RTCWEB Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Standards Track M. Westerlund
Expires: October 25, 2014 Ericsson
J. Ott
Aalto University
April 23, 2014
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-13
Abstract
The Web Real-Time Communication (WebRTC) framework provides support
for direct interactive rich communication using audio, video, text,
collaboration, games, etc. between two peers' web-browsers. This
memo describes the media transport aspects of the WebRTC framework.
It specifies how the Real-time Transport Protocol (RTP) is used in
the WebRTC context, and gives requirements for which RTP features,
profiles, and extensions need to be supported.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on October 25, 2014.
Copyright Notice
Copyright (c) 2014 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
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
Perkins, et al. Expires October 25, 2014 [Page 1]
Internet-Draft RTP for WebRTC April 2014
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5
4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7
4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 10
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11
4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11
4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 12
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12
5.1. Conferencing Extensions and Topologies . . . . . . . . . 12
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 14
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 14
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 14
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 15
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 15
5.1.6. Temporary Maximum Media Stream Bit Rate Request
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 15
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 16
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 16
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 16
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 17
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 17
6.1. Negative Acknowledgements and RTP Retransmission . . . . 17
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 18
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 19
7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 20
7.2. RTCP Limitations for Congestion Control . . . . . . . . . 20
7.3. Congestion Control Interoperability and Legacy Systems . 22
8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 23
9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 24
10. Signalling Considerations . . . . . . . . . . . . . . . . . . 24
11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 25
12. RTP Implementation Considerations . . . . . . . . . . . . . . 28
12.1. Configuration and Use of RTP Sessions . . . . . . . . . 28
Perkins, et al. Expires October 25, 2014 [Page 2]
Internet-Draft RTP for WebRTC April 2014
12.1.1. Use of Multiple Media Sources Within an RTP Session 28
12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 29
12.1.3. Differentiated Treatment of RTP Packet Streams . . . 34
12.2. Media Source, RTP Packet Streams, and Participant
Identification . . . . . . . . . . . . . . . . . . . . . 35
12.2.1. Media Source . . . . . . . . . . . . . . . . . . . . 36
12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36
12.2.3. Media Synchronisation Context . . . . . . . . . . . 37
13. Security Considerations . . . . . . . . . . . . . . . . . . . 38
14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39
15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39
16. References . . . . . . . . . . . . . . . . . . . . . . . . . 39
16.1. Normative References . . . . . . . . . . . . . . . . . . 39
16.2. Informative References . . . . . . . . . . . . . . . . . 42
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44
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 signalling, 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-aware end-
points, 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 [I-D.ietf-rtcweb-overview].
The structure of this memo is as follows. Section 2 outlines our
rationale in 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 that can be used in the
WebRTC context. Section 9 gives some guidelines for future
incorporation of other RTP and RTP Control Protocol (RTCP) extensions
Perkins, et al. Expires October 25, 2014 [Page 3]
Internet-Draft RTP for WebRTC April 2014
into this framework. Section 10 describes requirements placed on the
signalling 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
to support many new media encodings, but raises the question of what
extensions are to be supported by new implementations. The
development of the WebRTC framework provides an opportunity to review
the available RTP features and extensions, and to define a common
baseline feature set for all WebRTC implementations of RTP. This
builds on the past 20 years development of RTP to mandate the use of
extensions that have shown widespread utility, while still remaining
compatible with the wide installed base of RTP implementations where
possible.
RTP and RTCP extensions that are not discussed in this document can
be implemented by WebRTC end-points if they are beneficial for new
use cases. However, they are not necessary to address the WebRTC use
cases and requirements identified in
[I-D.ietf-rtcweb-use-cases-and-requirements].
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", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. The RFC
2119 interpretation of these key words applies only when written in
ALL CAPS. Lower- or mixed-case uses of these key words are not to be
interpreted as carrying special significance in this memo.
We define the following additional terms:
Perkins, et al. Expires October 25, 2014 [Page 4]
Internet-Draft RTP for WebRTC April 2014
WebRTC MediaStream: The MediaStream concept defined by the W3C in
the WebRTC API [W3C.WD-mediacapture-streams-20130903].
Transport-layer Flow: A uni-directional flow of transport packets
that are identified by having a particular 5-tuple of source IP
address, source port, destination IP address, destination port,
and transport protocol used.
Bi-directional Transport-layer Flow: A bi-directional transport-
layer flow is a transport-layer flow that is symmetric. That is,
the transport-layer flow in the reverse direction has a 5-tuple
where the source and destination address and ports are swapped
compared to the forward path transport-layer flow, and the
transport protocol is the same.
This document uses the terminology from
[I-D.ietf-avtext-rtp-grouping-taxonomy]. Other terms are used
according to their definitions from the RTP Specification [RFC3550].
We especially note the following frequently used terms: RTP Packet
Stream, RTP Session, and End-point.
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 implementations 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 in all WebRTC applications.
The following RTP and RTCP features are sometimes omitted in limited
functionality implementations of RTP, but are REQUIRED in all WebRTC
implementations:
o Support for use of multiple simultaneous SSRC values in a single
RTP session, including support for RTP end-points that send many
SSRC values simultaneously, following [RFC3550] and
[I-D.ietf-avtcore-rtp-multi-stream]. Support for the RTCP
optimisations for multi-SSRC sessions defined in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] is RECOMMENDED.
Perkins, et al. Expires October 25, 2014 [Page 5]
Internet-Draft RTP for WebRTC April 2014
o Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (see also Section 4.8).
o Support for reception of RTP data packets containing CSRC lists,
as generated by RTP mixers, and RTCP packets relating to CSRCs.
o Sending correct synchronisation information in the RTCP Sender
Reports, to allow receivers to implement lip-synchronisation;
support for the rapid RTP synchronisation extensions (see
Section 5.2.1) is RECOMMENDED.
o Support for multiple synchronisation contexts. Participants that
send multiple simultaneous RTP packet streams SHOULD do so as part
of a single synchronisation context, using a single RTCP CNAME for
all streams and allowing receivers to play the streams out in a
synchronised manner. For compatibility with potential future
versions of this specification, or for interoperability with non-
WebRTC devices through a gateway, receivers MUST support multiple
synchronisation contexts, indicated by the use of multiple RTCP
CNAMEs in an RTP session. This specification requires the usage
of a single CNAME when sending RTP Packet Streams in some
circumstances, see Section 4.9.
o Support for sending and receiving RTCP SR, RR, SDES, and BYE
packet types, with OPTIONAL support for other RTCP packet types
unless mandated by other parts of this specification;
implementations MUST ignore unknown RTCP packet types. Note that
additional RTCP Packet types are used by the RTP/SAVPF Profile
(Section 4.2) and the other RTCP extensions (Section 5).
o Support for multiple end-points in a single RTP session, and for
scaling the RTCP transmission interval according to the number of
participants in the session; support for randomised RTCP
transmission intervals to avoid synchronisation of RTCP reports;
support for RTCP timer reconsideration.
o 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] is
RECOMMENDED.
It is known that a significant number of legacy RTP implementations,
especially those targeted at VoIP-only systems, do not support all of
the above features, and in some cases do not support RTCP at all.
Implementers are advised to consider the requirements for graceful
degradation when interoperating with legacy implementations.
Perkins, et al. Expires October 25, 2014 [Page 6]
Internet-Draft RTP for WebRTC April 2014
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 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 end-point 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
end-points via a gateway is to set the trr-int parameter to a
value representing 4 seconds).
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 implementations MUST
NOT send packets using the basic RTP/AVP profile or the RTP/AVPF
profile; they MUST employ the full RTP/SAVPF profile to protect all
RTP and RTCP packets that are generated (i.e., implementations MUST
use SRTP and SRTCP). The RTP/SAVPF profile MUST be configured using
the cipher suites, DTLS-SRTP protection profiles, keying mechanisms,
and other parameters described in [I-D.ietf-rtcweb-security-arch].
4.3. Choice of RTP Payload Formats
The set of mandatory to implement codecs and RTP payload formats for
WebRTC is not specified in this memo, instead they are defined in
separate specifications, such as [I-D.ietf-rtcweb-audio].
Implementations can support any codec for which an RTP payload format
and associated signalling is defined. Implementation cannot assume
that the other participants in an RTP session understand any RTP
Perkins, et al. Expires October 25, 2014 [Page 7]
Internet-Draft RTP for WebRTC April 2014
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.
End-points 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 signalling context. This association is possible provided unique
RTP payload type numbers are used in each context. For example, an
RTP packet stream can be associated with an SDP "m=" line by
comparing the RTP payload type numbers used by the RTP packet stream
with payload types signalled in the "a=rtpmap:" lines in the media
sections of the SDP. If RTP packet streams are being associated with
signalling contexts based on the RTP payload type, then the
assignment of RTP payload type numbers MUST be unique across
signalling contexts; 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 signalling 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 different "m=" lines in an SDP bundle group
[I-D.ietf-mmusic-sdp-bundle-negotiation] form a single RTP session).
An end-point that has signalled support for multiple RTP payload
formats SHOULD be able to accept data in any of those payload formats
at any time, unless it has previously signalled limitations on its
decoding capability. This requirement is constrained if several
types of media (e.g., audio and video) are sent in the same RTP
session. In such a case, a source (SSRC) is restricted to switching
only between the RTP payload formats signalled for the type of media
that is being sent by that source; see Section 4.4. To support rapid
rate adaptation by changing codec, RTP does not require advance
signalling for changes between RTP payload formats used by a single
SSRC that were signalled during session set-up.
An RTP sender that changes between two RTP payload types that use
different RTP clock rates MUST follow the recommendations in
Section 4.1 of [RFC7160]. RTP receivers MUST follow the
recommendations in Section 4.3 of [RFC7160] in order to support
Perkins, et al. Expires October 25, 2014 [Page 8]
Internet-Draft RTP for WebRTC April 2014
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 end-points communicating using RTP is
known as an RTP session [RFC3550]. An end-point 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 implementations of RTP are REQUIRED to implement support for
multimedia sessions in this way, separating each session using
different transport-layer flows for compatibility with legacy
systems.
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 [I-D.ietf-avtcore-multi-media-rtp-session]. If
multiple types of media are to be used in a single RTP session, all
participants in that RTP session MUST agree to this usage. In an SDP
context, [I-D.ietf-mmusic-sdp-bundle-negotiation] 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 are
suitable can be found in [I-D.ietf-avtcore-multiplex-guidelines].
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 port for RTP and
one port for RTCP). With the increased use of Network Address/Port
Translation (NAT/NAPT) this has become problematic, since maintaining
multiple NAT bindings can be costly. It also complicates firewall
administration, since multiple ports need to be opened to allow RTP
traffic. To reduce these costs and session set-up times, support for
multiplexing RTP data packets and RTCP control packets on a single
transport-layer flow for each RTP session is REQUIRED, provided it is
Perkins, et al. Expires October 25, 2014 [Page 9]
Internet-Draft RTP for WebRTC April 2014
negotiated in the signalling channel before use as specified in
[RFC5761]. For backwards compatibility, implementations are also
REQUIRED to support RTP and RTCP sent on separate transport-layer
flows.
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, and is the recommend method for
application level keep-alives of RTP sessions [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 Sender Report (SR)
or Receiver Report (RR) packet. When using frequent RTCP feedback
messages under the RTP/AVPF Profile [RFC4585] these statistics are
not needed in every packet, and unnecessarily increase the mean RTCP
packet size. This can limit the frequency at which RTCP packets can
be sent within the RTCP bandwidth share.
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 behaviour. Support for non-
compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be
negotiated using the signalling channel before use. For backwards
compatibility, implementations are also REQUIRED to support the use
of compound RTCP feedback packets if the remote end-point does not
agree to the use of non-compound RTCP in the signalling 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 bi-directional transport-layer flows.
This will keep alive the NAT and firewall pinholes, and help indicate
consent that the receive direction is a transport-layer flow the
intended recipient actually wants. In addition, it saves resources,
specifically ports at the end-points, but also in the network as NAT
mappings or firewall state is not unnecessary bloated. The amount of
per flow QoS state kept in the network is also reduced.
Perkins, et al. Expires October 25, 2014 [Page 10]
Internet-Draft RTP for WebRTC April 2014
4.8. Choice of RTP Synchronisation Source (SSRC)
Implementations are REQUIRED to support signalled RTP synchronisation
source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined
in Section 4.1 and Section 5 of [RFC5576]. Implementations MUST also
support the "previous-ssrc" source attribute defined in Section 6.2
of [RFC5576]. Other per-SSRC attributes defined in [RFC5576] MAY be
supported.
Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP
session is OPTIONAL. Implementations MUST be prepared to accept RTP
and RTCP packets using SSRCs that have not been explicitly signalled
ahead of time. Implementations MUST support random SSRC assignment,
and MUST support SSRC collision detection and resolution, according
to [RFC3550]. When using signalled 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 are 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
signalled this is straightforward (in SDP the "a=ssrc:" line will be
at the media level, allowing a direct association with an "m=" line).
If SSRCs are not signalled, the RTP payload type numbers used in an
RTP packet stream are often sufficient to associate that packet
stream with a signalling context (e.g., if RTP payload type numbers
are assigned as described in 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 end-point. While the Synchronisation Source
(SSRC) identifier for an RTP end-point can change if a collision is
detected, or when the RTP application is restarted, its RTCP CNAME is
meant to stay unchanged for the duration of a RTCPeerConnection
[W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely
identified and associated with their RTP packet streams within a set
of related RTP sessions.
Each RTP end-point MUST have at least one RTCP CNAME, and that RTCP
CNAME MUST be unique within the RTCPeerConnection. RTCP CNAMEs
identify a particular synchronisation context, i.e., all SSRCs
associated with a single RTCP CNAME share a common reference clock.
If an end-point has SSRCs that are associated with several
Perkins, et al. Expires October 25, 2014 [Page 11]
Internet-Draft RTP for WebRTC April 2014
unsynchronised reference clocks, and hence different synchronisation
contexts, it will need to use multiple RTCP CNAMEs, one for each
synchronisation context.
Taking the discussion in Section 11 into account, a WebRTC end-point
MUST NOT use more than one RTCP CNAME in the RTP sessions belonging
to single RTCPeerConnection (that is, an RTCPeerConnection forms a
synchronisation context). RTP middleboxes MAY generate RTP packet
streams associated with more than one RTCP CNAME, to allow them to
avoid having to resynchronize media from multiple different end-
points part of a multi-party 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 as an existing
RTCPeerConnection within their common same-origin context.
An WebRTC end-point 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 regarding handling of leap seconds to limit their
impact on RTP media playout and synchronization given in [RFC7164]
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 application context. One set of these
extensions is related to conferencing, while others are more generic
in nature. The following subsections describe the various RTP
extensions mandated or suggested for use within the WebRTC context.
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
Perkins, et al. Expires October 25, 2014 [Page 12]
Internet-Draft RTP for WebRTC April 2014
an IP multicast group to distribute the RTP packet streams. These
topologies can be implemented in a number of ways as discussed in
[I-D.ietf-avtcore-rtp-topologies-update].
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 implementations
are not expected to support topologies based on IP multicast groups
or to support mesh-based topologies, such as a point-to-multipoint
mesh configured as a single RTP session (Topo-Mesh in the terminology
of [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to-
multipoint mesh constructed using several RTP sessions, implemented
in the WebRTC context using independent RTCPeerConnections, can be
expected to be utilised by WebRTC applications and needs to be
supported.
WebRTC implementations of RTP endpoints implemented according to this
memo are expected to support all the topologies described in
[I-D.ietf-avtcore-rtp-topologies-update] 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 the WebRTC environment. Specifically:
o Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
since they make the use of RTCP for congestion control and quality
of service reports problematic (see Section 3.8 of
[I-D.ietf-avtcore-rtp-topologies-update]).
o The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
SHOULD NOT be used because its safe use requires a congestion
control algorithm or RTP circuit breaker that handles point to
multipoint, which has not yet been standardised.
The following topology can be used, however it has some issues worth
noting:
o 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
Perkins, et al. Expires October 25, 2014 [Page 13]
Internet-Draft RTP for WebRTC April 2014
these functions (see section 3.9 of
[I-D.ietf-avtcore-rtp-topologies-update]).
The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
designed to be used with centralised conferencing, where an RTP
middlebox (e.g., a conference bridge) receives a participant's RTP
packet streams and distributes them to the other participants. These
extensions are not necessary for interoperability; an RTP end-point
that does not implement these extensions will work correctly, but
might offer poor performance. Support for the listed extensions will
greatly improve the quality of experience and, to provide a
reasonable baseline quality, some of these extensions are mandatory
to be supported by WebRTC end-points.
The RTCP conferencing extensions are defined in Extended RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
AVPF) [RFC4585] and the memo on Codec Control Messages (CCM) in RTP/
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 the 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 senders MUST understand and
react to FIR feedback messages they receiver, since this greatly
improves the user experience when using centralised 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 fulfil
the request. WebRTC senders 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
Perkins, et al. Expires October 25, 2014 [Page 14]
Internet-Draft RTP for WebRTC April 2014
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 learnt that encoder-decoder synchronisation has been
lost, then a known as correct reference picture can be used as a base
for future coding. The RPSI message allows this to be signalled.
Receivers that detect that encoder-decoder synchronisation has been
lost SHOULD generate an RPSI feedback message if codec being used
supports reference picture selection. A RTP packet stream sender
that receives such an RPSI message SHOULD act on that messages to
change the reference picture, if it is possible to do so within the
available bandwidth constraints, and with the codec being used.
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 TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of
the Codec Control Messages [RFC5104]. This request and its
notification message are used by a media receiver to inform the
sending party that there is a current limitation on the amount of
bandwidth available to this receiver. This can be various reasons
for this: for example, an RTP mixer can use this message to limit the
media rate of the sender being forwarded by the mixer (without doing
media transcoding) to fit the bottlenecks existing towards the other
session participants. WebRTC senders are REQUIRED to implement
support for TMMBR messages, and MUST follow bandwidth limitations set
by a TMMBR message received for their SSRC. The sending of TMMBR
requests is OPTIONAL.
Perkins, et al. Expires October 25, 2014 [Page 15]
Internet-Draft RTP for WebRTC April 2014
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 the WebRTC context, but if they are used,
they MUST be formatted and signalled following the general mechanism
for RTP header extensions defined in [RFC5285], since this gives
well-defined semantics to RTP header extensions.
As noted in [RFC5285], the requirement from the RTP specification
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 signalled (e.g., as defined by the payload type). Valid examples
of RTP header extensions might include metadata that is additional to
the usual RTP information, but that can safely be ignored without
compromising interoperability.
5.2.1. Rapid Synchronisation
Many RTP sessions require synchronisation between audio, video, and
other content. This synchronisation is performed by receivers, using
information contained in RTCP SR packets, as described in the RTP
specification [RFC3550]. This basic mechanism can be slow, however,
so it is RECOMMENDED that the rapid RTP synchronisation extensions
described in [RFC6051] be implemented in addition to RTCP SR-based
synchronisation. The rapid synchronisation extensions use the
general RTP header extension mechanism [RFC5285], which requires
signalling, but are otherwise backwards compatible.
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 mixer. 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 [RFC6464] header extension is
RECOMMENDED to be implemented. If this header extension is
implemented, it is REQUIRED that implementations are capable of
Perkins, et al. Expires October 25, 2014 [Page 16]
Internet-Draft RTP for WebRTC April 2014
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 is used,
unless the encryption has been explicitly disabled through API or
signalling.
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 mix by a RTP mixer. This enables a user interface to
indicate the relative activity level of each session participant,
rather than just being included or not based on the CSRC field. This
is a pure optimisations of non critical functions, and is hence
OPTIONAL to implement. If this header extension is implemented, it
is REQUIRED that implementations are 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 is used, unless the encryption has
been explicitly disabled through API or signalling.
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 all add overhead compared to a non-robust stream. The overhead
needs to be considered, and the aggregate bit-rate MUST be rate
controlled to avoid causing network congestion (see 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 sub-sections 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 optimise
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 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
Perkins, et al. Expires October 25, 2014 [Page 17]
Internet-Draft RTP for WebRTC April 2014
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 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 caused increased packet loss if
the original packet loss was caused by network congestion. We note,
however, that retransmission of an important lost packet to repair
decoder state can have lower cost than sending a full intra frame.
It is not appropriate to blindly retransmit RTP packets in response
to a NACK. The importance of lost packets and the likelihood of them
arriving in time to be useful needs to be considered before RTP
retransmission is used.
Receivers are REQUIRED to implement support for RTP retransmission
packets [RFC4588]. Senders MAY send RTP retransmission packets in
response to NACKs if the RTP retransmission payload format has been
negotiated for the session, and if the sender believes it is useful
to send a retransmission of the packet(s) referenced in the NACK. An
RTP sender does not need to retransmit every NACKed packet.
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, others operate across RTP packets and can be used
with any payload format. It needs to be noted that using redundant
encoding or FEC will lead to increased play out delay, which needs to
be considered when choosing the redundancy or FEC formats and their
respective parameters.
If an RTP payload format negotiated for use in a RTCPeerConnection
supports redundant transmission or FEC as a standard feature of that
payload format, then that support MAY be used in the
RTCPeerConnection, subject to any appropriate signalling.
Perkins, et al. Expires October 25, 2014 [Page 18]
Internet-Draft RTP for WebRTC April 2014
There are several block-based FEC schemes that are designed for use
with RTP independent of the chosen RTP payload format. At the time
of this writing there is no consensus on which, if any, of these FEC
schemes is appropriate for use in the WebRTC context. Accordingly,
this memo makes no recommendation on the choice of block-based FEC
for WebRTC use.
7. WebRTC Use of RTP: Rate Control and Media Adaptation
WebRTC will be used in heterogeneous network environments using a
variety set of link technologies, including both wired and wireless
links, to interconnect potentially large groups of users around the
world. As a result, the network paths between users can have widely
varying one-way delays, available bit-rates, load levels, and traffic
mixtures. Individual end-points can send one or more RTP packet
streams to each participant in a WebRTC conference, and there can be
several participants. Each of these RTP packet streams can contain
different types of media, and the type of media, bit rate, and number
of RTP packet streams as well as transport-layer flows can be highly
asymmetric. Non-RTP traffic can share the network paths with RTP
transport-layer flows. Since the network environment is not
predictable or stable, WebRTC end-points 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 implementation is very
dependent on effective adaptation of the media to the limitations of
the network. End-points have to be designed so they do not transmit
significantly more data than the network path can support, except for
very short time periods, otherwise high levels of network packet loss
or delay spikes will occur, causing media quality degradation. The
limiting factor on the capacity of the network path might be the link
bandwidth, or it might be competition with other traffic on the link
(this can be non-WebRTC traffic, traffic due to other WebRTC flows,
or even competition with other WebRTC flows in the same session).
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 [I-D.ietf-rmcat-cc-requirements].
It is expected that a future version of this memo will mandate the
use of a congestion control algorithm that satisfies these
requirements.
Perkins, et al. Expires October 25, 2014 [Page 19]
Internet-Draft RTP for WebRTC April 2014
7.1. Boundary Conditions and Circuit Breakers
In the absence of a concrete congestion control algorithm, all WebRTC
implementations MUST implement the RTP circuit breaker algorithm that
is described in [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP
circuit breaker is designed to enable applications to recognise and
react to situations of extreme network congestion. However, since
the RTP circuit breaker might not be triggered until congestion
becomes extreme, it cannot be considered a substitute for congestion
control, and applications MUST also implement congestion control to
allow them to adapt to changes in network capacity. Any future RTP
congestion control algorithms are expected to operate within the
envelope allowed by the circuit breaker.
The session establishment signalling will also necessarily establish
boundaries to which the media bit-rate will conform. The choice of
media codecs provides upper- and lower-bounds on the supported bit-
rates that the application can utilise to provide useful quality, and
the packetization choices that exist. In addition, the signalling
channel can establish maximum media bit-rate boundaries using the SDP
"b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media
Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo).
The combination of media codec choice and signalled bandwidth limits
SHOULD be used to limit traffic based on known bandwidth limitations,
for example the capacity of the edge links, to the extent possible.
7.2. RTCP Limitations for Congestion Control
Experience with the congestion control algorithms of TCP [RFC5681],
TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
that feedback on packet arrivals needs to be sent frequently (roughly
once per round trip time is common). We note that the real-time
media traffic might not be able to adapt to changing path conditions
as rapidly as elastic applications using TCP, but frequent feedback,
perhaps on the order of once per video frame, is still needed to
allow the congestion control algorithm to track the path dynamics.
As an example of the type of RTCP congestion control feedback that is
possible, consider one of the simplest scenarios for WebRTC: a point
to point video call between two end systems. There will be four RTP
flows in this scenario, two audio and two video, with all four flows
being active for essentially all the time (the audio flows will
likely use voice activity detection and comfort noise to reduce the
packet rate during silent periods, but doesn't cause transmissions to
stop). Assume all four flows are sent in a single RTP session, each
using a separate SSRC. Further, assume each SSRC sends RTCP reports
for all other SSRCs in the session (i.e., the optimisations in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] are not used, giving
Perkins, et al. Expires October 25, 2014 [Page 20]
Internet-Draft RTP for WebRTC April 2014
the worst case for the RTCP overhead). When all members are senders
like this, the RTCP timing rules in Sections 6.2 and 6.3 of [RFC3550]
and [RFC4585] reduce to:
rtcp_interval = avg_rtcp_size * n / rtcp_bw
where avg_rtcp_size is measured in octets, and the rtcp_bw is the
bandwidth available for RTCP. The average RTCP size will depend on
the amount of feedback that is sent in each RTCP packet, on the
number of members in the session, and on the size of source
description (RTCP SDES) information sent. As a baseline, each RTCP
packet will be a compound RTCP packet that contains an RTCP SR and an
RTCP SDES packet. In the scenario above, each RTCP SR packet will
contain three report blocks, once for each of the other RTP SSRCs
sending data, for a total of 100 octets (this is 8 octets header, 20
octets sender info, and 3 * 24 octets report blocks). The RTCP SDES
packet will comprise a header (4 octets), an originating SSRC (4
octets), a CNAME chunk, and padding. If the CNAME follows [RFC7022]
and it will be 19 octets in size, and require 1 octet of padding.
The resulting compound RTCP packet will be 128 octets in size. If
sent in UDP/IPv4 with no IP options and using Secure RTP, which adds
20 (IPv4) + 8 (UDP) + 14 (SRTP with 80 bit Authentication tag), the
avg_rtcp_size will therefore be 170 octets, including the header
overhead. The value n is this scenario is 4, and the rtcp_bw is
assumed to be 5% of the session bandwidth.
If it is desired to send RTCP feedback packets on average 30 times
per second, to correspond to one RTCP report every frame for 30fps
video, we can invert the above rtcp_interval calculation to get an
rtcp_bw that gives an interval of 1/30th of a second or lower. This
corresponds to an rtcp_bw of 20400 octets per second (since 1/30 =
170 * 4 / 20400). This is 163200 bits per second, which if 5% of the
session bandwidth, gives a session bandwidth of approximately 3.3Mbps
(i.e., 3.3Mbps media rate, plus an additional 5% for RTCP, to give a
total data rate of approximately 3.4Mbps). That is, RTCP can report
on every frame of video provided the session bandwidth is 3.3Mbps or
larger, when every SSRC sends a report for every video frame. Please
note that the actual RTCP transmission intervals will be within the
interval [0.0135, 0.0406]s, but maintaining an average RTCP
transmission interval of 0.033s.
Note: To achieve the RTCP transmission intervals above the RTP/
SAVPF profile with T_rr_interval=0 is used, since even when using
the reduced minimal transmission interval, the RTP/SAVP profile
would only allow sending RTCP at most every 0.11s (every third
frame of video). Using RTP/SAVPF with T_rr_interval=0 however is
capable of fully utilizing the configured 5% RTCP bandwidth
fraction.
Perkins, et al. Expires October 25, 2014 [Page 21]
Internet-Draft RTP for WebRTC April 2014
If additional feedback beyond the standard report block is needed,
the session bandwidth needed will increase. For example, with an
additional 20 octets data being reported in each RTCP packet, the
session bandwidth needed increases to 3.5Mbps for every SSRC to be
able to report on every frame. However, the above baseline might not
be the most appropriate usage of the RTCP bandwidth. Depending on
needs, a less frequent usage of regular RTCP compound packets,
controlled by T_rr_interval combined with using the reduced size RTCP
packets, can achieve more frequent and useful reporting. Also the
reporting requirements defined in
[I-D.ietf-avtcore-rtp-multi-stream-optimisation] will reduced the
amount of bandwidth consumed for reporting when each endpoint has
multiple SSRCs.
Calculations such as these show that RTCP cannot be used to send per-
packet congestion feedback. RTCP can, however, be used to send
congestion feedback on each frame of video sent in an interactive
video conferencing scenario, provided the RTCP parameters are
correctly configured and the overall session bandwidth exceeds a
couple of megabits per second (the exact rate depending on the number
of session participants, the RTCP bandwidth fraction, and whether
audio and video are sent in one or two RTP sessions). Using similar
calculations, it can be shown that RTCP can likely also be used to
send feedback on a per-RTT basis, provided the RTT is not too low.
Interactive communication might not be able to afford to wait for
packet losses to occur to indicate congestion, because an increase in
play out delay due to queuing (most prominent in wireless networks)
can easily lead to packets being dropped due to late arrival at the
receiver. Therefore, more sophisticated cues might need to be
reported -- to be defined in a suitable congestion control framework
as noted above -- which, in turn, increase the report size again.
For example, different RTCP XR report blocks (jointly) provide the
necessary details to implement a variety of congestion control
algorithms, but the (compound) report size grows quickly.
7.3. Congestion Control Interoperability and Legacy Systems
There are legacy RTP implementations that do not implement RTCP, and
hence do not provide any congestion feedback. Congestion control
cannot be performed with these end-points. WebRTC implementations
that need to interwork with such end-points MUST limit their
transmission to a low rate, equivalent to a VoIP call using a low
bandwidth codec, that is unlikely to cause any significant
congestion.
When interworking with legacy implementations that support RTCP using
the RTP/AVP profile [RFC3551], congestion feedback is provided in
Perkins, et al. Expires October 25, 2014 [Page 22]
Internet-Draft RTP for WebRTC April 2014
RTCP RR packets every few seconds. Implementations that have to
interwork with such end-points MUST ensure that they keep within the
RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers]
constraints to limit the congestion they can cause.
If a legacy end-point supports RTP/AVPF, this enables negotiation of
important parameters for frequent reporting, such as the "trr-int"
parameter, and the possibility that the end-point supports some
useful feedback format for congestion control purpose such as TMMBR
[RFC5104]. Implementations that have to interwork with such end-
points MUST ensure that they stay within the RTP circuit breaker
[I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the
congestion they can cause, but might find that they can achieve
better congestion response depending on the amount of feedback that
is available.
With proprietary congestion control algorithms issues can arise when
different algorithms and implementations interact in a communication
session. If the different implementations have made different
choices in regards to the type of adaptation, for example one sender
based, and one receiver based, then one could end up in situation
where one direction is dual controlled, when the other direction is
not controlled. This memo cannot mandate behaviour for proprietary
congestion control algorithms, but implementations that use such
algorithms ought to be aware of this issue, and try to ensure that
both effective congestion control is negotiated for media flowing in
both directions. If the IETF were to standardise both sender- and
receiver-based congestion control algorithms for WebRTC traffic in
the future, the issues of interoperability, control, and ensuring
that both directions of media flow are congestion controlled would
also need to be considered.
8. WebRTC Use of RTP: Performance Monitoring
As described in Section 4.1, implementations are REQUIRED to generate
RTCP Sender Report (SR) and Reception Report (RR) packets relating to
the RTP packet streams they send and receive. These RTCP reports can
be used for performance monitoring purposes, since they include basic
packet loss and jitter statistics.
A large number of additional performance metrics are supported by the
RTCP Extended Reports (XR) framework [RFC3611][RFC6792]. At the time
of this writing, it is not clear what extended metrics are suitable
for use in the WebRTC context, so there is no requirement that
implementations generate RTCP XR packets. However, implementations
that can use detailed performance monitoring data MAY generate RTCP
XR packets as appropriate; the use of such packets SHOULD be
signalled in advance.
Perkins, et al. Expires October 25, 2014 [Page 23]
Internet-Draft RTP for WebRTC April 2014
All WebRTC implementations MUST be prepared to receive RTP XR report
packets, whether or not they were signalled. There is no requirement
that the data contained in such reports be used, or exposed to the
Javascript application, however.
9. WebRTC Use of RTP: Future Extensions
It is possible that the core set of RTP protocols and RTP extensions
specified in this memo will prove insufficient for the future needs
of WebRTC applications. In this case, future updates to this memo
MUST be made following the Guidelines for Writers of RTP Payload
Format Specifications [RFC2736], How to Write an RTP Payload Format
[I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP
Control Protocol [RFC5968], and SHOULD take into account any future
guidelines for extending RTP and related protocols that have been
developed.
Authors of future extensions are urged to consider the wide range of
environments in which RTP is used when recommending extensions, since
extensions that are applicable in some scenarios can be problematic
in others. Where possible, the WebRTC framework 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. Signalling Considerations
RTP is built with the assumption that an external signalling channel
exists, and can be used to configure RTP sessions and their features.
The basic configuration of an RTP session consists of the following
parameters:
RTP Profile: The name of the RTP profile to be used in session. The
RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
on basic level, as can their secure variants RTP/SAVP [RFC3711]
and RTP/SAVPF [RFC5124]. The secure variants of the profiles do
not directly interoperate with the non-secure variants, due to the
presence of additional header fields for authentication in SRTP
packets and cryptographic transformation of the payload. WebRTC
requires the use of the RTP/SAVPF profile, and this MUST be
signalled if SDP is used. Interworking functions might transform
this into the RTP/SAVP profile for a legacy use case, by
indicating to the WebRTC end-point that the RTP/SAVPF is used, and
limiting the usage of the "a=rtcp-fb:" attribute to indicate a
trr-int value of 4 seconds.
Transport Information: Source and destination IP address(s) and
ports for RTP and RTCP MUST be signalled for each RTP session. In
Perkins, et al. Expires October 25, 2014 [Page 24]
Internet-Draft RTP for WebRTC April 2014
WebRTC these transport addresses will be provided by ICE 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 signalled (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 signalled. Each
media type MAY also have a number of media type parameters that
MUST also be signalled to configure the codec and RTP payload
format (the "a=fmtp:" line from SDP). Section 4.3 of this memo
discusses requirements for uniqueness of payload types.
RTP Extensions: The RTP extensions to be used SHOULD be agreed upon,
including any parameters for each respective extension. At the
very least, this will help avoiding using bandwidth for features
that the other end-point will ignore. But for certain mechanisms
there is requirement for this to happen as interoperability
failure otherwise happens.
RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the
end-points will be necessary. This SHALL be done as described in
"Session Description Protocol (SDP) Bandwidth Modifiers for RTP
Control Protocol (RTCP) Bandwidth" [RFC3556], or something
semantically equivalent. This also ensures that the end-points
have a common view of the RTCP bandwidth, this is important as too
different view of the bandwidths can lead to failure to
interoperate.
These parameters are often expressed in SDP messages conveyed within
an offer/answer exchange. RTP does not depend on SDP or on the offer
/answer model, but does require all the necessary parameters to be
agreed upon, and provided to the RTP implementation. We note that in
the WebRTC context it will depend on the signalling model and API how
these parameters need to be configured but they will be need to
either set in the API or explicitly signalled between the peers.
11. WebRTC API Considerations
The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and
Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses
the concept of a MediaStream that consists of zero or more
MediaStreamTracks. A MediaStreamTrack is an individual stream of
media from any type of media source like a microphone or a camera,
but also conceptual sources, like a audio mix or a video composition,
are possible. The MediaStreamTracks within a MediaStream need to be
possible to play out synchronised.
Perkins, et al. Expires October 25, 2014 [Page 25]
Internet-Draft RTP for WebRTC April 2014
A MediaStreamTrack's realisation in RTP in the context of an
RTCPeerConnection consists of a source packet stream identified with
an SSRC within an RTP session part of the RTCPeerConnection. The
MediaStreamTrack can also result in additional packet streams, and
thus SSRCs, in the same RTP session. These can be dependent packet
streams from scalable encoding of the source stream associated with
the MediaStreamTrack, if such a media encoder is used. They can also
be redundancy packet streams, these are created when applying 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 a RTCPeerConnection SHALL result
in an independent source packet stream, with its own set of
associated packet streams, and thus different SSRC(s). It will
depend on applied constraints and parameters if the source stream and
the encoding configuration will be identical between different
MediaStreamTracks sharing the same media source. Thus it is possible
for multiple source packet streams to share encoded streams (but not
packet streams), but this is an implementation choice to try to
utilise such optimisations. Note that such optimizations would need
to take into account that the constraints for one of the
MediaStreamTracks can at any moment change, meaning that the encoding
configurations might no longer be identical.
The same MediaStreamTrack can also be included in multiple
MediaStreams, thus multiple sets of MediaStreams can implicitly need
to use the same synchronisation base. To ensure that this works in
all cases, and don't forces a end-point to change synchronisation
base and CNAME in the middle of a ongoing delivery of any packet
streams, which would cause media disruption; all MediaStreamTracks
and their associated SSRCs originating from the same end-point needs
to be sent using the same CNAME within one RTCPeerConnection. This
is motivating the strong recommendation in Section 4.9 to only use a
single CNAME.
The requirement on using the same CNAME for all SSRCs that
originates from the same end-point, does not require middleboxes
that forwards traffic from multiple end-points 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
[I-D.ietf-rtcweb-security] for details). A web application can
Perkins, et al. Expires October 25, 2014 [Page 26]
Internet-Draft RTP for WebRTC April 2014
request that the CNAMEs used in different RTCPeerConnection within a
same-orign context to be the same, this allow 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.
The above will currently force a WebRTC end-point that receives an
MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing
on any RTCPeerConnection to perform resynchronisation of the stream.
This, as the sending party needs to change the CNAME, which implies
that it has to use a locally available system clock as timebase for
the synchronisation. Thus, the relative relation between the
timebase of the incoming stream and the system sending out needs to
defined. This relation also needs monitoring for clock drift and
likely adjustments of the synchronisation. The sending entity is
also responsible for congestion control for its the sent streams. In
cases of packet loss the loss of incoming data also needs to be
handled. This leads to the observation that the method that is least
likely to cause issues or interruptions in the outgoing source packet
stream is a model of full decoding, including repair etc followed by
encoding of the media again into the outgoing packet stream.
Optimisations of this method is clearly possible and implementation
specific.
A WebRTC end-point MUST support receiving multiple MediaStreamTracks,
where each of different MediaStreamTracks (and their sets of
associated packet streams) uses different CNAMEs. However,
MediaStreamTracks that are received with different CNAMEs have no
defined synchronisation.
Note: The motivation for supporting reception of multiple CNAMEs
are to allow for forward compatibility with any future changes
that enables more efficient stream handling when end-points relay/
forward streams. It also ensures that end-points can interoperate
with certain types of multi-stream middleboxes or end-points that
are not WebRTC.
The binding between the WebRTC MediaStreams, MediaStreamTracks and
the SSRC is done as specified in "Cross Session Stream Identification
in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This
document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to
map unknown source packet stream SSRCs to MediaStreamTracks and
MediaStreams. Commonly the RTP Payload Type of any incoming packets
will reveal if the packet stream is a source stream or a redundancy
or dependent packet stream. The association to the correct source
Perkins, et al. Expires October 25, 2014 [Page 27]
Internet-Draft RTP for WebRTC April 2014
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)
and 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
end-point implementation perspective, and while some mention is made
of the behaviour of middleboxes, that is not the focus of this memo.
12.1. Configuration and Use of RTP Sessions
A WebRTC end-point will be a simultaneous participant in one or more
RTP sessions. Each RTP session can convey multiple media sources,
and can include media data from multiple end-points. In the
following, we outline some ways in which WebRTC end-points can
configure and use RTP sessions.
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 sources (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 bi-
directional transport-layer flow, but will contain multiple RTP
packet streams, each containing a different type of media. A
common example might be an end-point 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 end-point might have multiple
cameras, microphones, or other media capture devices, and so might
want to generate several RTP packet streams of the same media
type. Alternatively, it might want to send media from a single
Perkins, et al. Expires October 25, 2014 [Page 28]
Internet-Draft RTP for WebRTC April 2014
capture device in several different formats or quality settings at
once. Both can result in a single end-point sending multiple RTP
packet streams of the same media type into a single RTP session at
the same time.
Associated Repair Data: An end-point might send a RTP packet stream
that is somehow associated with another stream. For example, it
might send an RTP packet stream that contains FEC or
retransmission data relating to another stream. Some RTP payload
formats send this sort of associated repair data as part of the
source packet stream, while others send it as a separate packet
stream.
Layered or Multiple Description Coding: An end-point can use a
layered media codec, for example H.264 SVC, or a multiple
description codec, that generates multiple RTP packet streams,
each with a distinct RTP SSRC, within a single RTP session.
RTP Mixers, Translators, and Other Middleboxes: An RTP session, in
the WebRTC context, is a point-to-point association between an
end-point and some other peer device, where those devices share a
common SSRC space. The peer device might be another WebRTC end-
point, or it might be an RTP mixer, translator, or some other form
of media processing middlebox. In the latter cases, the middlebox
might send mixed or relayed RTP streams from several participants,
that the WebRTC end-point will need to render. Thus, even though
a WebRTC end-point might only be a member of a single RTP session,
the peer device might be extending that RTP session to incorporate
other end-points. WebRTC is a group communication environment and
end-points need to be capable of receiving, decoding, and playing
out multiple RTP 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 end-point might participate in
multiple RTP sessions. There are several reasons why a WebRTC end-
point 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 end-points need to support the option of sending different
types of media on different RTP sessions, so they can interwork
with such legacy devices. This is discussed further in
Section 4.4.
Perkins, et al. Expires October 25, 2014 [Page 29]
Internet-Draft RTP for WebRTC April 2014
To provide enhanced quality of service: Some network-based quality
of service mechanisms operate on the granularity of transport
layer flows. If it is desired to use these mechanisms to provide
differentiated quality of service for some RTP packet streams,
then those RTP packet streams need to be sent in a separate RTP
session using a different transport-layer flow, and with
appropriate quality of service marking. This is discussed further
in Section 12.1.3.
To separate media with different purposes: An end-point might want
to send RTP packet streams that have different purposes on
different RTP sessions, to make it easy for the peer device to
distinguish them. For example, some centralised multiparty
conferencing systems display the active speaker in high
resolution, but show low resolution "thumbnails" of other
participants. Such systems might configure the end-points 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 to
accomplished by establishing multiple WebRTC MediaStreamTracks
that have the same media source in one (or more)
RTCPeerConnection. Each MediaStreamTrack is then configured to
deliver a particular media quality and thus media bit-rate, and
will produce an independently encoded version with the codec
parameters agreed specifically in the context of that
RTCPeerConnection. The RTP middlebox can distinguish packets
corresponding to the low- and high-resolution streams by
inspecting their SSRC, RTP payload type, or some other information
contained in RTP payload, RTP header extension or RTCP packets,
but it can be easier to distinguish the RTP packet streams if they
arrive on separate RTP sessions on separate transport-layer flows.
To directly connect with multiple peers: A multi-party conference
does not need to use an RTP middlebox. Rather, a multi-unicast
mesh can be created, comprising several distinct RTP sessions,
with each participant sending RTP traffic over a separate RTP
session (that is, using an independent RTCPeerConnection object)
to every other participant, as shown in 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 are
part of the same session beyond the sender itself.
Perkins, et al. Expires October 25, 2014 [Page 30]
Internet-Draft RTP for WebRTC April 2014
+---+ +---+
| 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 between the
relation between RTP sessions and RTCPeerConnection objects we
recommend that this is implemented as several individual RTP
sessions. The only downside is that end-point A will not learn of
the quality of any transmission happening between B and C, since
it will not see RTCP reports for the RTP session between B and C,
whereas it would it all three participants were part of a single
RTP session. Experience with the Mbone tools (experimental RTP-
based multicast conferencing tools from the late 1990s) has showed
that RTCP reception quality reports for third parties can usefully
be presented to the users in a way that helps them understand
asymmetric network problems, and the approach of using separate
RTP sessions prevents this. However, an advantage of using
separate RTP sessions is that it enables using different media
bit-rates and RTP session configurations between the different
peers, thus not forcing B to endure the same quality reductions if
there are limitations in the transport from A to C as C will. It
it believed that these advantages outweigh the limitations in
debugging power.
To indirectly connect with multiple peers: A common scenario in
multi-party conferencing is to create indirect connections to
multiple peers, using an RTP mixer, translator, or some other type
of RTP middlebox. Figure 2 outlines a simple topology that might
be used in a four-person centralised conference. The middlebox
acts to optimise the transmission of RTP packet streams from
certain perspectives, either by only sending some of the received
RTP packet stream to any given receiver, or by providing a
combined RTP packet stream out of a set of contributing streams.
Perkins, et al. Expires October 25, 2014 [Page 31]
Internet-Draft RTP for WebRTC April 2014
+---+ +-------------+ +---+
| 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
end-points in one multi-party session. Other types of middlebox
might use separate RTP sessions between each end-point and the
middlebox. A common aspect is that these RTP middleboxes can use
a number of tools to control the media encoding provided by a
WebRTC end-point. This includes functions like requesting
breaking the encoding chain and have the encoder produce a so
called Intra frame. Another is limiting the bit-rate of a given
stream to better suit the mixer view of the multiple down-streams.
Others are controlling the most suitable frame-rate, picture
resolution, the trade-off between frame-rate and spatial quality.
The middlebox gets the significant responsibility to correctly
perform congestion control, source identification, manage
synchronisation while providing the application with suitable
media optimizations. The middlebox is also has to be a trusted
node when it comes to security, since it manipulates either the
RTP header or the media itself (or both) received from one end-
point, before sending it on towards the end-point(s), thus they
need to be able to decrypt and then encrypt it before sending it
out.
RTP Mixers can create a situation where an end-point experiences a
situation in-between a session with only two end-points and
multiple RTP sessions. Mixers are expected to not forward RTCP
reports regarding RTP packet streams across themselves. This is
due to the difference in the RTP packet streams provided to the
different end-points. The original media source lacks information
about a mixer's manipulations prior to sending it the different
receivers. This scenario also results in that an end-point's
feedback or requests goes to the mixer. When the mixer can't act
on this by itself, it is forced to go to the original media source
to fulfil the receivers request. This will not necessarily be
explicitly visible any RTP and RTCP traffic, but the interactions
and the time to complete them will indicate such dependencies.
Perkins, et al. Expires October 25, 2014 [Page 32]
Internet-Draft RTP for WebRTC April 2014
Providing source authentication in multi-party scenarios is a
challenge. In the mixer-based topologies, end-points source
authentication is based on, firstly, verifying that media comes
from the mixer by cryptographic verification and, secondly, trust
in the mixer to correctly identify any source towards the end-
point. In RTP sessions where multiple end-points are directly
visible to an end-point, all end-points will have knowledge about
each others' master keys, and can thus inject packets claimed to
come from another end-point in the session. Any node performing
relay can perform non-cryptographic mitigation by preventing
forwarding of packets that have SSRC fields that came from other
end-points before. For cryptographic verification of the source
SRTP would require additional security mechanisms, for example
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 end-point that receives an RTP packet stream to be able to
forward that RTP packet stream to a third party. The are some
obvious security and privacy implications in supporting this, but
also potential uses. This is supported in the W3C API by taking
the received and decoded media and using it as media source that
is re-encoding and transmitted as a new stream.
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 end-point that is performing the forwarding is responsible for
producing an RTP packet stream suitable for onwards transmission.
The outgoing RTP session that is used to send the forwarded media
is entirely separate to the RTP session on which the media was
received. This will require media transcoding for congestion
control purpose to produce a suitable bit-rate for the outgoing
RTP session, reducing media quality and forcing the forwarding
end-point to spend the resource on the transcoding. The media
transcoding does result in a separation of the two different legs
removing almost all dependencies, and allowing the forwarding end-
point to optimize its media transcoding operation. The cost is
greatly increased computational complexity on the forwarding node.
Receivers of the forwarded stream will see the forwarding device
as the sender of the stream, and will not be able to tell from the
Perkins, et al. Expires October 25, 2014 [Page 33]
Internet-Draft RTP for WebRTC April 2014
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 Packet 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 end-point
sending the media, which controls, both which RTP packet streams that
will be sent, and their allocation of bit-rate out of the current
available aggregate as determined by the congestion control.
It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will
allow the application to indicate relative priorities for different
MediaStreamTracks. These priorities can then be used to influence
the local RTP processing, especially when it comes to congestion
control response in how to divide the available bandwidth between the
RTP packet streams. Any changes in relative priority will also need
to be considered for RTP packet streams that are associated with the
main RTP packet streams, such as redundant streams for RTP
retransmission and FEC. The importance of such redundant RTP packet
streams is dependent on the media type and codec used, in regards to
how robust that codec is to packet loss. However, a default policy
might to be to use the same priority for redundant RTP packet stream
as for the source RTP packet stream.
Secondly, the network can prioritize transport-layer flows and sub-
flows, including RTP packet streams. Typically, differential
treatment includes two steps, the first being identifying whether an
IP packet belongs to a class that has to be treated differently, the
second the actual mechanism to prioritize packets. This is done
according to three methods:
DiffServ: The end-point marks a packet with a DiffServ code point to
indicate to the network that the packet belongs to a particular
class.
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
Perkins, et al. Expires October 25, 2014 [Page 34]
Internet-Draft RTP for WebRTC April 2014
be possible to provide differential treatment compared to best-effort
for all the RTP packet streams in a WebRTC application. When flow-
based differentiation is available the WebRTC application needs to
know about it so that it can provide the separation of the RTP packet
streams onto different UDP flows to enable a more granular usage of
flow based differentiation. That way at least providing different
prioritization of audio and video if desired by application.
DiffServ assumes that either the end-point or a classifier can mark
the packets with an appropriate DSCP so that the packets are treated
according to that marking. If the end-point is to mark the traffic
two requirements arise in the WebRTC context: 1) The WebRTC
application or browser has to know which DSCP to use and that it can
use them on some set of RTP 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 "DSCP and other packet markings for
RTCWeb QoS" [I-D.ietf-tsvwg-rtcweb-qos].
For packet based marking schemes it might be possible to mark
individual RTP packets differently based on the relative priority of
the RTP payload. For example video codecs that have I, P, and B
pictures could prioritise any payloads carrying only B frames less,
as these are less damaging to loose. However, depending on the QoS
mechanism and what markings that are applied, this can result in not
only different packet drop probabilities but also packet reordering,
see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As default
policy all RTP packets related to a RTP packet stream ought to be
provided with the same prioritization; per-packet prioritization is
outside the scope of this memo, but might be specified elsewhere in
future.
It is also important to consider how RTCP packets associated with a
particular RTP packet stream need to be marked. RTCP compound
packets with Sender Reports (SR), ought to be marked with the same
priority as the RTP packet stream itself, so the RTCP-based round-
trip time (RTT) measurements are done using the same transport-layer
flow priority as the RTP packet stream experiences. RTCP compound
packets containing RR packet ought to be sent with the priority used
by the majority of the RTP packet streams reported on. RTCP packets
containing time-critical feedback packets can use higher priority to
improve the timeliness and likelihood of delivery of such feedback.
12.2. Media Source, RTP Packet Streams, and Participant Identification
Perkins, et al. Expires October 25, 2014 [Page 35]
Internet-Draft RTP for WebRTC April 2014
12.2.1. Media Source
Each RTP packet stream is identified by a unique synchronisation
source (SSRC) identifier. The SSRC identifier is carried in each of
the RTP packets comprising a RTP packet stream, and is also used to
identify that stream in the corresponding RTCP reports. The SSRC is
chosen as discussed in Section 4.8. The first stage in
demultiplexing RTP and RTCP packets received on a single transport
layer flow at a WebRTC end-point 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 needs to be exposed to the WebRTC
application using some API, if the application is to be able to track
changes in session participation. It is desirable to map CSRC values
back into WebRTC MediaStream identities as they cross this API, to
avoid exposing the SSRC/CSRC name space to JavaScript applications.
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.
This information can usefully be exposed in the user interface.
12.2.2. SSRC Collision Detection
The RTP standard [RFC3550] requires any RTP implementation to have
support for detecting and handling SSRC collisions, i.e., resolve the
conflict when two different end-points use the same SSRC value. This
requirement also applies to WebRTC end-points. There are several
scenarios where SSRC collisions can occur:
o In a point-to-point session where each SSRC is associated with
either of the two end-points and where the main media carrying
SSRC identifier will be announced in the signalling channel, a
collision is less likely to occur due to the information about
used SSRCs provided by Source-Specific SDP Attributes [RFC5576].
Perkins, et al. Expires October 25, 2014 [Page 36]
Internet-Draft RTP for WebRTC April 2014
Still, collisions can occur if both end-points start uses an new
SSRC identifier prior to having signalled it to the peer and
received acknowledgement on the signalling message. The Source-
Specific SDP Attributes [RFC5576] contains no mechanism to resolve
SSRC collisions or reject a end-points usage of an SSRC.
o SSRC values that have not been signalled could also appear in an
RTP session. This is more likely than it appears, since some RTP
functions use extra SSRCs to provide their functionality. For
example, retransmission data might be transmitted using a separate
RTP packet stream that requires its own SSRC, separate to the SSRC
of the source RTP packet stream [RFC4588]. In those cases, an
end-point can create a new SSRC that strictly doesn't need to be
announced over the signalling channel to function correctly on
both RTP and RTCPeerConnection level.
o Multiple end-points 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 end-
points from the RTP middlebox collisions can occur.
o An RTP middlebox could connect an end-point's RTCPeerConnection to
another RTCPeerConnection from the same end-point, thus forming a
loop where the end-point will receive its own traffic. While is
is clearly considered a bug, it is important that the end-point is
able to recognise and handle the case when it occurs. This case
becomes even more problematic when media mixers, and so on, are
involved, where the stream received is a different stream but
still contains this client's input.
These SSRC/CSRC collisions can only be handled on RTP level as long
as the same RTP session is extended across multiple
RTCPeerConnections by a RTP middlebox. To resolve the more generic
case where multiple RTCPeerConnections are interconnected, then
identification of the media source(s) part of a MediaStreamTrack
being propagated across multiple interconnected RTCPeerConnection
needs to be preserved across these interconnections.
12.2.3. Media Synchronisation Context
When an end-point sends media from more than one media source, it
needs to consider if (and which of) these media sources are to be
synchronized. In RTP/RTCP, synchronisation is provided by having a
set of RTP packet streams be indicated as coming from the same
synchronisation context and logical end-point by using the same RTCP
CNAME identifier.
Perkins, et al. Expires October 25, 2014 [Page 37]
Internet-Draft RTP for WebRTC April 2014
The next provision is that the internal clocks of all media sources,
i.e., what drives the RTP timestamp, can be correlated to a system
clock that is provided in RTCP Sender Reports encoded in an NTP
format. By correlating all RTP timestamps to a common system clock
for all sources, the timing relation of the different RTP packet
streams, also across multiple RTP sessions can be derived at the
receiver and, if desired, the streams can be synchronized. The
requirement is for the media sender to provide the correlation
information; it is up to the receiver to use it or not.
13. Security Considerations
The overall security architecture for WebRTC is described in
[I-D.ietf-rtcweb-security-arch], and security considerations for the
WebRTC framework are described in [I-D.ietf-rtcweb-security]. 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.
We do not believe there are any new security considerations resulting
from the combination of these various protocol extensions.
The Extended Secure RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
handling of fundamental issues by offering confidentiality, integrity
and partial source authentication. A mandatory to implement media
security solution is created by combing this secured RTP profile and
DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of
[I-D.ietf-rtcweb-security-arch].
RTCP packets convey a Canonical Name (CNAME) identifier that is used
to associate RTP packet streams that need to be synchronised across
related RTP sessions. Inappropriate choice of CNAME values can be a
privacy concern, since long-term persistent CNAME identifiers can be
used to track users across multiple WebRTC calls. Section 4.9 of
this memo provides guidelines for generation of untraceable CNAME
values that alleviate this risk.
The guidelines in [RFC6562] apply when using variable bit rate (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 or third party monitoring that will greatly benefit from
the information, and this has been expressed using API or signalling.
Perkins, et al. Expires October 25, 2014 [Page 38]
Internet-Draft RTP for WebRTC April 2014
If further evidence are produced to show that information leakage is
significant from audio level indications, then use of encryption
needs to be mandated at that time.
14. IANA Considerations
This memo makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as
an RFC.
15. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand,
Cary Bran, Charles Eckel, Christian Groves, Cullen Jennings, Dan
Romascanu, Martin Thomson, and the other members of the IETF RTCWEB
working group for their valuable feedback.
16. References
16.1. Normative References
[I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Sending
Multiple Types of Media in a Single RTP Session", draft-
ietf-avtcore-multi-media-rtp-session-05 (work in
progress), February 2014.
[I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", draft-ietf-
avtcore-rtp-circuit-breakers-05 (work in progress),
February 2014.
[I-D.ietf-avtcore-rtp-multi-stream-optimisation]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session:
Grouping RTCP Reception Statistics and Other Feedback",
draft-ietf-avtcore-rtp-multi-stream-optimisation-02 (work
in progress), February 2014.
[I-D.ietf-avtcore-rtp-multi-stream]
Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
"Sending Multiple Media Streams in a Single RTP Session",
draft-ietf-avtcore-rtp-multi-stream-03 (work in progress),
February 2014.
Perkins, et al. Expires October 25, 2014 [Page 39]
Internet-Draft RTP for WebRTC April 2014
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-09 (work in progress), February 2014.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-06 (work in progress), January 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736, December
1999.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
3556, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[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, July
2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, July 2007.
Perkins, et al. Expires October 25, 2014 [Page 40]
Internet-Draft RTP for WebRTC April 2014
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
[RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time
Transport Protocol (RTP) Header Extension for Client-to-
Mixer Audio Level Indication", RFC 6464, December 2011.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 2011.
[RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP", RFC 6562, March
2012.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904, April
2013.
[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,
August 2013.
Perkins, et al. Expires October 25, 2014 [Page 41]
Internet-Draft RTP for WebRTC April 2014
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, September 2013.
[RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session", RFC 7160, April 2014.
[RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC
7164, March 2014.
[W3C.WD-mediacapture-streams-20130903]
Burnett, D., Bergkvist, A., Jennings, C., and A.
Narayanan, "Media Capture and Streams", World Wide Web
Consortium WD WD-mediacapture-streams-20130903, September
2013, <http://www.w3.org/TR/2013/
WD-mediacapture-streams-20130903>.
[W3C.WD-webrtc-20130910]
Bergkvist, A., Burnett, D., Jennings, C., and A.
Narayanan, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-
webrtc-20130910, September 2013,
<http://www.w3.org/TR/2013/WD-webrtc-20130910>.
16.2. Informative References
[I-D.ietf-avtcore-multiplex-guidelines]
Westerlund, M., Perkins, C., and H. Alvestrand,
"Guidelines for using the Multiplexing Features of RTP to
Support Multiple Media Streams", draft-ietf-avtcore-
multiplex-guidelines-02 (work in progress), January 2014.
[I-D.ietf-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies", draft-
ietf-avtcore-rtp-topologies-update-01 (work in progress),
October 2013.
[I-D.ietf-avtext-rtp-grouping-taxonomy]
Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,
"A Taxonomy of Grouping Semantics and Mechanisms for Real-
Time Transport Protocol (RTP) Sources", draft-ietf-avtext-
rtp-grouping-taxonomy-01 (work in progress), February
2014.
[I-D.ietf-mmusic-msid]
Alvestrand, H., "WebRTC MediaStream Identification in the
Session Description Protocol", draft-ietf-mmusic-msid-05
(work in progress), March 2014.
Perkins, et al. Expires October 25, 2014 [Page 42]
Internet-Draft RTP for WebRTC April 2014
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-07 (work in progress), April 2014.
[I-D.ietf-payload-rtp-howto]
Westerlund, M., "How to Write an RTP Payload Format",
draft-ietf-payload-rtp-howto-13 (work in progress),
January 2014.
[I-D.ietf-rmcat-cc-requirements]
Jesup, R., "Congestion Control Requirements For RMCAT",
draft-ietf-rmcat-cc-requirements-04 (work in progress),
April 2014.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-05 (work in
progress), February 2014.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-09 (work
in progress), February 2014.
[I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-14 (work in
progress), February 2014.
[I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS", draft-ietf-tsvwg-
rtcweb-qos-00 (work in progress), April 2014.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611, November
2003.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006.
Perkins, et al. Expires October 25, 2014 [Page 43]
Internet-Draft RTP for WebRTC April 2014
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
March 2006.
[RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient
Stream Loss-Tolerant Authentication (TESLA) in the Secure
Real-time Transport Protocol (SRTP)", RFC 4383, February
2006.
[RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828, April
2007.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC
5348, September 2008.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.
[RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP
Control Protocol (RTCP)", RFC 5968, September 2010.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, June 2011.
[RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the
RTP Monitoring Framework", RFC 6792, November 2012.
Authors' Addresses
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
URI: http://csperkins.org/
Perkins, et al. Expires October 25, 2014 [Page 44]
Internet-Draft RTP for WebRTC April 2014
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Joerg Ott
Aalto University
School of Electrical Engineering
Espoo 02150
Finland
Email: jorg.ott@aalto.fi
Perkins, et al. Expires October 25, 2014 [Page 45]
