP bandwidth allocation if there are many events that
require feedback. The timer rules are also needed to make use of the
RTP conferencing extensions discussed in <xref
target="conf-ext"></xref>.</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 RTP/AVP or RTP/SAVP profile, given some constraints on
parameter configuration such as the RTCP bandwidth value and
"trr-int" (the most important factor for interworking with
RTP/(S)AVP endpoints via a gateway is to set the trr-int parameter
to a value representing 4 seconds, see Section 6.1 in <xref
target="I-D.ietf-avtcore-rtp-multi-stream"></xref>).</t>
</list></t>
<t>The secure RTP (SRTP) profile extensions <xref
target="RFC3711"></xref> are needed to provide media encryption,
integrity protection, replay protection and a limited form of source
authentication. WebRTC Endpoints 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 (i.e., implementations MUST use SRTP and SRTCP). The
RTP/SAVPF profile MUST be configured using the cipher suites,
DTLS-SRTP protection profiles, keying mechanisms, and other parameters
described in <xref target="I-D.ietf-rtcweb-security-arch"></xref>.</t>
</section>
<section anchor="sec.codecs" title="Choice of RTP Payload Formats">
<t>Mandatory to implement audio codecs and RTP payload formats for
WebRTC endpoints are defined in <xref
target="I-D.ietf-rtcweb-audio"></xref>. Mandatory to implement video
codecs and RTP payload formats for WebRTC endpoints are defined in
<xref target="I-D.ietf-rtcweb-video"></xref>. WebRTC endpoints MAY
additionally implement any other codec for which an RTP payload format
and associated signalling has been defined.</t>
<t>WebRTC Endpoints 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, along with any other SDP attributes needed to configure
the RTP payload format.</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"></xref>,
the RTP payload type number is sometimes used to associate an RTP
packet stream with a signalling context. This association is possible
provided unique RTP payload type numbers are used in each context. For
example, an RTP packet stream can be associated with an SDP "m=" line
by comparing the RTP payload type numbers used by the RTP packet
stream with payload types signalled in the "a=rtpmap:" lines in the
media sections of the SDP. This leads to the following
considerations:<list style="empty">
<t>If RTP packet 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.</t>
<t>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.</t>
<t>If the RTP payload type number is not being used to associate
RTP packet 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>
</list>A single RTP payload type number MUST NOT be assigned to
different RTP payload formats, or different configurations of the same
RTP payload format, within a single RTP session (note that the "m="
lines in an <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation">SDP
bundle group</xref> form a single RTP session).</t>
<t>An endpoint that has signalled support for multiple RTP payload
formats MUST be able to 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"></xref>. To
support rapid rate adaptation by changing codec, RTP does not require
advance signalling for changes between RTP payload formats used by a
single SSRC that were signalled during session set-up.</t>
<t>If performing changes between two RTP payload types that use
different RTP clock rates, an RTP sender MUST follow the
recommendations in Section 4.1 of <xref target="RFC7160"></xref>. RTP
receivers MUST follow the recommendations in Section 4.3 of <xref
target="RFC7160"></xref> 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 endpoints communicating using RTP
is known as an RTP session <xref target="RFC3550"></xref>. An endpoint
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 a different transport-layer flow for the
video). WebRTC Endpoints are REQUIRED to implement support for
multimedia sessions in this way, separating each RTP session using
different transport-layer flows for compatibility with legacy systems
(this is sometimes called session multiplexing).</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 packet 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"></xref>). Implementations are
therefore also REQUIRED to support transport of all RTP packet
streams, independent of media type, in a single RTP session using a
single transport layer flow, according to <xref
target="I-D.ietf-avtcore-multi-media-rtp-session"></xref> (this is
sometimes called SSRC multiplexing). If multiple types of media are to
be used in a single RTP session, all participants in that RTP session
MUST agree to this usage. In an SDP context, <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation"></xref> can be used to
signal such a bundle of RTP packet streams forming a single RTP
session.</t>
<t>Further discussion about the suitability of different RTP session
structures and multiplexing methods to different scenarios can be
found in <xref
target="I-D.ietf-avtcore-multiplex-guidelines"></xref>.</t>
</section>
<section anchor="sec.rtcp-mux" title="RTP and RTCP Multiplexing">
<t>Historically, RTP and RTCP have been run on separate transport
layer flows (e.g., two UDP ports for each RTP session, one port for
RTP and one port for RTCP). With the increased use of Network
Address/Port Translation (NAT/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,
implementations are REQUIRED to support multiplexing RTP data packets
and RTCP control packets on a single transport-layer flow <xref
target="RFC5761"></xref>. Such RTP and RTCP multiplexing MUST be
negotiated in the signalling channel before it is used. If SDP is used
for signalling, this negotiation MUST use the attributes defined in
<xref target="RFC5761"></xref>. For backwards compatibility,
implementations are also REQUIRED to support RTP and RTCP sent on
separate transport-layer flows.</t>
<t>Note that the use of RTP and RTCP multiplexed onto a single
transport-layer flow 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 <xref target="RFC6263"></xref>.</t>
</section>
<section title="Reduced Size RTCP">
<t>RTCP packets are usually sent as compound RTCP packets, and <xref
target="RFC3550"></xref> 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"></xref> 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"></xref> 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.
Implementations MUST support sending and receiving non-compound RTCP
feedback packets <xref target="RFC5506"></xref>. Use of non-compound
RTCP packets MUST be negotiated using the signalling channel. If SDP
is used for signalling, this negotiation MUST use the attributes
defined in <xref target="RFC5506"></xref>. 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 reason for using symmetric RTP is primarily to avoid
issues with NATs and Firewalls by ensuring that the send and receive
RTP packet streams, as well as RTCP, are actually bi-directional
transport-layer flows. This will keep alive the NAT and firewall
pinholes, and help indicate consent that the receive direction is a
transport-layer flow the intended recipient actually wants. In
addition, it saves resources, specifically ports at the endpoints, but
also in the network as NAT mappings or firewall state is not
unnecessary bloated. The amount of per flow QoS state kept in the
network is also 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. If SDP is used, this MUST
be done using the "a=ssrc:" SDP attribute defined in Section 4.1 and
Section 5 of <xref target="RFC5576"></xref> and the "previous-ssrc"
source attribute defined in Section 6.2 of <xref
target="RFC5576"></xref>; other per-SSRC attributes defined in <xref
target="RFC5576"></xref> MAY be supported.</t>
<t>While support for signalled SSRC identifiers is mandated, their use
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"></xref>. When using signalled SSRC
values, collision detection MUST be performed as described in Section
5 of <xref target="RFC5576"></xref>.</t>
<t>It is often desirable to associate an RTP packet stream with a
non-RTP context. For users of the WebRTC API a mapping between SSRCs
and MediaStreamTracks are provided per <xref
target="sec-webrtc-api"></xref>. For gateways or other usages it is
possible to associate an RTP packet 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 packet
stream are often sufficient to associate that packet stream with a
signalling context (e.g., if RTP payload type numbers are assigned as
described in <xref target="sec.codecs"></xref> of this memo, the RTP
payload types used by an RTP packet stream can be compared with values
in SDP "a=rtpmap:" lines, which are at the media level in SDP, 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 for the duration
of a <xref target="W3C.WD-webrtc-20130910">RTCPeerConnection</xref>,
so that RTP endpoints can be uniquely identified and associated with
their RTP packet streams within a set of related RTP sessions.</t>
<t>Each RTP endpoint MUST have at least one RTCP CNAME, and that RTCP
CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs
identify a particular synchronisation context, i.e., all SSRCs
associated with a single RTCP CNAME share a common reference clock. If
an endpoint has SSRCs that are associated with several unsynchronised
reference clocks, and hence different synchronisation contexts, it
will need to use multiple RTCP CNAMEs, one for each synchronisation
context.</t>
<t>Taking the discussion in <xref target="sec-webrtc-api"></xref> into
account, a WebRTC Endpoint MUST NOT use more than one RTCP CNAME in
the RTP sessions belonging to single RTCPeerConnection (that is, an
RTCPeerConnection forms a synchronisation context). RTP middleboxes
MAY generate RTP packet streams associated with more than one RTCP
CNAME, to allow them to avoid having to resynchronize media from
multiple different endpoints part of a multi-party RTP session.</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 <xref
target="sec-security">privacy viewpoint</xref>. Accordingly, a WebRTC
Endpoint MUST generate a new, unique, short-term persistent RTCP CNAME
for each RTCPeerConnection, following <xref target="RFC7022"></xref>,
with a single exception; if explicitly requested at creation an
RTCPeerConnection MAY use the same CNAME as as an existing
RTCPeerConnection within their common same-origin context.</t>
<t>An WebRTC Endpoint 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 anchor="sec-leap-sec" title="Handling of Leap Seconds">
<t>The guidelines regarding handling of leap seconds to limit their
impact on RTP media play-out and synchronization given in <xref
target="RFC7164"></xref> SHOULD be followed.</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 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 WebRTC.</t>
<section anchor="conf-ext"
title="Conferencing Extensions and Topologies">
<t>RTP is a protocol that inherently supports group communication.
Groups can be implemented by having each endpoint send its RTP packet
streams to an RTP middlebox that redistributes the traffic, by using a
mesh of unicast RTP packet streams between endpoints, or by using an
IP multicast group to distribute the RTP packet streams. These
topologies can be implemented in a number of ways as discussed in
<xref target="I-D.ietf-avtcore-rtp-topologies-update"></xref>.</t>
<t>While the use of IP multicast groups is popular in IPTV systems,
the topologies based on RTP middleboxes are dominant in interactive
video conferencing environments. Topologies based on a mesh of unicast
transport-layer flows to create a common RTP session have not seen
widespread deployment to date. Accordingly, WebRTC Endpoints are not
expected to support topologies based on IP multicast groups or to
support mesh-based topologies, such as a point-to-multipoint mesh
configured as a single RTP session (Topo-Mesh in the terminology of
<xref target="I-D.ietf-avtcore-rtp-topologies-update"></xref>).
However, a point-to-multipoint mesh constructed using several RTP
sessions, implemented in WebRTC using independent <xref
target="W3C.WD-webrtc-20130910">RTCPeerConnections</xref>, can be
expected to be used in WebRTC, and needs to be supported.</t>
<t>WebRTC Endpoints implemented according to this memo are expected to
support all the topologies described in <xref
target="I-D.ietf-avtcore-rtp-topologies-update"></xref> where the RTP
endpoints send and receive unicast RTP packet streams to and from some
peer device, provided that peer can participate in performing
congestion control on the RTP packet streams. The peer device could be
another RTP endpoint, or it could be an RTP middlebox that
redistributes the RTP packet streams to other RTP endpoints. This
limitation means that some of the RTP middlebox-based topologies are
not suitable for use in WebRTC. Specifically: <list style="symbols">
<t>Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be
used, since they make the use of RTCP for congestion control and
quality of service reports problematic (see Section 3.8 of <xref
target="I-D.ietf-avtcore-rtp-topologies-update"></xref>).</t>
<t>The Relay-Transport Translator (Topo-PtM-Trn-Translator)
topology SHOULD NOT be used because its safe use requires a
congestion control algorithm or RTP circuit breaker that handles
point to multipoint, which has not yet been standardised.</t>
</list></t>
<t>The following topology can be used, however it has some issues
worth noting:<list style="symbols">
<t>Content modifying MCUs with RTCP termination
(Topo-RTCP-terminating-MCU) MAY be used. Note that in this RTP
Topology, RTP loop detection and identification of active senders
is the responsibility of the WebRTC application; since the clients
are isolated from each other at the RTP layer, RTP cannot assist
with these functions (see section 3.9 of <xref
target="I-D.ietf-avtcore-rtp-topologies-update"></xref>).</t>
</list></t>
<t>The RTP extensions described in <xref target="sec-fir"></xref> to
<xref target="sec.tmmbr"></xref> are designed to be used with
centralised conferencing, where an RTP middlebox (e.g., a conference
bridge) receives a participant's RTP packet 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 of
these extensions are mandatory to be supported by WebRTC
Endpoints.</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 memo on <xref
target="RFC5104">Codec Control Messages (CCM) in RTP/AVPF</xref>; they
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 message is defined in Sections 3.5.1 and
4.3.1 of the <xref target="RFC5104">Codec Control Messages</xref>.
It 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
Endpoints that are sending media MUST understand and react to FIR
feedback messages they receive, since this greatly improves the user
experience when using centralised mixer-based conferencing. Support
for sending FIR messages is OPTIONAL.</t>
</section>
<section title="Picture Loss Indication (PLI)">
<t>The Picture Loss Indication message 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 Endpoints that
are sending media MUST understand and react to PLI feedback messages
as a loss tolerance mechanism. Receivers MAY send PLI messages.</t>
</section>
<section title="Slice Loss Indication (SLI)">
<t>The Slice Loss Indication message 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. It is RECOMMENDED that receivers
generate SLI feedback messages if slices are lost when using a codec
that supports the concept of macro blocks. A sender that receives an
SLI feedback message SHOULD attempt to repair the lost slice(s).</t>
</section>
<section title="Reference Picture Selection Indication (RPSI)">
<t>Reference Picture Selection Indication (RPSI) messages are
defined in Section 6.3.3 of the <xref target="RFC4585">RTP/AVPF
profile </xref>. Some video encoding standards allow the use of
older reference pictures than the most recent one for predictive
coding. If such a codec is in use, and if the encoder has learnt
that encoder-decoder synchronisation has been lost, then a known as
correct reference picture can be used as a base for future coding.
The RPSI message allows this to be signalled. Receivers that detect
that encoder-decoder synchronisation has been lost SHOULD generate
an RPSI feedback message if codec being used supports reference
picture selection. A RTP packet stream sender that receives such an
RPSI message SHOULD act on that messages to change the reference
picture, if it is possible to do so within the available bandwidth
constraints, and with the codec being used.</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"></xref>. 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>The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1
of the <xref target="RFC5104">Codec Control Messages</xref>. This
request 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. There 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 Endpoints that are
sending media 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 WebRTC, but if they are
used, they MUST be formatted and signalled following the general
mechanism for RTP header extensions defined in <xref
target="RFC5285"></xref>, since this gives well-defined semantics to
RTP header extensions.</t>
<t>As noted in <xref target="RFC5285"></xref>, the requirement from
the RTP specification that header extensions are "designed so that the
header extension may be ignored" <xref target="RFC3550"></xref>
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 of RTP header
extensions might include metadata that is additional to the usual RTP
information, but that can safely be ignored without compromising
interoperability.</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"></xref> be implemented in addition to RTCP SR-based
synchronisation.</t>
<t>This header extension uses the <xref target="RFC5285"></xref>
generic header extension framework, and so needs to be negotiated
before it can be used.</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 an endpoint to
inform a mixer about the level of audio activity in the packet to
which the header is attached. This enables an RTP middlebox to make
mixing or selection decisions without decoding or detailed
inspection of the payload, reducing the complexity in some types of
mixers. It can also save decoding resources in receivers, which can
choose to decode only the most relevant RTP packet streams based on
audio activity levels.</t>
<t>The <xref target="RFC6464">Client-to-Mixer Audio Level</xref>
header extension MUST be implemented. It is REQUIRED that
implementations are capable of encrypting the header extension
according to <xref target="RFC6904"></xref> since the information
contained in these header extensions can be considered sensitive.
The use of this encryption is RECOMMENDED, however usage of the
encryption can be explicitly disabled through API or signalling.</t>
<t>This header extension uses the <xref target="RFC5285"></xref>
generic header extension framework, and so needs to be negotiated
before it can be used.</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 an endpoint with the audio level of the
different sources mixed into a common source stream by a RTP mixer.
This enables a user interface 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 optimisation of non
critical functions, and is hence OPTIONAL to implement. If this
header extension is implemented, it is REQUIRED that implementations
are capable of encrypting the header extension according to <xref
target="RFC6904"></xref> since the information contained in these
header extensions can be considered sensitive. It is further
RECOMMENDED that this encryption is used, unless the encryption has
been explicitly disabled through API or signalling.</t>
<t>This header extension uses the <xref target="RFC5285"></xref>
generic header extension framework, and so needs to be negotiated
before it can be used.</t>
</section>
<section anchor="sec-mid" title="Media Stream Identification">
<t>WebRTC endpoints that implement the SDP bundle negotiation
extension will use the SDP grouping framework 'mid' attribute to
identify media streams. Such endpoints MUST implement the RTP MID
header extension described in <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation"></xref>.</t>
<t>This header extension uses the <xref target="RFC5285"></xref>
generic header extension framework, and so needs to be negotiated
before it can be used.</t>
</section>
<section anchor="sec-cvo" title="Coordination of Video Orientation">
<t>WebRTC endpoints that send or receive video MUST implement the
coordination of video orientation (CVO) RTP header extension as
described in Section 4 of <xref
target="I-D.ietf-rtcweb-video"></xref>.</t>
<t>This header extension uses the <xref target="RFC5285"></xref>
generic header extension framework, and so needs to be negotiated
before it can be used.</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 packet streams robust against
packet loss and reduce the impact of loss on media quality. However,
they generally add some overhead compared to a non-robust stream. The
overhead needs to be considered, and the aggregate bit-rate MUST be rate
controlled to avoid causing network congestion (see <xref
target="sec-rate-control"></xref>). 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 send negative acknowledgements (NACKs) for RTP
data packets <xref target="RFC4585"></xref>. 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.</t>
<t>RTP packet stream senders are REQUIRED to understand the Generic
NACK message defined in Section 6.2.1 of <xref
target="RFC4585"></xref>, but MAY choose to ignore some or all of this
feedback (following Section 4.2 of <xref target="RFC4585"></xref>).
Receivers MAY send NACKs for missing RTP packets. Guidelines on when
to send NACKs are provided in <xref target="RFC4585"></xref>. 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"></xref>). The use of
retransmissions can also increase the forward RTP bandwidth, and can
potentially caused increased packet loss if the original packet loss
was caused by network congestion. 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"></xref> sent using SSRC multiplexing,
and MAY also support RTP retransmission packets sent using session
multiplexing. Senders MAY send RTP retransmission packets in response
to NACKs if support for the RTP retransmission payload format has been
negotiated, and if the sender believes it is useful to send a
retransmission of the packet(s) referenced in the NACK. Senders do 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 FEC schemes and their parameters.</t>
<t>WebRTC endpoints MUST follow the recommendations for FEC use given
in <xref target="I-D.ietf-rtcweb-fec"></xref>. WebRTC endpoints MAY
support other types of FEC, but these MUST be negotiated before they
are used.</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 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 endpoints can send one or more RTP packet streams to each
participant, and there can be several participants. Each of these RTP
packet streams can contain different types of media, and the type of
media, bit rate, and number of RTP packet streams as well as
transport-layer flows can be highly asymmetric. Non-RTP traffic can
share the network paths with RTP transport-layer 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 is very dependent on
effective adaptation of the media to the limitations of the network.
Endpoints 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's flows. Some
requirements for congestion control algorithms for RTCPeerConnections
are discussed in <xref target="I-D.ietf-rmcat-cc-requirements"></xref>.
If a standardized congestion control algorithm that satisfies these
requirements is developed in the future, this memo will need to be be
updated to mandate its use.</t>
<section title="Boundary Conditions and Circuit Breakers">
<t>WebRTC Endpoints MUST implement the RTP circuit breaker algorithm
that is described in <xref
target="I-D.ietf-avtcore-rtp-circuit-breakers"></xref>. 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. The congestion control
algorithm will have to be proprietary until a standardized congestion
control algorithm is available. 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 packetisation choices that exist. In addition, the
signalling channel can establish maximum media bit-rate boundaries
using, for example, 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"></xref> of this memo). Signalled bandwidth
limitations, such as SDP "b=AS:" or "b=CT:" lines received from the
peer, MUST be followed when sending RTP packet streams. A WebRTC
Endpoint receiving media SHOULD signal its bandwidth limitations.
These limitations have to be based on known bandwidth limitations, for
example the capacity of the edge links.</t>
</section>
<section title="Congestion Control Interoperability and Legacy Systems">
<t>All endpoints that wish to interwork with WebRTC MUST implement
RTCP and provide congestion feedback via the defined RTCP reporting
mechanisms.</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 endpoints 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 endpoint supports RTP/AVPF, this enables negotiation of
important parameters for frequent reporting, such as the "trr-int"
parameter, and the possibility that the endpoint supports some useful
feedback format for congestion control purpose such as <xref
target="RFC5104"> TMMBR</xref>. Implementations that have to interwork
with such endpoints 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 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"></xref>, implementations
are REQUIRED to generate RTCP Sender Report (SR) and Reception Report
(RR) packets relating to the RTP packet 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, see <xref
target="RFC3611"></xref><xref target="RFC6792"></xref>. At the time of
this writing, it is not clear what extended metrics are suitable for use
in WebRTC, 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
RTCP XR packets SHOULD be signalled; implementations MUST ignore RTCP XR
packets that are unexpected or not understood.</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. In this case, future updates to this memo have to be made
following the <xref target="RFC2736"> Guidelines for Writers of RTP
Payload Format Specifications </xref>, <xref
target="I-D.ietf-payload-rtp-howto">How to Write an RTP Payload
Format</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. Interworking functions might transform this into the
RTP/SAVP profile for a legacy use case, by indicating to the WebRTC
Endpoint that the RTP/SAVPF is used and configuring 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
<xref target="RFC5245">ICE</xref> 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, i.e. transport-layer flow, is used for RTP
and RTCP flows, this MUST be signalled (see <xref
target="sec.rtcp-mux"></xref>).</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"></xref> of
this memo discusses requirements for uniqueness of payload
types.</t>
<t hangText="RTP Extensions:">The use of any additional RTP header
extensions and RTCP packet types, including any necessary
parameters, MUST be signalled. This signalling is to ensure that a
WebRTC Endpoint's behaviour, especially when sending, of any
extensions is predictable and consistent. For robustness, and for
compatibility with non-WebRTC systems that might be connected to a
WebRTC session via a gateway, implementations are REQUIRED to ignore
unknown RTCP packets and RTP header extensions (see also <xref
target="sec-rtp-rtcp"></xref>).</t>
<t hangText="RTCP Bandwidth:">Support for exchanging RTCP Bandwidth
values to the endpoints 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> if using SDP, or something semantically
equivalent. This also ensures that the endpoints have a common view
of the RTCP bandwidth. A common view of the RTCP bandwidth among
different endpoints is important, to prevent differences in RTCP
packet timing and timeout intervals causing interoperability
problems.</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. Note that in WebRTC
it will depend on the signalling model and API how these parameters need
to be configured but they will be need to either be 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 might need to be synchronized during play back.</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.</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. If the encoding parameters and constraints are the same, an
implementation could choose to use only one encoded stream to create the
different RTP packet streams. Note that such optimisations would need to
take into account that the constraints for one of the MediaStreamTracks
can at any moment change, meaning that the encoding configurations might
no longer be identical and two different encoder instances would then be
needed.</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 does not force an endpoint to disrupt the media by changing
synchronisation base and CNAME during delivery of any ongoing packet
streams, all MediaStreamTracks and their associated SSRCs originating
from the same endpoint need to be sent using the same CNAME within one
RTCPeerConnection. This is motivating the use of a single CNAME in <xref
target="sec-cname"></xref>. <list style="empty">
<t>The requirement on using the same CNAME for all SSRCs that
originate from the same endpoint, does not require a middlebox that
forwards traffic from multiple endpoints to only use a single
CNAME.</t>
</list></t>
<t>Different CNAMEs normally need to be used for different
RTCPeerConnection instances, as specified in <xref
target="sec-cname"></xref>. Having two communication sessions with the
same CNAME could enable tracking of a user or device across different
services (see Section 4.4.1 of <xref
target="I-D.ietf-rtcweb-security"></xref> for details). A web
application can request that the CNAMEs used in different
RTCPeerConnections (within a same-orign context) be the same, this
allows for synchronization of the endpoint's RTP packet streams across
the different RTCPeerConnections.<list style="empty">
<t>Note: this doesn't result in a tracking issue, since the creation
of matching CNAMEs depends on existing tracking within a single
origin.</t>
</list>The above will currently force a WebRTC Endpoint that receives
a MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing
on any RTCPeerConnection to perform resynchronisation of the stream.
Since the sending party needs to change the CNAME to the one it uses,
this implies it has to use a local system clock as timebase for the
synchronisation. Thus, the relative relation between the timebase of the
incoming stream and the system sending out needs to be 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 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 are
clearly possible and implementation specific.</t>
<t>A WebRTC Endpoint MUST support receiving multiple MediaStreamTracks,
where each of the 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
is to allow for forward compatibility with any future changes that
enable 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><xref target="I-D.ietf-rtcweb-jsep">Javascript Session Establishment
Protocol</xref> specifies that 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">The MSID document</xref> also defines, in
section 4.1, how to map unknown source packet stream SSRCs to
MediaStreamTracks and MediaStreams. This later is relevant to handle
some cases of legacy interoperability. 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>
<t>Finally this specification puts a requirement on the WebRTC API to
realize a method for determining the <xref target="sec-rtp-rtcp">CSRC
list</xref> as well as the <xref
target="sec-mixer-to-client">Mixer-to-Client audio levels</xref> (when
supported) and the basic requirements for this is further discussed in
<xref target="sec-media-stream-id"></xref>.</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
Endpoint 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 Endpoint will be a simultaneous participant in one or more
RTP sessions. Each RTP session can convey multiple media sources, and
can include media data from multiple endpoints. In the following, some
ways in which WebRTC Endpoints can configure and use RTP sessions are
outlined.</t>
<section anchor="sec.multiple-flows"
title="Use of Multiple Media Sources Within an RTP Session">
<t>RTP is a group communication protocol, and every RTP session can
potentially contain multiple RTP packet streams. 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 source
(e.g., one RTP session for audio sources and one for video
sources, each sent over different transport layer flows).
However, to reduce the number of UDP ports used, the default in
WebRTC is to send all types of media in a single RTP session, as
described in <xref target="sec.session-mux"></xref>, using RTP
and RTCP multiplexing (<xref target="sec.rtcp-mux"></xref>) to
further reduce the number of UDP ports needed. This RTP session
then uses only one bi-directional transport-layer flow, but will
contain multiple RTP packet streams, each containing a different
type of media. A common example might be an endpoint with a
camera and microphone that sends two RTP packet streams, one
video and one audio, into a single RTP session.</t>
<t hangText="Multiple Capture Devices:">A WebRTC Endpoint might
have multiple cameras, microphones, or other media capture
devices, and so might want to generate several RTP packet
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
endpoint sending multiple RTP packet streams of the same media
type into a single RTP session at the same time.</t>
<t hangText="Associated Repair Data:">An endpoint might send a
RTP packet stream that is somehow associated with another
stream. For example, it might send an RTP packet stream that
contains FEC or retransmission data relating to another stream.
Some RTP payload formats send this sort of associated repair
data as part of the source packet stream, while others send it
as a separate packet stream.</t>
<t hangText="Layered or Multiple Description Coding:">An
endpoint can use a layered media codec, for example H.264 SVC,
or a multiple description codec, that generates multiple RTP
packet streams, 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 endpoint and some other peer device,
where those devices share a common SSRC space. The peer device
might be another WebRTC Endpoint, or it might be an RTP mixer,
translator, or some other form of media processing middlebox. In
the latter cases, the middlebox might send mixed or relayed RTP
streams from several participants, that the WebRTC Endpoint will
need to render. Thus, even though a WebRTC Endpoint might only
be a member of a single RTP session, the peer device might be
extending that RTP session to incorporate other endpoints.
WebRTC is a group communication environment and endpoints need
to be capable of receiving, decoding, and playing out multiple
RTP packet streams at once, even in a single RTP session.</t>
</list></t>
</section>
<section anchor="sec.multiple-sessions"
title="Use of Multiple RTP Sessions">
<t>In addition to sending and receiving multiple RTP packet streams
within a single RTP session, a WebRTC Endpoint might participate in
multiple RTP sessions. There are several reasons why a WebRTC
Endpoint 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 separate
transport layer flow, for video. All WebRTC Endpoints need to
support the option of sending different types of media on
different RTP sessions, so they can interwork with such legacy
devices. This is discussed further in <xref
target="sec.session-mux"></xref>.</t>
<t hangText="To provide enhanced quality of service:">Some
network-based quality of service mechanisms operate on the
granularity of transport layer flows. If it is desired to use
these mechanisms to provide differentiated quality of service
for some RTP packet streams, then those RTP packet streams need
to be sent in a separate RTP session using a different
transport-layer flow, and with appropriate quality of service
marking. This is discussed further in <xref
target="sec-differentiated"></xref>.</t>
<t hangText="To separate media with different purposes:">An
endpoint might want to send RTP packet streams that have
different purposes on different RTP sessions, to make it easy
for the peer device to distinguish them. For example, some
centralised multiparty conferencing systems display the active
speaker in high resolution, but show low resolution "thumbnails"
of other participants. Such systems might configure the
endpoints to send simulcast high- and low-resolution versions of
their video using separate RTP sessions, to simplify the
operation of the RTP middlebox. In the WebRTC context this is
currently possible by establishing multiple WebRTC
MediaStreamTracks that have the same media source in one (or
more) RTCPeerConnection. Each MediaStreamTrack is then
configured to deliver a particular media quality and thus media
bit-rate, and will produce an independently encoded version with
the codec parameters agreed specifically in the context of that
RTCPeerConnection. The RTP middlebox can distinguish packets
corresponding to the low- and high-resolution streams by
inspecting their SSRC, RTP payload type, or some other
information contained in RTP payload, RTP header extension or
RTCP packets, but it can be easier to distinguish the RTP packet
streams if they arrive on separate RTP sessions on separate
transport-layer flows.</t>
<t hangText="To directly connect with multiple peers:">A
multi-party conference does not need to use an RTP middlebox.
Rather, a multi-unicast mesh can be created, comprising several
distinct RTP sessions, with each participant sending RTP traffic
over a separate RTP session (that is, using an independent
RTCPeerConnection object) to every other participant, as shown
in <xref target="fig-mesh"></xref>. This topology has the
benefit of not requiring an RTP middlebox node that is trusted
to access and manipulate the media data. The downside is that it
increases the used bandwidth at each sender by requiring one
copy of the RTP packet streams for each participant that 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 of
the relationship between RTP sessions and RTCPeerConnection
objects it is recommend that this is implemented as several
individual RTP sessions. The only downside is that endpoint 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 if 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 be presented to 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 is 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"></xref> outlines a simple topology that
might be used in a four-person centralised conference. The
middlebox acts to optimise the transmission of RTP packet
streams from certain perspectives, either by only sending some
of the received RTP packet stream to any given receiver, or by
providing a combined RTP packet stream out of a set of
contributing streams.</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 endpoints in one multi-party session. Other
types of middlebox might use separate RTP sessions between each
endpoint and the middlebox. A common aspect is that these RTP
middleboxes can use a number of tools to control the media
encoding provided by a WebRTC Endpoint. This includes functions
like requesting the breaking of 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 has the responsibility to
correctly perform congestion control, source identification,
manage synchronisation while providing the application with
suitable media optimisations. The middlebox 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 endpoint, before sending it on towards the endpoint(s),
thus they need to be able to decrypt and then re-encrypt the RTP
packet stream before sending it out.</t>
<t>RTP Mixers can create a situation where an endpoint
experiences a situation in-between a session with only two
endpoints and multiple RTP sessions. Mixers are expected to not
forward RTCP reports regarding RTP packet streams across
themselves. This is due to the difference in the RTP packet
streams provided to the different endpoints. 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 endpoint's feedback or requests go 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 to 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, endpoints 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
endpoint. In RTP sessions where multiple endpoints are directly
visible to an endpoint, all endpoints will have knowledge about
each others' master keys, and can thus inject packets claimed to
come from another endpoint 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
endpoints 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 endpoint that receives an RTP packet
stream to be able to forward that RTP packet 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 endpoint that is performing the forwarding is responsible
for producing an RTP packet 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 endpoint 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 endpoint to optimise 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 RTP packet stream generated by the forwarding device.</t>
</list></t>
</section>
<section anchor="sec-differentiated"
title="Differentiated Treatment of RTP Packet Streams">
<t>There are use cases for differentiated treatment of RTP packet
streams. Such differentiation can happen at several places in the
system. First of all is the prioritization within the endpoint
sending the media, which controls, both which RTP packet 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 <xref
target="W3C.WD-webrtc-20130910">WebRTC API</xref> 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 packet streams. Any changes in relative priority will also
need to be considered for RTP packet streams that are associated
with the main RTP packet streams, such as redundant streams for RTP
retransmission and FEC. The importance of such redundant RTP packet
streams 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 redundant RTP packet stream
as for the source RTP packet stream.</t>
<t>Secondly, the network can prioritize transport-layer flows and
sub-flows, including RTP packet 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 consisting of the actual mechanism to prioritize packets.
Three common methods for classifying IP packets are: <list
style="hanging">
<t hangText="DiffServ:">The endpoint 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 transport-layer 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 RTP packet streams used in a WebRTC session.
The use of flow-based differentiation needs to be coordinated
between the WebRTC system and the network(s). The WebRTC endpoint
needs to know that flow-based differentiation might be used to
provide the separation of the RTP packet streams onto different UDP
flows to enable a more granular usage of flow based differentiation.
The used flows, their 5-tuples and prioritization will need to be
communicated to the network so that it can identify the flows
correctly to enable prioritization. No specific protocol support for
this is specified.</t>
<t>DiffServ assumes that either the endpoint or a classifier can
mark the packets with an appropriate DSCP so that the packets are
treated according to that marking. If the endpoint is to mark the
traffic two requirements arise in the WebRTC context: 1) The WebRTC
Endpoint has to know which DSCP to use and that it can use them on
some set of RTP packet 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.ietf-tsvwg-rtcweb-qos">"DSCP
and other packet markings for RTCWeb QoS"</xref>.</t>
<t>Deep Packet Inspectors will, despite the SRTP media encryption,
still be fairly capable at classifying the RTP streams. The reason
is that SRTP leaves the first 12 bytes of the RTP header
unencrypted. This enables easy RTP stream identification using the
SSRC and provides the classifier with useful information that can be
correlated to determine for example the stream's media type. Using
packet sizes, reception times, packet inter-spacing, RTP timestamp
increments and sequence numbers, fairly reliable classifications are
achieved.</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. However, depending on the QoS
mechanism and what markings that are applied, this can result in not
only different packet drop probabilities but also packet reordering,
see <xref target="I-D.ietf-tsvwg-rtcweb-qos"></xref> and <xref
target="I-D.ietf-dart-dscp-rtp"></xref> for further discussion. As a
default policy all RTP packets related to a RTP packet 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 packet stream need to be marked. RTCP compound
packets with Sender Reports (SR), ought to be marked with the same
priority as the RTP packet stream itself, so the RTCP-based
round-trip time (RTT) measurements are done using the same
transport-layer flow priority as the RTP packet stream experiences.
RTCP compound packets containing RR packet ought to be sent with the
priority used by the majority of the RTP packet streams 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="Media Source, RTP Packet Streams, and Participant Identification">
<section anchor="sec-media-stream-id"
title="Media Source Identification">
<t>Each RTP packet stream is identified by a unique synchronisation
source (SSRC) identifier. The SSRC identifier is carried in each of
the RTP packets comprising a RTP packet 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"></xref>. The first
stage in demultiplexing RTP and RTCP packets received on a single
transport layer flow at a WebRTC Endpoint is to separate the RTP
packet 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
encoded streams from multiple media sources to form a new encoded
stream from a new media source (the mixer). The RTP packets in that
new RTP packet stream can include a Contributing Source (CSRC) list,
indicating which original SSRCs contributed to the combined source
stream. As described in <xref target="sec-rtp-rtcp"></xref>,
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 media 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 WebRTC
applications.</t>
<t>If the mixer-to-client audio level extension <xref
target="RFC6465"></xref> is being used in the session (see <xref
target="sec-mixer-to-client"></xref>), the information in the CSRC
list is augmented by audio level information for each contributing
source. It is desirable to expose this information to the WebRTC
application using some API, after mapping the CSRC values to WebRTC
MediaStream identities, so it can be exposed in the user
interface.</t>
</section>
<section title="SSRC Collision Detection">
<t>The RTP standard requires RTP implementations to have support for
detecting and handling SSRC collisions, i.e., resolve the conflict
when two different endpoints use the same SSRC value (see section
8.2 of <xref target="RFC3550"></xref>). This requirement also
applies to WebRTC Endpoints. There are several scenarios where SSRC
collisions can occur: <list style="symbols">
<t>In a point-to-point session where each SSRC is associated
with either of the two endpoints 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. If SDP is used, this information
is provided by <xref target="RFC5576">Source-Specific SDP
Attributes</xref>. Still, collisions can occur if both endpoints
start using a new SSRC identifier prior to having signalled it
to the peer and received acknowledgement on the signalling
message. The <xref target="RFC5576">Source-Specific SDP
Attributes</xref> contains a mechanism to signal how the
endpoint resolved the SSRC collision.</t>
<t>SSRC values that have not been signalled could also appear in
an RTP session. This is more likely than it appears, since some
RTP functions use extra SSRCs to provide their functionality.
For example, retransmission data might be transmitted using a
separate RTP packet stream that requires its own SSRC, separate
to the SSRC of the source RTP packet stream <xref
target="RFC4588"></xref>. In those cases, an endpoint 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>Multiple endpoints in a multiparty conference can create new
sources and signal those towards the RTP middlebox. In cases
where the SSRC/CSRC are propagated between the different
endpoints from the RTP middlebox collisions can occur.</t>
<t>An RTP middlebox could connect an endpoint's
RTCPeerConnection to another RTCPeerConnection from the same
endpoint, thus forming a loop where the endpoint will receive
its own traffic. While it is clearly considered a bug, it is
important that the endpoint 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>
</list></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,
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 endpoint 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 packet streams be indicated as coming from the same
synchronisation context and logical endpoint 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
packet 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"></xref>, and security
considerations for the WebRTC framework are described in <xref
target="I-D.ietf-rtcweb-security"></xref>. These considerations also
apply to this memo.</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. It
is not believed 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
and use media security solution is created by combining 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 RTP packet streams 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"></xref> of this memo mandates generation of
short-term persistent RTCP CNAMES, as specified in RFC7022, resulting in
untraceable CNAME values that alleviate this risk.</t>
<t>Some potential denial of service attacks exist if the RTCP reporting
interval is configured to an inappropriate value. This could be done by
configuring the RTCP bandwidth fraction to an excessively large or small
value using the SDP "b=RR:" or "b=RS:" lines <xref
target="RFC3556"></xref>, or some similar mechanism, or by choosing an
excessively large or small value for the RTP/AVPF minimal receiver
report interval (if using SDP, this is the "a=rtcp-fb:... trr-int"
parameter) <xref target="RFC4585"></xref>. The risks are as
follows:<list style="numbers">
<t>the RTCP bandwidth could be configured to make the regular
reporting interval so large that effective congestion control cannot
be maintained, potentially leading to denial of service due to
congestion caused by the media traffic;</t>
<t>the RTCP interval could be configured to a very small value,
causing endpoints to generate high rate RTCP traffic, potentially
leading to denial of service due to the non-congestion controlled
RTCP traffic; and</t>
<t>RTCP parameters could be configured differently for each
endpoint, with some of the endpoints using a large reporting
interval and some using a smaller interval, leading to denial of
service due to premature participant timeouts due to mismatched
timeout periods which are based on the reporting interval (this is a
particular concern if endpoints use a small but non-zero value for
the RTP/AVPF minimal receiver report interval (trr-int) <xref
target="RFC4585"></xref>, as discussed in Section 6.1 of <xref
target="I-D.ietf-avtcore-rtp-multi-stream"></xref>).</t>
</list>Premature participant timeout can be avoided by using the fixed
(non-reduced) minimum interval when calculating the participant timeout
(see <xref target="sec-rtp-rtcp"></xref> of this memo and Section 6.1 of
<xref target="I-D.ietf-avtcore-rtp-multi-stream"></xref>). To address
the other concerns, endpoints SHOULD ignore parameters that configure
the RTCP reporting interval to be significantly longer than the default
five second interval specified in <xref target="RFC3550"></xref> (unless
the media data rate is so low that the longer reporting interval roughly
corresponds to 5% of the media data rate), or that configure the RTCP
reporting interval small enough that the RTCP bandwidth would exceed the
media bandwidth.</t>
<t>The guidelines in <xref target="RFC6562"></xref> apply when using
variable bit rate (VBR) audio codecs such as Opus (see <xref
target="sec.codecs"></xref> for discussion of mandated audio codecs).
The guidelines in <xref target="RFC6562"></xref> also apply, but are of
lesser importance, when using the client-to-mixer audio level header
extensions (<xref target="sec-client-to-mixer"></xref>) or the
mixer-to-client audio level header extensions (<xref
target="sec-mixer-to-client"></xref>). The use of the encryption of the
header extensions are RECOMMENDED, unless there are known reasons, like
RTP middleboxes performing voice activity based source selection or
third party monitoring that will greatly benefit from the information,
and this has been expressed using API or signalling. If further evidence
are produced to show that information leakage is significant from audio
level indications, then use of encryption needs to be mandated at that
time.</t>
<t>In multi-party communication scenarios using RTP Middleboxes, a lot
of trust is placed on these middleboxes to preserve the sessions
security. The middlebox needs to maintain the confidentiality, integrity
and perform source authentication. As discussed in <xref
target="sec.multiple-flows"></xref> the middlebox can perform checks
that prevents any endpoint participating in a conference to impersonate
another. Some additional security considerations regarding multi-party
topologies can be found in <xref
target="I-D.ietf-avtcore-rtp-topologies-update"></xref>.</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 anchor="Acknowledgements" title="Acknowledgements">
<t>The authors would like to thank Bernard Aboba, Harald Alvestrand,
Cary Bran, Ben Campbell, Alissa Cooper, Spencer Dawkins, Charles Eckel,
Alex Eleftheriadis, Christian Groves, Chris Inacio, Cullen Jennings,
Olle Johansson, Suhas Nandakumar, Dan Romascanu, Jim Spring, Martin
Thomson, and the other members of the IETF RTCWEB working group for
their valuable feedback.</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.4566'?>
<?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.RFC.7160'?>
<?rfc include='reference.RFC.7164'?>
<?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-audio'?>
<?rfc include='reference.I-D.ietf-rtcweb-video'?>
<?rfc include='reference.I-D.ietf-rtcweb-security'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-circuit-breakers'?>
<?rfc include='reference.I-D.ietf-rtcweb-security-arch'?>
<?rfc include='reference.I-D.ietf-rtcweb-fec'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
<?rfc include='reference.I-D.ietf-rtcweb-overview'?>
<?rfc include='reference.I-D.ietf-avtcore-rtp-topologies-update'?>
<?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.4383'?>
<?rfc include='reference.RFC.5245'?>
<?rfc include='reference.RFC.5576'?>
<?rfc include='reference.RFC.5968'?>
<?rfc include='reference.RFC.6263'?>
<?rfc include='reference.RFC.6792'?>
<?rfc include='reference.RFC.7478'?>
<?rfc include='reference.I-D.ietf-mmusic-msid'?>
<?rfc include='reference.I-D.ietf-avtcore-multiplex-guidelines'?>
<?rfc include='reference.I-D.ietf-payload-rtp-howto'?>
<?rfc include='reference.I-D.ietf-rmcat-cc-requirements'?>
<?rfc include='reference.I-D.ietf-tsvwg-rtcweb-qos'?>
<?rfc include='reference.I-D.ietf-avtext-rtp-grouping-taxonomy'?>
<?rfc include='reference.I-D.ietf-dart-dscp-rtp'?>
<?rfc include='reference.I-D.ietf-rtcweb-jsep'?>
</references>
</back>
</rfc>
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