rfc8834
Internet Engineering Task Force (IETF) C. Perkins
Request for Comments: 8834 University of Glasgow
Category: Standards Track M. Westerlund
ISSN: 2070-1721 Ericsson
J. Ott
Technical University Munich
January 2021
Media Transport and Use of RTP in WebRTC
Abstract
The 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.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8834.
Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction
2. Rationale
3. Terminology
4. WebRTC Use of RTP: Core Protocols
4.1. RTP and RTCP
4.2. Choice of the RTP Profile
4.3. Choice of RTP Payload Formats
4.4. Use of RTP Sessions
4.5. RTP and RTCP Multiplexing
4.6. Reduced Size RTCP
4.7. Symmetric RTP/RTCP
4.8. Choice of RTP Synchronization Source (SSRC)
4.9. Generation of the RTCP Canonical Name (CNAME)
4.10. Handling of Leap Seconds
5. WebRTC Use of RTP: Extensions
5.1. Conferencing Extensions and Topologies
5.1.1. Full Intra Request (FIR)
5.1.2. Picture Loss Indication (PLI)
5.1.3. Slice Loss Indication (SLI)
5.1.4. Reference Picture Selection Indication (RPSI)
5.1.5. Temporal-Spatial Trade-Off Request (TSTR)
5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
5.2. Header Extensions
5.2.1. Rapid Synchronization
5.2.2. Client-to-Mixer Audio Level
5.2.3. Mixer-to-Client Audio Level
5.2.4. Media Stream Identification
5.2.5. Coordination of Video Orientation
6. WebRTC Use of RTP: Improving Transport Robustness
6.1. Negative Acknowledgements and RTP Retransmission
6.2. Forward Error Correction (FEC)
7. WebRTC Use of RTP: Rate Control and Media Adaptation
7.1. Boundary Conditions and Circuit Breakers
7.2. Congestion Control Interoperability and Legacy Systems
8. WebRTC Use of RTP: Performance Monitoring
9. WebRTC Use of RTP: Future Extensions
10. Signaling Considerations
11. WebRTC API Considerations
12. RTP Implementation Considerations
12.1. Configuration and Use of RTP Sessions
12.1.1. Use of Multiple Media Sources within an RTP Session
12.1.2. Use of Multiple RTP Sessions
12.1.3. Differentiated Treatment of RTP Streams
12.2. Media Source, RTP Streams, and Participant Identification
12.2.1. Media Source Identification
12.2.2. SSRC Collision Detection
12.2.3. Media Synchronization Context
13. Security Considerations
14. IANA Considerations
15. References
15.1. Normative References
15.2. Informative References
Acknowledgements
Authors' Addresses
1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] 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 [RFC8825].
The structure of this memo is as follows. Section 2 outlines our
rationale for preparing this memo and choosing these RTP features.
Section 3 defines terminology. Requirements for core RTP protocols
are described in Section 4, and suggested RTP extensions are
described in Section 5. Section 6 outlines mechanisms that can
increase robustness to network problems, while Section 7 describes
congestion control and rate adaptation mechanisms. The discussion of
mandated RTP mechanisms concludes in Section 8 with a review of
performance monitoring and network management tools. Section 9 gives
some guidelines for future incorporation of other RTP and RTP Control
Protocol (RTCP) extensions into this framework. Section 10 describes
requirements placed on the signaling channel. Section 11 discusses
the relationship between features of the RTP framework and the WebRTC
application programming interface (API), and Section 12 discusses RTP
implementation considerations. The memo concludes with security
considerations (Section 13) and IANA considerations (Section 14).
2. Rationale
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
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 [RFC7478].
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.
3. 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 [RFC2119] [RFC8174] 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 [W3C.WD-mediacapture-streams]. A MediaStream
consists of zero or more MediaStreamTracks.
MediaStreamTrack: Part of the MediaStream concept defined by the W3C
in the WebRTC API [W3C.WD-mediacapture-streams]. 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 [RFC7656] and [RFC8825].
Other terms are used according to their definitions from the RTP
specification [RFC3550]. In particular, note the following
frequently used terms: RTP stream, RTP session, and endpoint.
4. WebRTC Use of RTP: Core Protocols
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.
4.1. RTP and RTCP
The Real-time Transport Protocol (RTP) [RFC3550] 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
[RFC3550] and [RFC8108]. The RTCP optimizations for multi-SSRC
sessions defined in [RFC8861] 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 Section 4.8).
* 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
Section 5.2.1 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 Section 4.9.
* 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
(Section 4.2) and the other RTCP extensions (Section 5). 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 [RFC8843].
* 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 (Section 6.3.6 of
[RFC3550]) and reverse reconsideration (Section 6.3.4 of
[RFC3550]).
* 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
[RFC4566] [RFC3556].
* Support for the reduced minimum RTCP reporting interval described
in Section 6.2 of [RFC3550]. When using the reduced minimum RTCP
reporting interval, the fixed (nonreduced) minimum interval MUST
be used when calculating the participant timeout interval (see
Sections 6.2 and 6.3.5 of [RFC3550]). The delay before sending
the initial compound RTCP packet can be set to zero (see
Section 6.2 of [RFC3550] as updated by [RFC8108]).
* 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 Section 12.
4.2. Choice of the RTP Profile
The 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) [RFC5124], as
extended by [RFC7007], MUST be implemented. The RTP/SAVPF profile is
the combination of the basic RTP/AVP profile [RFC3551], the RTP
profile for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure
RTP profile (RTP/SAVP) [RFC3711].
The RTCP-based feedback extensions [RFC4585] 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 Section 5.1.
| 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 7.1.3 of [RFC8108].
The secure RTP (SRTP) profile extensions [RFC3711] 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 [RFC8827].
4.3. Choice of RTP Payload Formats
Mandatory-to-implement audio codecs and RTP payload formats for
WebRTC endpoints are defined in [RFC7874]. Mandatory-to-implement
video codecs and RTP payload formats for WebRTC endpoints are defined
in [RFC7742]. 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 Section 4.8, 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 [RFC8843] 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 Section 4.4. 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 Section 4.1 of [RFC7160]. RTP receivers MUST
follow the recommendations in Section 4.3 of [RFC7160] 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.
4.4. Use of RTP Sessions
An association amongst a set of endpoints communicating using RTP is
known as an RTP session [RFC3550]. 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 Section 12.1.3. 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 [RFC8860] (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 [RFC8843] 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 [RFC8872].
4.5. RTP and RTCP Multiplexing
Historically, 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 [RFC5761].
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 [RFC5761].
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 [RFC8858].
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 [RFC6263].
4.6. Reduced Size RTCP
RTCP packets are usually sent as compound RTCP packets, and [RFC3550]
requires that those compound packets start with an SR or RR packet.
When using frequent RTCP feedback messages under the RTP/AVPF profile
[RFC4585], 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, [RFC5506] specifies how to reduce the mean
RTCP message size and allow for more frequent feedback. Frequent
feedback, in turn, is essential to make real-time applications
quickly aware of changing network conditions and to allow them to
adapt their transmission and encoding behavior. Implementations MUST
support sending and receiving noncompound RTCP feedback packets
[RFC5506]. 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 [RFC5506]. 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.
4.7. Symmetric RTP/RTCP
To ease traversal of NAT and firewall devices, implementations are
REQUIRED to implement and use symmetric RTP [RFC4961]. 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.
4.8. 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 4.1 and 5 of
[RFC5576] and the "previous-ssrc" source attribute defined in
Section 6.2 of [RFC5576]; other per-SSRC attributes defined in
[RFC5576] 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 [RFC3550]. When using signaled SSRC values, collision
detection MUST be performed as described in Section 5 of [RFC5576].
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 Section 11. 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 Section 4.3 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.
4.9. 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 [W3C.WebRTC], 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 Section 11 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 [RFC3550] 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 (Section 13). Accordingly, a
WebRTC endpoint MUST generate a new, unique, short-term persistent
RTCP CNAME for each RTCPeerConnection, following [RFC7022], 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 [RFC3550]
and cannot assume that any CNAME will be chosen according to the form
suggested above.
4.10. Handling of Leap Seconds
The guidelines given in [RFC7164] regarding handling of leap seconds
to limit their impact on RTP media play-out and synchronization
SHOULD be followed.
5. WebRTC Use of RTP: Extensions
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.
5.1. Conferencing Extensions and Topologies
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
[RFC7667].
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
[RFC7667]). However, a point-to-multipoint mesh constructed using
several RTP sessions, implemented in WebRTC using independent
RTCPeerConnections [W3C.WebRTC], 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 [RFC7667] 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 Section 3.8 of [RFC7667]).
* 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 Section 3.9 of [RFC7667]).
The RTP extensions described in Sections 5.1.1 to 5.1.6 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)" [RFC4585] and "Codec Control Messages in the RTP Audio-Visual
Profile with Feedback (AVPF)" [RFC5104]; they are fully usable by the
secure variant of this profile (RTP/SAVPF) [RFC5124].
5.1.1. Full Intra Request (FIR)
The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1
of Codec Control Messages [RFC5104]. 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.
5.1.2. Picture Loss Indication (PLI)
The Picture Loss Indication message is defined in Section 6.3.1 of
the RTP/AVPF profile [RFC4585]. 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.
5.1.3. Slice Loss Indication (SLI)
The Slice Loss Indication message is defined in Section 6.3.2 of the
RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the
encoder that it has detected the loss or corruption of one or more
consecutive 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).
5.1.4. Reference Picture Selection Indication (RPSI)
Reference Picture Selection Indication (RPSI) messages are defined in
Section 6.3.3 of the RTP/AVPF profile [RFC4585]. 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.
5.1.5. Temporal-Spatial Trade-Off Request (TSTR)
The temporal-spatial trade-off request and notification are defined
in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used
to ask the video encoder to change the trade-off it makes between
temporal and spatial resolution -- for example, to prefer high
spatial image quality but low frame rate. Support for TSTR requests
and notifications is OPTIONAL.
5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
The Temporary Maximum Media Stream Bit Rate Request (TMMBR) feedback
message is defined in Sections 3.5.4 and 4.2.1 of Codec Control
Messages [RFC5104]. This request and its corresponding Temporary
Maximum Media Stream Bit Rate Notification (TMMBN) message [RFC5104]
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.
5.2. Header Extensions
The RTP specification [RFC3550] 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 [RFC8285], since this gives well-defined
semantics to RTP header extensions.
As noted in [RFC8285], the requirement from the RTP specification
that header extensions are "designed so that the header extension may
be ignored" [RFC3550] stands. To be specific, header extensions MUST
only be used for data that can safely be ignored by the recipient
without affecting interoperability and MUST NOT be used when the
presence of the extension has changed the form or nature of the rest
of the packet in a way that is not compatible with the way the stream
is 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.
5.2.1. Rapid Synchronization
Many 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 [RFC3550]. This basic mechanism can be slow, however,
so it is RECOMMENDED that the rapid RTP synchronization extensions
described in [RFC6051] be implemented in addition to RTCP SR-based
synchronization.
This header extension uses the generic header extension framework
described in [RFC8285] and so needs to be negotiated before it can be
used.
5.2.2. Client-to-Mixer Audio Level
The client-to-mixer audio level extension [RFC6464] 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 [RFC6464] MUST be
implemented. It is REQUIRED that implementations be capable of
encrypting the header extension according to [RFC6904], 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 [RFC8285] and so needs to be negotiated before it can be
used.
5.2.3. Mixer-to-Client Audio Level
The mixer-to-client audio level header extension [RFC6465] 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 [RFC6904], 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 [RFC8285] and so needs to be negotiated before it can be
used.
5.2.4. Media Stream Identification
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 [RFC8843].
This header extension uses the generic header extension framework
described in [RFC8285] and so needs to be negotiated before it can be
used.
5.2.5. Coordination of Video Orientation
WebRTC endpoints that send or receive video MUST implement the
coordination of video orientation (CVO) RTP header extension as
described in Section 4 of [RFC7742].
This header extension uses the generic header extension framework
described in [RFC8285] and so needs to be negotiated before it can be
used.
6. WebRTC Use of RTP: Improving Transport Robustness
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 bitrate MUST
be rate controlled to avoid causing network congestion (see
Section 7). 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.
6.1. Negative Acknowledgements and RTP Retransmission
As a consequence of supporting the RTP/SAVPF profile, implementations
can send negative acknowledgements (NACKs) for RTP data packets
[RFC4585]. This feedback can be used to inform a sender of the loss
of particular RTP packets, subject to the capacity limitations of the
RTCP feedback channel. A sender can use this information to 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 Section 6.2.1 of [RFC4585], but they MAY choose to
ignore some or all of this feedback (following Section 4.2 of
[RFC4585]). Receivers MAY send NACKs for missing RTP packets.
Guidelines on when to send NACKs are provided in [RFC4585]. 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 [RFC4588] 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
Section 6.4.1 of [RFC3550]). 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 [RFC4588] 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.
6.2. 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
[RFC8854]. WebRTC endpoints MAY support other types of FEC, but
these MUST be negotiated before they are used.
7. WebRTC Use of RTP: Rate Control and Media Adaptation
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 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 [RFC8836]. 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.
7.1. Boundary Conditions and Circuit Breakers
WebRTC endpoints MUST implement the RTP circuit breaker algorithm
that is described in [RFC8083]. 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 Section 5.1.6 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.
7.2. Congestion Control Interoperability and Legacy Systems
All 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 [RFC3551], 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 [RFC8083] 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
[RFC5104]. Implementations that have to interwork with such
endpoints MUST ensure that they stay within the RTP circuit breaker
[RFC8083] 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.
8. WebRTC Use of RTP: Performance Monitoring
As described in Section 4.1, 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 [RFC3611] and [RFC6792].
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.
9. WebRTC Use of RTP: Future Extensions
It is possible that the core set of RTP protocols and RTP extensions
specified in this memo will prove insufficient for the future needs
of WebRTC. In this case, future updates to this memo have to be made
following "Guidelines for Writers of RTP Payload Format
Specifications" [RFC2736], "How to Write an RTP Payload Format"
[RFC8088], and "Guidelines for Extending the RTP Control Protocol
(RTCP)" [RFC5968]. 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.
10. Signaling Considerations
RTP 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 [RFC3551] and RTP/AVPF [RFC4585] profiles can
interoperate on a basic level, as can their secure variants, RTP/
SAVP [RFC3711] and RTP/SAVPF [RFC5124]. The secure variants of
the profiles do not directly interoperate with the 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) [RFC8445] that signals candidates
and arrives at nominated candidate address pairs. If RTP and RTCP
multiplexing [RFC5761] 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 Section 4.5).
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). Section 4.3 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 Section 4.1).
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" [RFC3556] 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.
11. WebRTC API Considerations
The WebRTC API [W3C.WebRTC] and the Media Capture and Streams API
[W3C.WD-mediacapture-streams] 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 (Section 6.2) or
RTP retransmission (Section 6.1) 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 Section 4.9.
| The requirement to use 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.
Different CNAMEs normally need to be used for different
RTCPeerConnection instances, as specified in Section 4.9. 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
[RFC8826] 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.
| Note: This doesn't result in a tracking issue, since the
| creation of matching CNAMEs depends on existing tracking within
| a single origin.
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.
| 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 multistream
| middleboxes or endpoints that are not WebRTC.
"JavaScript Session Establishment Protocol (JSEP)" [RFC8829]
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"
[RFC8830]. Section 4.1 of the MediaStream Identification (MSID)
document [RFC8830] 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 (Section 4.1) as well
as the mixer-to-client audio levels (Section 5.2.3) (when supported);
the basic requirements for this is further discussed in
Section 12.2.1.
12. RTP Implementation Considerations
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 behavior of middleboxes, that is not the focus of this memo.
12.1. Configuration and Use of RTP Sessions
A 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.
12.1.1. Use of Multiple Media Sources within an RTP Session
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:
* 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 Section 4.4, using RTP and RTCP
multiplexing (Section 4.5) 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.
12.1.2. Use of Multiple RTP Sessions
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:
* 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 Section 4.4.
* 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 Section 12.1.3.
* 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
Figure 1. 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.
+---+ +---+
| A |<--->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 1: Multi-unicast Using Several RTP Sessions
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. Figure 2 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.
+---+ +-------------+ +---+
| A |<---->| |<---->| B |
+---+ | RTP mixer, | +---+
| translator, |
| or other |
+---+ | middlebox | +---+
| C |<---->| |<---->| D |
+---+ +-------------+ +---+
Figure 2: RTP Mixer with Only Unicast Paths
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 [RFC4383] -- 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.
12.1.3. Differentiated Treatment of RTP Streams
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 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 [W3C.WebRTC] 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" [RFC8837].
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
[RFC8837] and [RFC7657] 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.
12.2. Media Source, RTP Streams, and Participant Identification
12.2.1. Media Source Identification
Each 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 Section 4.8. 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 Section 4.1, 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 [RFC6465] is being used
in the session (see Section 5.2.3), 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.
12.2.2. SSRC Collision Detection
The 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 Section 8.2 of [RFC3550]). 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 [RFC5576]. 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)" [RFC5576] 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 [RFC4588]. 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.
12.2.3. Media Synchronization Context
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, 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.
13. Security Considerations
The overall security architecture for WebRTC is described in
[RFC8827], and security considerations for the WebRTC framework are
described in [RFC8826]. 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)" [RFC5124] 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 [RFC5764], as defined by Section 5.5
of [RFC8827].
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. Section 4.9 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 [RFC3556] 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) [RFC4585]. The risks are
as follows:
1. 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;
2. 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
3. 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)
[RFC4585], as discussed in Section 6.1 of [RFC8108].
Premature participant timeout can be avoided by using the fixed
(nonreduced) minimum interval when calculating the participant
timeout (see Section 4.1 of this memo and Section 7.1.2 of
[RFC8108]). 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 [RFC3550] (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 [RFC6562] apply when using variable bitrate (VBR)
audio codecs such as Opus (see Section 4.3 for discussion of mandated
audio codecs). The guidelines in [RFC6562] also apply, but are of
lesser importance, when using the client-to-mixer audio level header
extensions (Section 5.2.2) or the mixer-to-client audio level header
extensions (Section 5.2.3). The use of the encryption of the header
extensions 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
Section 12.1.1, 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 [RFC7667].
14. IANA Considerations
This document has no IANA actions.
15. References
15.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736,
DOI 10.17487/RFC2736, December 1999,
<https://www.rfc-editor.org/info/rfc2736>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<https://www.rfc-editor.org/info/rfc3551>.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, DOI 10.17487/RFC3556, July 2003,
<https://www.rfc-editor.org/info/rfc3556>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<https://www.rfc-editor.org/info/rfc4585>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<https://www.rfc-editor.org/info/rfc4588>.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
<https://www.rfc-editor.org/info/rfc4961>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <https://www.rfc-editor.org/info/rfc5104>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <https://www.rfc-editor.org/info/rfc5124>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <https://www.rfc-editor.org/info/rfc5506>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<https://www.rfc-editor.org/info/rfc5761>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, DOI 10.17487/RFC6051, November 2010,
<https://www.rfc-editor.org/info/rfc6051>.
[RFC6464] Lennox, J., Ed., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464,
DOI 10.17487/RFC6464, December 2011,
<https://www.rfc-editor.org/info/rfc6464>.
[RFC6465] Ivov, E., Ed., Marocco, E., Ed., and J. Lennox, "A Real-
time Transport Protocol (RTP) Header Extension for Mixer-
to-Client Audio Level Indication", RFC 6465,
DOI 10.17487/RFC6465, December 2011,
<https://www.rfc-editor.org/info/rfc6465>.
[RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP", RFC 6562,
DOI 10.17487/RFC6562, March 2012,
<https://www.rfc-editor.org/info/rfc6562>.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904,
DOI 10.17487/RFC6904, April 2013,
<https://www.rfc-editor.org/info/rfc6904>.
[RFC7007] Terriberry, T., "Update to Remove DVI4 from the
Recommended Codecs for the RTP Profile for Audio and Video
Conferences with Minimal Control (RTP/AVP)", RFC 7007,
DOI 10.17487/RFC7007, August 2013,
<https://www.rfc-editor.org/info/rfc7007>.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
September 2013, <https://www.rfc-editor.org/info/rfc7022>.
[RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
Clock Rates in an RTP Session", RFC 7160,
DOI 10.17487/RFC7160, April 2014,
<https://www.rfc-editor.org/info/rfc7160>.
[RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds",
RFC 7164, DOI 10.17487/RFC7164, March 2014,
<https://www.rfc-editor.org/info/rfc7164>.
[RFC7742] Roach, A.B., "WebRTC Video Processing and Codec
Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
<https://www.rfc-editor.org/info/rfc7742>.
[RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
<https://www.rfc-editor.org/info/rfc7874>.
[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", RFC 8083,
DOI 10.17487/RFC8083, March 2017,
<https://www.rfc-editor.org/info/rfc8083>.
[RFC8108] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session",
RFC 8108, DOI 10.17487/RFC8108, March 2017,
<https://www.rfc-editor.org/info/rfc8108>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8285] Singer, D., Desineni, H., and R. Even, Ed., "A General
Mechanism for RTP Header Extensions", RFC 8285,
DOI 10.17487/RFC8285, October 2017,
<https://www.rfc-editor.org/info/rfc8285>.
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/info/rfc8825>.
[RFC8826] Rescorla, E., "Security Considerations for WebRTC",
RFC 8826, DOI 10.17487/RFC8826, January 2021,
<https://www.rfc-editor.org/info/rfc8826>.
[RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
DOI 10.17487/RFC8827, January 2021,
<https://www.rfc-editor.org/info/rfc8827>.
[RFC8843] Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", RFC 8843,
DOI 10.17487/RFC8843, January 2021,
<https://www.rfc-editor.org/info/rfc8843>.
[RFC8854] Uberti, J., "WebRTC Forward Error Correction
Requirements", RFC 8854, DOI 10.17487/RFC8854, January
2021, <https://www.rfc-editor.org/info/rfc8854>.
[RFC8858] Holmberg, C., "Indicating Exclusive Support of RTP and RTP
Control Protocol (RTCP) Multiplexing Using the Session
Description Protocol (SDP)", RFC 8858,
DOI 10.17487/RFC8858, January 2021,
<https://www.rfc-editor.org/info/rfc8858>.
[RFC8860] Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session",
RFC 8860, DOI 10.17487/RFC8860, January 2021,
<https://www.rfc-editor.org/info/rfc8860>.
[RFC8861] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
"Sending Multiple RTP Streams in a Single RTP Session:
Grouping RTP Control Protocol (RTCP) Reception Statistics
and Other Feedback", RFC 8861, DOI 10.17487/RFC8861,
January 2021, <https://www.rfc-editor.org/info/rfc8861>.
[W3C.WD-mediacapture-streams]
Jennings, C., Aboba, B., Bruaroey, J-I., and H. Boström,
"Media Capture and Streams", W3C Candidate Recommendation,
<https://www.w3.org/TR/mediacapture-streams/>.
[W3C.WebRTC]
Jennings, C., Boström, H., and J-I. Bruaroey, "WebRTC 1.0:
Real-time Communication Between Browsers", W3C Proposed
Recommendation, <https://www.w3.org/TR/webrtc/>.
15.2. Informative References
[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
"RTP Control Protocol Extended Reports (RTCP XR)",
RFC 3611, DOI 10.17487/RFC3611, November 2003,
<https://www.rfc-editor.org/info/rfc3611>.
[RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient
Stream Loss-Tolerant Authentication (TESLA) in the Secure
Real-time Transport Protocol (SRTP)", RFC 4383,
DOI 10.17487/RFC4383, February 2006,
<https://www.rfc-editor.org/info/rfc4383>.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
<https://www.rfc-editor.org/info/rfc5576>.
[RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP
Control Protocol (RTCP)", RFC 5968, DOI 10.17487/RFC5968,
September 2010, <https://www.rfc-editor.org/info/rfc5968>.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263,
DOI 10.17487/RFC6263, June 2011,
<https://www.rfc-editor.org/info/rfc6263>.
[RFC6792] Wu, Q., Ed., Hunt, G., and P. Arden, "Guidelines for Use
of the RTP Monitoring Framework", RFC 6792,
DOI 10.17487/RFC6792, November 2012,
<https://www.rfc-editor.org/info/rfc6792>.
[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015,
<https://www.rfc-editor.org/info/rfc7478>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<https://www.rfc-editor.org/info/rfc7656>.
[RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services
(Diffserv) and Real-Time Communication", RFC 7657,
DOI 10.17487/RFC7657, November 2015,
<https://www.rfc-editor.org/info/rfc7657>.
[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
DOI 10.17487/RFC7667, November 2015,
<https://www.rfc-editor.org/info/rfc7667>.
[RFC8088] Westerlund, M., "How to Write an RTP Payload Format",
RFC 8088, DOI 10.17487/RFC8088, May 2017,
<https://www.rfc-editor.org/info/rfc8088>.
[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", RFC 8445,
DOI 10.17487/RFC8445, July 2018,
<https://www.rfc-editor.org/info/rfc8445>.
[RFC8829] Uberti, J., Jennings, C., and E. Rescorla, Ed.,
"JavaScript Session Establishment Protocol (JSEP)",
RFC 8829, DOI 10.17487/RFC8829, January 2021,
<https://www.rfc-editor.org/info/rfc8829>.
[RFC8830] Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", RFC 8830,
DOI 10.17487/RFC8830, January 2021,
<https://www.rfc-editor.org/info/rfc8830>.
[RFC8836] Jesup, R. and Z. Sarker, Ed., "Congestion Control
Requirements for Interactive Real-Time Media", RFC 8836,
DOI 10.17487/RFC8836, January 2021,
<https://www.rfc-editor.org/info/rfc8836>.
[RFC8837] Jones, P., Dhesikan, S., Jennings, C., and D. Druta,
"Differentiated Services Code Point (DSCP) Packet Markings
for WebRTC QoS", RFC 8837, DOI 10.17487/RFC8837, January
2021, <https://www.rfc-editor.org/info/rfc8837>.
[RFC8872] Westerlund, M., Burman, B., Perkins, C., Alvestrand, H.,
and R. Even, "Guidelines for Using the Multiplexing
Features of RTP to Support Multiple Media Streams",
RFC 8872, DOI 10.17487/RFC8872, January 2021,
<https://www.rfc-editor.org/info/rfc8872>.
Acknowledgements
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.
Authors' Addresses
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow
G12 8QQ
United Kingdom
Email: csp@csperkins.org
URI: https://csperkins.org/
Magnus Westerlund
Ericsson
Torshamnsgatan 23
SE-164 80 Kista
Sweden
Email: magnus.westerlund@ericsson.com
Jörg Ott
Technical University Munich
Department of Informatics
Chair of Connected Mobility
Boltzmannstrasse 3
85748 Garching
Germany
Email: ott@in.tum.de
ERRATA
