Media Transport and Use of RTP in WebRTCUniversity of GlasgowSchool of Computing ScienceGlasgowG12 8QQUnited Kingdomcsp@csperkins.orghttps://csperkins.org/EricssonTorshamnsgatan 23Kista164 80Swedenmagnus.westerlund@ericsson.comTechnical University MunichDepartment of InformaticsChair of Connected MobilityBoltzmannstrasse 3Garching85748Germanyott@in.tum.deThe framework for Web Real-Time Communication (WebRTC) 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.IntroductionThe Real-time Transport Protocol (RTP)
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 signaling, these form
the basis for many teleconferencing systems.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.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 .The structure of this memo is as follows. outlines our rationale for preparing this
memo and choosing these RTP features. defines terminology. Requirements for
core RTP protocols are described in ,
and suggested RTP extensions are described in .
outlines mechanisms that can increase robustness to network problems,
while describes
congestion control and rate adaptation mechanisms. The discussion of
mandated RTP
mechanisms concludes in with a review of
performance monitoring and network management tools. gives some guidelines for future incorporation
of other RTP and RTP Control Protocol (RTCP) extensions into this
framework. describes requirements
placed on the signaling channel.
discusses the relationship between features of the RTP framework and the
WebRTC application programming interface (API), and discusses RTP implementation
considerations. The memo concludes with security considerations and IANA considerations.RationaleThe 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 support
many new media encodings, but it 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 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.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 .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.Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED",
"MAY", and "OPTIONAL" in this document are to be interpreted as
described in BCP 14
when, and only when, they appear in all capitals, as shown here.
Lower- or mixed-case uses of
these key words are not to be interpreted as carrying special
significance in this memo.
We define the following additional terms:
WebRTC MediaStream:
The MediaStream concept defined by
the W3C in the WebRTC API. A
MediaStream consists of zero or more MediaStreamTracks.
MediaStreamTrack:
Part of the MediaStream concept
defined by the W3C in the WebRTC API. A
MediaStreamTrack is an individual stream of media from any type of
media source such as a microphone or a camera, but conceptual
sources such as an audio mix or a video composition are also possible.
Transport-layer flow:
A unidirectional flow of
transport packets that are identified by a particular 5-tuple
of source IP address, source port, destination IP address,
destination port, and transport protocol.
Bidirectional transport-layer flow:
A bidirectional
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.
This document uses the terminology from and . Other terms are used
according to their definitions from the RTP
specification. In particular, note the following frequently used
terms: RTP stream, RTP session, and endpoint.WebRTC Use of RTP: Core ProtocolsThe 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.RTP and RTCPThe Real-time Transport Protocol (RTP)
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.The following RTP and RTCP features are sometimes omitted in
limited-functionality implementations of RTP, but they are REQUIRED in all
WebRTC endpoints:
Support for use of multiple simultaneous synchronization source
(SSRC) values in a
single RTP session, including support for RTP endpoints that send
many SSRC values simultaneously, following and . The RTCP
optimizations for multi-SSRC sessions defined in MAY be supported; if supported, the usage MUST be signaled.
Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (see also ).
Support for reception of RTP data packets containing
contributing source (CSRC)
lists, as generated by RTP mixers, and RTCP packets relating to
CSRCs.
Sending correct synchronization information in the RTCP Sender
Reports, to allow receivers to implement lip synchronization; see
regarding support for the rapid
RTP synchronization extensions.
Support for multiple synchronization contexts. Participants
that send multiple simultaneous RTP packet streams SHOULD do so as
part of a single synchronization context, using a single RTCP
CNAME for all streams and allowing receivers to play the streams
out in a synchronized manner. For compatibility with potential
future versions of this specification, or for interoperability
with non-WebRTC devices through a gateway, receivers MUST support
multiple synchronization 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
streams in some circumstances; see .
Support for sending and receiving RTCP Sender Report (SR), Receiver Report (RR), Source Description (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 RTP/SAVPF profile and the other RTCP extensions. WebRTC endpoints
that implement the Session Description Protocol (SDP) bundle
negotiation extension will use the
SDP Grouping Framework "mid" attribute to identify media streams.
Such endpoints MUST implement the RTCP SDES media
identification (MID) item described in
.
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 randomized RTCP
transmission intervals to avoid synchronization of RTCP reports;
support for RTCP timer reconsideration () and
reverse reconsideration ().
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
.
Support for the reduced minimum RTCP reporting interval
described in . When
using the reduced minimum RTCP reporting interval, the fixed
(nonreduced) minimum interval MUST be used when calculating the
participant timeout interval (see Sections and of ). The delay before sending the
initial
compound RTCP packet can be set to zero (see as updated by ).
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 it is not audio specific and can be used
with any RTP payload format.
Ignore unknown RTCP packet types and RTP header extensions.
This is to ensure robust handling of future extensions, middlebox
behaviors, etc., that can result in receiving RTP header
extensions or RTCP packet types that were not signaled. If a compound RTCP
packet that contains a mixture of known and unknown
RTCP packet types is received, the known packet types need to be processed as
usual, with only the unknown packet types being discarded.
It is known that a significant number of legacy RTP
implementations, especially those targeted at systems with
only Voice over IP (VoIP), 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.Other implementation considerations are discussed in .Choice of the RTP ProfileThe complete specification of RTP for a particular application
domain requires the choice of an RTP profile. For WebRTC use, the
extended secure RTP profile for
RTCP-based feedback
(RTP/SAVPF), as extended by , MUST be implemented. The RTP/SAVPF profile
is the combination of the basic RTP/AVP
profile, the RTP profile for RTCP-based
feedback (RTP/AVPF), and the secure RTP
profile (RTP/SAVP).The RTCP-based feedback extensions
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 .The secure RTP (SRTP) profile extensions 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. In other words, implementations MUST use SRTP and Secure RTCP (SRTCP). The
RTP/SAVPF profile MUST be configured using the cipher suites,
DTLS-SRTP protection profiles, keying mechanisms, and other parameters
described in .Choice of RTP Payload FormatsMandatory-to-implement audio codecs and RTP payload formats for
WebRTC endpoints are defined in . Mandatory-to-implement video
codecs and RTP payload formats for WebRTC endpoints are defined in
. WebRTC endpoints MAY
additionally implement any other codec for which an RTP payload format
and associated signaling has been defined.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.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 ,
the RTP payload type number is sometimes used to associate an RTP
packet stream with a signaling 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 signaled in the "a=rtpmap:" lines in the
media sections of the SDP. This leads to the following
considerations:
If RTP packet streams are being associated with signaling
contexts based on the RTP payload type, then the assignment of RTP
payload type numbers MUST be unique across signaling
contexts.
If the same RTP payload format configuration is used in
multiple contexts, then a different RTP payload type number has to
be assigned in each context to ensure uniqueness.
If the RTP payload type number is not being used to associate
RTP packet streams with a signaling context, then the same RTP
payload type number can be used to indicate the exact same RTP
payload format configuration in multiple contexts.
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 SDP
BUNDLE group form a single RTP session).An endpoint that has signaled 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 signaled 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 signaled for the type of media that is being
sent by that source; see . To
support rapid rate adaptation by changing codecs, RTP does not require
advance signaling for changes between RTP payload formats used by a
single SSRC that were signaled during session setup.If performing changes between two RTP payload types that use
different RTP clock rates, an RTP sender MUST follow the
recommendations in . RTP
receivers MUST follow the recommendations in
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.Use of RTP SessionsAn association amongst a set of endpoints communicating using RTP
is known as an RTP session . 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).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
. 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 (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, the
mechanisms described in can be used to
signal such a bundle of RTP packet streams forming a single RTP
session.Further discussion about the suitability of different RTP session
structures and multiplexing methods to different scenarios can be
found in .RTP and RTCP MultiplexingHistorically, RTP and RTCP have been run on separate
transport-layer flows (e.g., two UDP ports for each RTP session, one
for RTP and one 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 setup times,
implementations are REQUIRED to support multiplexing RTP data packets
and RTCP control packets on a single transport-layer flow . Such RTP and RTCP multiplexing MUST be
negotiated in the signaling channel before it is used. If SDP is used
for signaling, this negotiation MUST use the mechanism defined in
. Implementations can also support sending RTP and RTCP on
separate transport-layer flows, but this is OPTIONAL to implement. If
an implementation does not support RTP and RTCP sent on separate
transport-layer flows, it MUST indicate that using the mechanism
defined in .
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 .Reduced Size RTCPRTCP packets are usually sent as compound RTCP packets, and requires that those compound packets start
with an SR or RR packet. When using
frequent RTCP feedback messages under the RTP/AVPF profile , these statistics are not needed in every
packet, and they unnecessarily increase the mean RTCP packet size. This can
limit the frequency at which RTCP packets can be sent within the RTCP
bandwidth share.To avoid this problem, 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 behavior.
Implementations MUST support sending and receiving noncompound RTCP
feedback packets . Use of noncompound
RTCP packets MUST be negotiated using the signaling channel. If SDP
is used for signaling, this negotiation MUST use the attributes
defined in . 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 noncompound RTCP in the signaling exchange.Symmetric RTP/RTCPTo ease traversal of NAT and firewall devices, implementations are
REQUIRED to implement and use symmetric
RTP. 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 bidirectional
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, because the NAT mappings or firewall state is not
unnecessarily bloated. The amount of per-flow QoS state kept in the
network is also reduced.Choice of RTP Synchronization Source (SSRC)Implementations are REQUIRED to support signaled RTP
synchronization source (SSRC) identifiers. If SDP is used, this MUST
be done using the "a=ssrc:" SDP attribute defined in Sections
and of and the "previous-ssrc" source attribute defined in ; other per-SSRC attributes defined in MAY be supported.While support for signaled 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
signaled ahead of time. Implementations MUST support random SSRC
assignment and MUST support SSRC collision detection and resolution,
according to . When using signaled SSRC
values, collision detection MUST be performed as described in
.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 is provided per . 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 signaled, 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 signaled, the RTP payload type numbers used in an RTP packet
stream are often sufficient to associate that packet stream with a
signaling context. For example, if RTP payload type numbers are assigned as
described in 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.Generation of the RTCP Canonical Name (CNAME)The RTCP Canonical Name (CNAME) provides a persistent
transport-level identifier for an RTP endpoint. While the
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 an RTCPeerConnection,
so that RTP endpoints can be uniquely identified and associated with
their RTP packet streams within a set of related RTP sessions.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 synchronization 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 unsynchronized
reference clocks, and hence different synchronization contexts, it
will need to use multiple RTCP CNAMEs, one for each synchronization
context.Taking the discussion in into
account, a WebRTC endpoint MUST NOT use more than one RTCP CNAME in
the RTP sessions belonging to a single RTCPeerConnection (that is, an
RTCPeerConnection forms a synchronization 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 that are part of a multiparty RTP
session.The RTP specification includes
guidelines for choosing a unique RTP CNAME, but these are not
sufficient in the presence of NAT devices. In addition, long-term
persistent identifiers can be problematic from a privacy viewpoint. Accordingly, a WebRTC
endpoint MUST generate a new, unique, short-term persistent RTCP CNAME
for each RTCPeerConnection, following ,
with a single exception; if explicitly requested at creation, an
RTCPeerConnection MAY use the same CNAME as an existing
RTCPeerConnection within their common same-origin context.A WebRTC endpoint MUST support reception of any CNAME that matches
the syntax limitations specified by the RTP
specification and cannot assume that any CNAME will be chosen
according to the form suggested above.Handling of Leap SecondsThe guidelines given in regarding
handling of leap seconds to limit their
impact on RTP media play-out and synchronization SHOULD be followed.WebRTC Use of RTP: ExtensionsThere 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.Conferencing Extensions and TopologiesRTP 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
.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
mesh-based topologies, such as a point-to-multipoint mesh
configured as a single RTP session ("Topo-Mesh" in the terminology of
).
However, a point-to-multipoint mesh constructed using several RTP
sessions, implemented in WebRTC using independent RTCPeerConnections, can be
expected to be used in WebRTC and needs to be supported.WebRTC endpoints implemented according to this memo are expected to
support all the topologies described in 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:
Video-switching Multipoint Control Units (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 ).
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 standardized.
The following topology can be used, however it has some issues
worth noting:
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 ).
The RTP extensions described in Sections to are designed to be used with
centralized 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; to provide a reasonable baseline quality, some of
these extensions are mandatory to be supported by WebRTC
endpoints.The RTCP conferencing extensions are defined in "Extended RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)"
and "Codec Control
Messages in the RTP Audio-Visual Profile with Feedback (AVPF)"; they
are fully usable by the secure variant of this
profile (RTP/SAVPF).Full Intra Request (FIR)The Full Intra Request message is defined in Sections and
of Codec Control Messages.
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 centralized mixer-based conferencing. Support
for sending FIR messages is OPTIONAL.Picture Loss Indication (PLI)The Picture Loss Indication message is defined in
the RTP/AVPF profile. 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 fulfill 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.Slice Loss Indication (SLI)The Slice Loss Indication message is defined in the RTP/AVPF profile. 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).Reference Picture Selection Indication (RPSI)Reference Picture Selection Indication (RPSI) messages are
defined in the RTP/AVPF
profile . 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 learned
that encoder-decoder synchronization 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 signaled. Receivers that
detect that encoder-decoder synchronization has been lost SHOULD
generate an RPSI feedback message if the codec being used supports
reference-picture selection. An 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.Temporal-Spatial Trade-Off Request (TSTR)The temporal-spatial trade-off request and notification are
defined in Sections and of . 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.Temporary Maximum Media Stream Bit Rate Request (TMMBR)The Temporary Maximum Media Stream Bit Rate Request (TMMBR) feedback message is defined in Sections and
of Codec Control Messages. This
request and its corresponding Temporary Maximum Media Stream Bit
Rate Notification (TMMBN) 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 messages is
OPTIONAL.Header ExtensionsThe RTP specification 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 signaled following the general
mechanism for RTP header extensions defined in , since this gives well-defined semantics to
RTP header extensions.As noted in , the requirement from
the RTP specification that header extensions are "designed so that the
header extension may be ignored"
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 signaled
(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.Rapid SynchronizationMany RTP sessions require synchronization between audio, video,
and other content. This synchronization is performed by receivers,
using information contained in RTCP SR packets, as described in the
RTP specification. This basic
mechanism can be slow, however, so it is RECOMMENDED that the rapid
RTP synchronization extensions described in be implemented in addition to RTCP SR-based
synchronization.This header extension uses the
generic header extension framework described in and so needs to be negotiated
before it can be used.Client-to-Mixer Audio LevelThe client-to-mixer audio level
extension 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.The client-to-mixer audio level header
extension MUST be implemented. It is REQUIRED that
implementations be capable of encrypting the header extension
according to , 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 signaling.This header extension uses the
generic header extension framework described in and so needs to be negotiated
before it can be used.Mixer-to-Client Audio LevelThe mixer-to-client audio level header
extension provides an endpoint with the audio level of the
different sources mixed into a common source stream by an 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 optimization of non-critical functions and is hence OPTIONAL to implement. If this
header extension is implemented, it is REQUIRED that implementations
be capable of encrypting the header extension according to , since the information contained in these
header extensions can be considered sensitive. It is further
RECOMMENDED that this encryption be used, unless the encryption has
been explicitly disabled through API or signaling.This header extension uses the
generic header extension framework described in and so needs to be negotiated
before it can be used.Media Stream IdentificationWebRTC 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 .This header extension uses the
generic header extension framework described in and so needs to be negotiated
before it can be used.Coordination of Video OrientationWebRTC endpoints that send or receive video MUST implement the
coordination of video orientation (CVO) RTP header extension as
described in .This header extension uses the
generic header extension framework described in and so needs to be negotiated
before it can be used.WebRTC Use of RTP: Improving Transport RobustnessThere 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 bitrate MUST be rate
controlled to avoid causing network congestion (see ). 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 subsections can be used to improve tolerance to packet
loss.Negative Acknowledgements and RTP RetransmissionAs a consequence of supporting the RTP/SAVPF profile,
implementations can send negative acknowledgements (NACKs) for RTP
data packets . 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 optimize the user experience by adapting the
media encoding to compensate for known lost packets.RTP packet stream senders are REQUIRED to understand the generic
NACK message defined in , but they MAY choose to ignore some or all of this
feedback (following ).
Receivers MAY send NACKs for missing RTP packets. Guidelines on when
to send NACKs are provided in . 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.The RTP retransmission payload format
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 it 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
). The use of
retransmissions can also increase the forward RTP bandwidth and can
potentially cause 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 need
to be considered before RTP retransmission is used.Receivers are REQUIRED to implement support for RTP retransmission
packets 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 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.Forward Error Correction (FEC)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, and others operate across RTP packets and can be used with
any payload format. Note 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.WebRTC endpoints MUST follow the recommendations for FEC use given
in . WebRTC endpoints MAY
support other types of FEC, but these MUST be negotiated before they
are used.WebRTC Use of RTP: Rate Control and Media AdaptationWebRTC 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 bitrates, 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, bitrate, 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.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).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 .
If a standardized congestion control algorithm that satisfies these
requirements is developed in the future, this memo will need to be
updated to mandate its use.Boundary Conditions and Circuit BreakersWebRTC endpoints MUST implement the RTP circuit breaker algorithm
that is described in . The RTP
circuit breaker is designed to enable applications to recognize 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.The session-establishment signaling will also necessarily
establish boundaries to which the media bitrate will conform. The
choice of media codecs provides upper and lower bounds on the
supported bitrates that the application can utilize to provide useful
quality, and the packetization choices that exist. In addition, the
signaling channel can establish maximum media bitrate boundaries
using, for example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF
TMMBR messages (see of this memo). Signaled 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.Congestion Control Interoperability and Legacy SystemsAll endpoints that wish to interwork with WebRTC MUST implement
RTCP and provide congestion feedback via the defined RTCP reporting
mechanisms.When interworking with legacy implementations that support RTCP
using the RTP/AVP profile, 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 RTP
circuit breaker constraints to limit the
congestion they can cause.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 purposes such as TMMBR. Implementations that have to interwork
with such endpoints MUST ensure that they stay within
the RTP circuit
breaker constraints to limit the
congestion they can cause, but they
might find that they can achieve better congestion response depending
on the amount of feedback that is available.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 a
situation where one direction is dual controlled when the other
direction is not controlled. This memo cannot mandate behavior 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 standardize 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.WebRTC Use of RTP: Performance MonitoringAs described in , implementations
are REQUIRED to generate RTCP Sender Report (SR) and Receiver 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.A large number of additional performance metrics are supported by the
RTCP Extended Reports (XR) framework; see and . 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 signaled; implementations MUST ignore RTCP XR
packets that are unexpected or not understood.WebRTC Use of RTP: Future ExtensionsIt 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 "Guidelines for Writers of RTP
Payload Format Specifications", "How to Write an RTP Payload
Format", and "Guidelines for Extending the
RTP Control Protocol (RTCP)". They also SHOULD take into account any future
guidelines for extending RTP and related protocols that have been
developed.Authors of future extensions are urged to consider the wide range of
environments in which RTP is used when recommending extensions, since
extensions that are applicable in some scenarios can be problematic in
others. Where possible, the WebRTC framework 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.Signaling ConsiderationsRTP is built with the assumption that an external signaling 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:
RTP profile:
The name of the RTP profile to be used in the
session. The RTP/AVP and RTP/AVPF profiles can interoperate on a basic
level, as can their secure variants, RTP/SAVP and RTP/SAVPF. The secure variants of the
profiles do not directly interoperate with the nonsecure 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
signaled. 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.
Transport information:
Source and destination IP
address(es) and ports for RTP and RTCP MUST be signaled for each RTP
session. In WebRTC, these transport addresses will be provided by
Interactive Connectivity Establishment
(ICE) that signals candidates and
arrives at nominated candidate address pairs. If RTP and RTCP multiplexing is to be used
such that a single port -- i.e., transport-layer flow -- is used for RTP
and RTCP flows, this MUST be signaled (see ).
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 signaled.
Each media type MAY also have a number of media type parameters that
MUST also be signaled to configure the codec and RTP payload format
(the "a=fmtp:" line from SDP). of
this memo discusses requirements for uniqueness of payload
types.
RTP extensions:
The use of any additional RTP header
extensions and RTCP packet types, including any necessary
parameters, MUST be signaled. This signaling ensures
that a WebRTC endpoint's behavior, especially when sending, is predictable and consistent. For robustness and
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 ).
RTCP bandwidth:
Support for exchanging RTCP bandwidth
values with the endpoints will be necessary. This SHALL be done as
described in "Session Description Protocol
(SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP)
Bandwidth" 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.
These parameters are often expressed in SDP messages conveyed within
an offer/answer exchange. RTP does not depend on SDP or 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 signaling model and API how these parameters need
to be configured, but they will need to either be set in the API or
explicitly signaled between the peers.WebRTC API ConsiderationsThe WebRTC API and the
Media Capture and
Streams API define and use 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, such as a
microphone or a camera, but conceptual sources, like an audio mix or
a video composition, are also possible. The MediaStreamTracks within a
MediaStream might need to be synchronized during playback.A MediaStreamTrack's realization in RTP, in the context of an
RTCPeerConnection, consists of a source packet stream, identified by an
SSRC, sent within an RTP session that is 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 Forward Error Correction or RTP retransmission to the source packet
stream.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 an 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 optimizations would need to
take into account that the constraints for one of the MediaStreamTracks
can change at any moment, meaning that the encoding configurations might
no longer be identical, and two different encoder instances would then be
needed.The same MediaStreamTrack can also be included in multiple
MediaStreams; thus, multiple sets of MediaStreams can implicitly need to
use the same synchronization base. To ensure that this works in all
cases and does not force an endpoint to disrupt the media by changing
synchronization 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 . Different CNAMEs normally need to be used for different
RTCPeerConnection instances, as specified in . Having two communication sessions with the
same CNAME could enable tracking of a user or device across different
services (see for details). A web
application can request that the CNAMEs used in different
RTCPeerConnections (within a same-origin context) be the same; this
allows for synchronization of the endpoint's RTP packet streams across
the different RTCPeerConnections.The above will currently force a WebRTC endpoint that receives
a MediaStreamTrack on one RTCPeerConnection and adds it as outgoing one
on any RTCPeerConnection to perform resynchronization 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 the timebase for the
synchronization. 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 synchronization. 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, followed by encoding of the media again
into the outgoing packet stream. Optimizations of this method are
clearly possible and implementation specific.A WebRTC endpoint MUST support receiving multiple MediaStreamTracks,
where each of the different MediaStreamTracks (and its sets of
associated packet streams) uses different CNAMEs. However,
MediaStreamTracks that are received with different CNAMEs have no
defined synchronization."JavaScript Session Establishment
Protocol (JSEP)" specifies that the binding between the WebRTC
MediaStreams, MediaStreamTracks, and the SSRC is done as specified in "WebRTC MediaStream Identification in the Session
Description Protocol". Section 4.1 of the MediaStream Identification (MSID) document also defines
how to map source packet streams with unknown 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.Finally, this specification puts a requirement on the WebRTC API to
realize a method for determining the CSRC
list as well as the mixer-to-client audio levels (when
supported); the basic requirements for this is further discussed in
.RTP Implementation ConsiderationsThe 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 behavior of middleboxes, that is not the focus of this memo.Configuration and Use of RTP SessionsA WebRTC endpoint will be a simultaneous participant in one or more
RTP sessions. Each RTP session can convey multiple media sources and
include media data from multiple endpoints. In the following, some
ways in which WebRTC endpoints can configure and use RTP sessions are
outlined.Use of Multiple Media Sources within an RTP SessionRTP 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:
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 , using RTP
and RTCP multiplexing () to
further reduce the number of UDP ports needed. This RTP session
then uses only one bidirectional transport-layer flow but will
contain multiple RTP packet streams, each containing a different
type of media. A common example might be an endpoint with a
camera and microphone that sends two RTP packet streams, one
video and one audio, into a single RTP session.
Multiple capture devices:A WebRTC endpoint might
have multiple cameras, microphones, or other media capture
devices, and so it 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.
Associated repair data:An endpoint might send an
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.
Layered or multiple-description coding:Within a single
RTP session, an endpoint can use a layered media codec -- for
example, H.264 Scalable Video Coding (SVC) --
or a multiple-description codec that generates multiple RTP
packet streams, each with a distinct RTP SSRC.
RTP mixers, translators, and other middleboxes:An
RTP session, in the WebRTC context, is a point-to-point
association between an endpoint and some other peer device,
where those devices share a common SSRC space. The peer device
might be another WebRTC endpoint, or it might be an RTP mixer,
translator, or some other form of media-processing middlebox. In
the latter cases, the middlebox might send mixed or relayed RTP
streams from several participants, which the WebRTC endpoint will
need to render. Thus, even though a WebRTC endpoint might only
be a member of a single RTP session, the peer device might be
extending that RTP session to incorporate other endpoints.
WebRTC is a group communication environment, and endpoints need
to be capable of receiving, decoding, and playing out multiple
RTP packet streams at once, even in a single RTP session.
Use of Multiple RTP SessionsIn 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:
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 .
To provide enhanced quality of service:Some
network-based quality-of-service mechanisms operate on the
granularity of transport-layer flows. If use of
these mechanisms to provide differentiated quality of service
for some RTP packet streams is desired, 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 .
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
centralized multiparty conferencing systems display the active
speaker in high resolution but show low-resolution "thumbnails"
of other participants. Such systems might configure the
endpoints to send simulcast high- and low-resolution versions of
their video using separate RTP sessions to simplify the
operation of the RTP middlebox. In the WebRTC context, this is
currently possible by establishing multiple WebRTC
MediaStreamTracks that have the same media source in one (or
more) RTCPeerConnection. Each MediaStreamTrack is then
configured to deliver a particular media quality and thus media
bitrate, and it will produce an independently encoded version with
the codec parameters agreed specifically in the context of that
RTCPeerConnection. The RTP middlebox can distinguish packets
corresponding to the low- and high-resolution streams by
inspecting their SSRC, RTP payload type, or some other
information contained in RTP payload, RTP header extension, or
RTCP packets. However, it can be easier to distinguish the RTP packet
streams if they arrive on separate RTP sessions on separate
transport-layer flows.
To directly connect with multiple peers:A
multiparty conference does not need to use an RTP middlebox.
Rather, a multi-unicast mesh can be created, comprising several
distinct RTP sessions, with each participant sending RTP traffic
over a separate RTP session (that is, using an independent
RTCPeerConnection object) to every other participant, as shown
in . This topology has the
benefit of not requiring an RTP middlebox node that is trusted
to access and manipulate the media data. The downside is that it
increases the used bandwidth at each sender by requiring one
copy of the RTP packet streams for each participant that is
part of the same session beyond the sender itself.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 RECOMMENDED that this be 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 shown 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 bitrates and RTP session
configurations between the different peers, thus not forcing B
to endure the same quality reductions as C will if there are limitations
in the transport from A to C. It is believed that
these advantages outweigh the limitations in debugging
power.
To indirectly connect with multiple peers:A
common scenario in multiparty conferencing is to create
indirect connections to multiple peers, using an RTP mixer,
translator, or some other type of RTP middlebox. outlines a simple topology that
might be used in a four-person centralized conference. The
middlebox acts to optimize 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.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 multiparty session. Other
types of middleboxes 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 having the
encoder produce a so-called Intra frame. Another common aspect
is limiting the bitrate of a stream to better match the mixed
output. Other aspects are controlling the most suitable
frame rate, picture resolution, and the trade-off between frame rate
and spatial quality. The middlebox has the responsibility to
correctly perform congestion control, identify sources, and
manage synchronization while providing the application with
suitable media optimizations. 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 them 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.Mixers are expected to not
forward RTCP reports regarding RTP packet streams across
themselves. This is due to the difference between the RTP packet
streams provided to the different endpoints. The original media
source lacks information about a mixer's manipulations prior to being
sent to the different receivers. This scenario also results
in an endpoint's feedback or requests going to the mixer. When
the mixer can't act on this by itself, it is forced to go to the
original media source to fulfill the receiver's 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.Providing source authentication in multiparty 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 claiming to
come from another endpoint in the session. Any node performing
relay can perform noncryptographic 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,
Timed Efficient Stream Loss-Tolerant
Authentication (TESLA) for SRTP -- that are not part
of the base WebRTC standards.
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 a media source that is re-encoded and transmitted as a new
stream.At the RTP layer, media forwarding acts as a back-to-back RTP
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.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 from the RTP session on which
the media was received. This will require media transcoding for
congestion control purposes to produce a suitable bitrate 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 optimize its media transcoding
operation. The cost is greatly increased computational
complexity on the forwarding node. Receivers of the forwarded
stream will see the forwarding device as the sender of the
stream and will not be able to tell from the RTP layer that
they are receiving a forwarded stream rather than an entirely
new RTP packet stream generated by the forwarding device.
Differentiated Treatment of RTP StreamsThere 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
will be sent and their allocation of bitrate out of the
current available aggregate, as determined by the congestion
control.It is expected that the WebRTC API will allow the
application to indicate relative priorities for different
MediaStreamTracks. These priorities can then be used to influence
the local RTP processing, especially when it comes to determining
how to divide the available bandwidth between
the RTP packet streams for the sake of congestion control. 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, with regard to
how robust that codec is against packet loss. However, a default policy
might be to use the same priority for a redundant RTP packet stream
as for the source RTP packet stream.Secondly, the network can prioritize transport-layer flows and
subflows, 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 for prioritizing packets.
Three common methods for classifying IP packets are:
DiffServ:
The endpoint marks a packet with a
DiffServ code point to indicate to the network that the packet
belongs to a particular class.
Flow based:
Packets that need to be given a
particular treatment are identified using a combination of IP
and port address.
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.
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.DiffServ assumes that either the endpoint or a classifier can
mark the packets with an appropriate Differentiated Services Code
Point (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 DSCPs to use and know 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 "Differentiated Services Code Point (DSCP) Packet
Markings for WebRTC QoS".Despite the SRTP media encryption, deep packet inspectors will
still be fairly capable of
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.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 prioritize any payloads carrying only B frames less,
as these are less damaging to lose. However, depending on the QoS
mechanism and what markings are applied, this can result in not
only different packet-drop probabilities but also packet reordering;
see and for further discussion. As a
default policy, all RTP packets related to an 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.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 (SRs) 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 an 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.Media Source, RTP Streams, and Participant IdentificationMedia Source IdentificationEach RTP packet stream is identified by a unique synchronization
source (SSRC) identifier. The SSRC identifier is carried in each of
the RTP packets comprising an RTP packet stream, and is also used to
identify that stream in the corresponding RTCP reports. The SSRC is
chosen as discussed in . 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.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 ,
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 need 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 namespace to WebRTC
applications.If the mixer-to-client audio level extension is being used in the session (see ), 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.SSRC Collision DetectionThe RTP standard requires RTP implementations to have support for
detecting and handling SSRC collisions -- i.e., be able to resolve the conflict
when two different endpoints use the same SSRC value (see ). This requirement also
applies to WebRTC endpoints. There are several scenarios where SSRC
collisions can occur:
In a point-to-point session where each SSRC is associated
with either of the two endpoints and the main media-carrying SSRC
identifier will be announced in the signaling
channel, a collision is less likely to occur due to the
information about used SSRCs. If SDP is used, this information
is provided by source-specific SDP
attributes. Still, collisions can occur if both endpoints
start using a new SSRC identifier prior to having signaled it
to the peer and received acknowledgement on the signaling
message. "Source-Specific Media Attributes in the
Session Description Protocol (SDP)"
contains a mechanism to signal how the
endpoint resolved the SSRC collision.
SSRC values that have not been signaled 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
from the SSRC of the source RTP packet stream . In those cases, an endpoint can create
a new SSRC that strictly doesn't need to be announced over the
signaling channel to function correctly on both RTP and
RTCPeerConnection level.
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.
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 be able to recognize and handle the
case when it occurs. This case becomes even more problematic
when media mixers and such are involved, where the stream
received is a different stream but still contains this client's
input.
These SSRC/CSRC collisions can only be handled on the RTP level
when the same RTP session is extended across multiple
RTCPeerConnections by an RTP middlebox. To resolve the more generic
case where multiple RTCPeerConnections are interconnected,
identification of the media source or sources that are part of a MediaStreamTrack
being propagated across multiple interconnected RTCPeerConnection
needs to be preserved across these interconnections.Media Synchronization ContextWhen 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, synchronization is provided by having a
set of RTP packet streams be indicated as coming from the same
synchronization context and logical endpoint by using the same RTCP
CNAME identifier.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; whether or not the information is used is up to the receiver.Security ConsiderationsThe overall security architecture for WebRTC is described in , and security
considerations for the WebRTC framework are described in . These considerations also
apply to this memo.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 believed that there are no new security considerations resulting from
the combination of these various protocol extensions."Extended Secure RTP
Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)"
provides handling of fundamental issues by offering confidentiality,
integrity, and partial source authentication. A media-security solution
that is mandatory to implement and use is created by combining this secured RTP
profile and DTLS-SRTP keying, as defined by
.RTCP packets convey a Canonical Name (CNAME) identifier that is used
to associate RTP packet streams that need to be synchronized 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. of this memo mandates generation of
short-term persistent RTCP CNAMES, as specified in RFC 7022, resulting in
untraceable CNAME values that alleviate this risk.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 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) . The risks are as
follows:
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;
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 RTCP traffic not being
congestion controlled; and
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 that are based on the reporting interval. This is a
particular concern if endpoints use a small but nonzero value for
the RTP/AVPF minimal receiver report interval (trr-int) , as discussed in
.
Premature participant timeout can be avoided by using the fixed
(nonreduced) minimum interval when calculating the participant timeout
(see of this memo and
). 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 (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.The guidelines in apply when using
variable bitrate (VBR) audio codecs such as Opus (see for discussion of mandated audio codecs).
The guidelines in also apply, but are of
lesser importance, when using the client-to-mixer audio level header
extensions () or the
mixer-to-client audio level header extensions (). 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 signaling. If further evidence
is produced to show that information leakage is significant from
audio-level indications, then use of encryption needs to be mandated at
that time.In multiparty communication scenarios using RTP middleboxes, a lot
of trust is placed on these middleboxes to preserve the session's
security. The middlebox needs to maintain confidentiality and integrity
and perform source authentication. As discussed in , the middlebox can perform checks
that prevent any endpoint participating in a conference from impersonating
another. Some additional security considerations regarding multiparty
topologies can be found in .IANA ConsiderationsThis document has no IANA actions.ReferencesNormative ReferencesSending Multiple Types of Media in a Single RTP SessionSending Multiple RTP Streams in a Single RTP Session:
Grouping RTP Control Protocol (RTCP) Reception Statistics and Other FeedbackIndicating Exclusive Support of RTP and RTP Control Protocol (RTCP)
Multiplexing Using the Session Description Protocol (SDP)WebRTC Forward Error Correction RequirementsOverview: Real-Time Protocols for Browser-Based ApplicationsSecurity Considerations for WebRTCWebRTC Security ArchitectureNegotiating Media Multiplexing Using the Session Description Protocol (SDP)WebRTC 1.0: Real-time Communication Between BrowsersW3C Proposed RecommendationMedia Capture and StreamsW3C Candidate RecommendationInformative ReferencesWebRTC MediaStream Identification in the Session Description ProtocolGuidelines for Using the Multiplexing Features of RTP to Support
Multiple Media StreamsCongestion Control Requirements for Interactive Real-Time MediaDifferentiated Services Code Point (DSCP) Packet Markings for
WebRTC QoSJavaScript Session Establishment Protocol (JSEP)AcknowledgementsThe authors would like to thank ,
, ,
, ,
, ,
, , , , , , , , , and the other members of the
IETF RTCWEB working group for their valuable feedback.