text:
draft-ietf-rtcweb-rtp-usage-05
Internet-Draft
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
This is an older version of an Internet-Draft that was ultimately published as RFC 8834.
RFC 8834 draft-ietf-rtcweb-rtp-usage-26 draft-ietf-rtcweb-rtp-usage-25 draft-ietf-rtcweb-rtp-usage-24 draft-ietf-rtcweb-rtp-usage-23 draft-ietf-rtcweb-rtp-usage-22 draft-ietf-rtcweb-rtp-usage-21 draft-ietf-rtcweb-rtp-usage-20 draft-ietf-rtcweb-rtp-usage-19 draft-ietf-rtcweb-rtp-usage-18 draft-ietf-rtcweb-rtp-usage-17 draft-ietf-rtcweb-rtp-usage-16 draft-ietf-rtcweb-rtp-usage-15 draft-ietf-rtcweb-rtp-usage-14 draft-ietf-rtcweb-rtp-usage-13 draft-ietf-rtcweb-rtp-usage-12 draft-ietf-rtcweb-rtp-usage-11 draft-ietf-rtcweb-rtp-usage-10 draft-ietf-rtcweb-rtp-usage-09 draft-ietf-rtcweb-rtp-usage-08 draft-ietf-rtcweb-rtp-usage-07 draft-ietf-rtcweb-rtp-usage-06 draft-ietf-rtcweb-rtp-usage-05 draft-ietf-rtcweb-rtp-usage-04 draft-ietf-rtcweb-rtp-usage-03 draft-ietf-rtcweb-rtp-usage-02 draft-ietf-rtcweb-rtp-usage-01 draft-ietf-rtcweb-rtp-usage-00 draft-perkins-rtcweb-rtp-usage-03 draft-perkins-rtcweb-rtp-usage-02 draft-perkins-rtcweb-rtp-usage-01 draft-perkins-rtcweb-rtp-usage-00 RFC 8834 draft-ietf-rtcweb-rtp-usage-26 draft-ietf-rtcweb-rtp-usage-25 draft-ietf-rtcweb-rtp-usage-24 draft-ietf-rtcweb-rtp-usage-23 draft-ietf-rtcweb-rtp-usage-22 draft-ietf-rtcweb-rtp-usage-21 draft-ietf-rtcweb-rtp-usage-20 draft-ietf-rtcweb-rtp-usage-19 draft-ietf-rtcweb-rtp-usage-18 draft-ietf-rtcweb-rtp-usage-17 draft-ietf-rtcweb-rtp-usage-16 draft-ietf-rtcweb-rtp-usage-15 draft-ietf-rtcweb-rtp-usage-14 draft-ietf-rtcweb-rtp-usage-13 draft-ietf-rtcweb-rtp-usage-12 draft-ietf-rtcweb-rtp-usage-11 draft-ietf-rtcweb-rtp-usage-10 draft-ietf-rtcweb-rtp-usage-09 draft-ietf-rtcweb-rtp-usage-08 draft-ietf-rtcweb-rtp-usage-07 draft-ietf-rtcweb-rtp-usage-06 draft-ietf-rtcweb-rtp-usage-05 draft-ietf-rtcweb-rtp-usage-04 draft-ietf-rtcweb-rtp-usage-03 draft-ietf-rtcweb-rtp-usage-02 draft-ietf-rtcweb-rtp-usage-01 draft-ietf-rtcweb-rtp-usage-00 draft-perkins-rtcweb-rtp-usage-03 draft-perkins-rtcweb-rtp-usage-02 draft-perkins-rtcweb-rtp-usage-01 draft-perkins-rtcweb-rtp-usage-00
Colin Perkins , Magnus Westerlund , Joerg Ott
draft-perkins-rtcweb-rtp-usage
Mailing list discussion
Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Standards Track                           M. Westerlund
Expires: April 25, 2013                                         Ericsson
J. Ott
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.
This Internet-Draft is submitted in full conformance with the
This Internet-Draft will expire on April 25, 2013.
Perkins, et al.          Expires April 25, 2013                 [Page 1]
Internet-Draft               RTP for WebRTC                 October 2012
2.  Rationale  . . . . . . . . . . . . . . . . . . . . . . . . . .  4
4.  WebRTC Use of RTP: Core Protocols  . . . . . . . . . . . . . .  6
4.1.  RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . .  6
4.2.  Choice of the RTP Profile  . . . . . . . . . . . . . . . .  7
4.3.  Choice of RTP Payload Formats  . . . . . . . . . . . . . .  8
4.4.  RTP Session Multiplexing . . . . . . . . . . . . . . . . .  8
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)  . . . . . . . 10
4.9.  Generation of the RTCP Canonical Name (CNAME)  . . . . . . 11
5.  WebRTC Use of RTP: Extensions  . . . . . . . . . . . . . . . . 11
5.1.  Conferencing Extensions  . . . . . . . . . . . . . . . . . 11
5.1.1.  Full Intra Request (FIR) . . . . . . . . . . . . . . . 12
5.1.2.  Picture Loss Indication (PLI)  . . . . . . . . . . . . 12
5.1.3.  Slice Loss Indication (SLI)  . . . . . . . . . . . . . 13
5.1.4.  Reference Picture Selection Indication (RPSI)  . . . . 13
5.1.5.  Temporal-Spatial Trade-off Request (TSTR)  . . . . . . 13
5.1.6.  Temporary Maximum Media Stream Bit Rate Request
(TMMBR)  . . . . . . . . . . . . . . . . . . . . . . . 13
5.2.  Header Extensions  . . . . . . . . . . . . . . . . . . . . 14
5.2.1.  Rapid Synchronisation  . . . . . . . . . . . . . . . . 14
5.2.2.  Client-to-Mixer Audio Level  . . . . . . . . . . . . . 14
5.2.3.  Mixer-to-Client Audio Level  . . . . . . . . . . . . . 15
6.  WebRTC Use of RTP: Improving Transport Robustness  . . . . . . 15
6.1.  Negative Acknowledgements and RTP Retransmission . . . . . 15
6.2.  Forward Error Correction (FEC) . . . . . . . . . . . . . . 16
7.  WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 16
7.1.  Boundary Conditions and Circuit Breakers . . . . . . . . . 17
7.2.  RTCP Limitations for Congestion Control  . . . . . . . . . 18
7.3.  Congestion Control Interoperability With Legacy Systems  . 19
8.  WebRTC Use of RTP: Performance Monitoring  . . . . . . . . . . 19
9.  WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 20
10. Signalling Considerations  . . . . . . . . . . . . . . . . . . 20
11. WebRTC API Considerations  . . . . . . . . . . . . . . . . . . 21
11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 21
12. RTP Implementation Considerations  . . . . . . . . . . . . . . 22
12.1. RTP Sessions and PeerConnection  . . . . . . . . . . . . . 22
12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 24
12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 24
12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 25
12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 26
12.6. Media Synchronization  . . . . . . . . . . . . . . . . . . 27
12.7. Multiple RTP End-points  . . . . . . . . . . . . . . . . . 27
12.8. Simulcast  . . . . . . . . . . . . . . . . . . . . . . . . 28
12.9. Differentiated Treatment of Flows  . . . . . . . . . . . . 29
13. Open Issues  . . . . . . . . . . . . . . . . . . . . . . . . . 30
14. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 31
15. Security Considerations  . . . . . . . . . . . . . . . . . . . 31
Appendix A.  Supported RTP Topologies  . . . . . . . . . . . . . . 36
A.1.  Point to Point . . . . . . . . . . . . . . . . . . . . . . 37
A.2.  Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 40
A.3.  Mixer Based  . . . . . . . . . . . . . . . . . . . . . . . 43
A.3.1.  Media Mixing . . . . . . . . . . . . . . . . . . . . . 43
A.3.2.  Media Switching  . . . . . . . . . . . . . . . . . . . 46
A.3.3.  Media Projecting . . . . . . . . . . . . . . . . . . . 49
A.4.  Translator Based . . . . . . . . . . . . . . . . . . . . . 52
A.4.1.  Transcoder . . . . . . . . . . . . . . . . . . . . . . 52
A.4.2.  Gateway / Protocol Translator  . . . . . . . . . . . . 53
A.4.3.  Relay  . . . . . . . . . . . . . . . . . . . . . . . . 55
A.5.  End-point Forwarding . . . . . . . . . . . . . . . . . . . 59
A.6.  Simulcast  . . . . . . . . . . . . . . . . . . . . . . . . 60
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61
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.
The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete
WebRTC framework, of which this memo is a part.
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 requirement 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
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.  This memo concludes with an appendix discussing
several different RTP Topologies, and how they affect the RTP
session(s) and various implementation details of possible realization
of central nodes.
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 for us to
review the available RTP features and extensions, and to define a
common baseline feature set for all WebRTC implementations of RTP.
This builds on the past 15 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.
Other RTP and RTCP extensions not discussed in this document can be
implemented by WebRTC end-points if they are beneficial for new use
cases.  However, they are not necessary to address the WebRTC use
cases and requirements identified to date
[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
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 terms:
RTP Media Stream:  A sequence of RTP packets, and associated RTCP
packets, using a single synchronisation source (SSRC) that
together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session.
RTP Session:  As defined by [RFC3550], the endpoints belonging to the
same RTP Session are those that share a single SSRC space.  That
is, those endpoints can see an SSRC identifier transmitted by any
one of the other endpoints.  An endpoint can see an SSRC either
directly in RTP and RTCP packets, or as a contributing source
(CSRC) in RTP packets from a mixer.  The RTP Session scope is
hence decided by the endpoints' network interconnection topology,
in combination with RTP and RTCP forwarding strategies deployed by
endpoints and any interconnecting middle nodes.
WebRTC MediaStream:  The MediaStream concept defined by the W3C in
the API.
Other terms are used according to their definitions from the RTP
Specification [RFC3550] and WebRTC overview
[I-D.ietf-rtcweb-overview] documents.
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 and
payload formats.  Also described are the core extensions providing
essential features that all WebRTC implementations need to implement
to function effectively on today's networks.
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.
o  Random choice of SSRC on joining a session; collision detection
and resolution for SSRC values (but 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  Support for sending correct synchronization information in the
RTCP Sender Reports, to allow a receiver to implement lip-sync,
with RECOMMENDED support for the rapid RTP synchronisation
extensions (see Section 5.2.1).
o  Support for sending and receiving RTCP SR, RR, SDES, and BYE
packet types, with OPTIONAL support for other RTCP packet types;
implementations MUST ignore unknown RTCP packet types.
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.
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.
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 Real-time Transport Control Protocol (RTCP)-
Based Feedback (RTP/SAVPF)" [RFC5124] as extended by
[I-D.terriberry-avp-codecs] MUST be implemented.  This builds on the
basic RTP/AVP profile [RFC3551], the RTP profile for RTCP-based
feedback (RTP/AVPF) [RFC4585], and the secure RTP profile (RTP/SAVP)
The RTCP-based feedback extensions [RFC4585] are needed for the
improved RTCP timer model, that allows more flexible transmission of
RTCP packets in response to events, rather than strictly according to
bandwidth.  This is vital for being able to report congestion events.
These extensions also save RTCP bandwidth, and will commonly only use
the full RTCP bandwidth allocation if there are many events that
require feedback.  They are also needed to make use of the RTP
conferencing extensions discussed in 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 base RTP/AVP profile, given some constraints on parameter
configuration such as the RTCP bandwidth value and "trr-int" (the
most important factor for interworking with RTP/AVP end-points via
a gateway is to set the trr-int parameter to a value representing
4 seconds).
The secure RTP profile [RFC3711] is needed to provide media
encryption, integrity protection, replay protection and a limited
form of source authentication.  WebRTC implementations MUST NOT send
packets using the basic RTP/AVP profile or the RTP/AVPF profile; they
MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP
packets that are generated.  The default and mandatory to implement
transforms listed in Section 5 of [RFC3711] SHALL apply.
Implementations MUST support DTLS-SRTP [RFC5764] for key-management.
Other key management schemes MAY be supported.
4.3.  Choice of RTP Payload Formats
Implementations MUST follow the WebRTC Audio Codec and Processing
Requirements [I-D.ietf-rtcweb-audio] and SHOULD follow the updated
recommendations for audio codecs in the RTP/AVP Profile
[I-D.terriberry-avp-codecs].  Support for other audio codecs is
(tbd: the mandatory to implement video codec is not yet decided)
Endpoints MAY signal support for multiple RTP payload formats, or
multiple configurations of a single RTP payload format, provided each
payload format uses a different RTP payload type number.  An endpoint
that has signalled support for multiple RTP payload formats SHOULD
accept data in any of those payload formats at any time, unless it
has previously signalled limitations on its decoding capability.
This requirement is constrained if several media types 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
media type 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 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 [I-D.ietf-avtext-multiple-clock-rates].  RTP receivers MUST
follow the recommendations in Section 4.3 of
[I-D.ietf-avtext-multiple-clock-rates], in order to support sources
that switch between clock rates in an RTP session (these
recommendations for receivers are backwards compatible with the case
where senders use only a single clock rate).
4.4.  RTP Session Multiplexing
An association amongst a set of participants communicating with RTP
is known as an RTP session.  A participant can be involved in
multiple RTP sessions at the same time.  In a multimedia session,
each medium has typically been carried in a separate RTP session with
its own RTCP packets (i.e., one RTP session for the audio, with a
separate RTP session using a different transport address for the
video; if SDP is used, this corresponds to one RTP session for each
"m=" line in the SDP).  WebRTC implementations of RTP are REQUIRED to
implement support for multimedia sessions in this way, for
compatibility with legacy systems.
In today's networks, however, with the widespread use of Network
Address/Port Translators (NAT/NAPT) and Firewalls (FW), it is
desirable to reduce the number of transport addresses used by real-
time media applications using RTP by combining multimedia traffic in
a single RTP session.  (Details of how this is to be done are tbd,
but see [I-D.lennox-rtcweb-rtp-media-type-mux],
[I-D.holmberg-mmusic-sdp-bundle-negotiation] and
[I-D.westerlund-avtcore-multiplex-architecture].)  Using a single RTP
session also effects the possibility for differentiated treatment of
media flows.  This is further discussed in Section 12.9.
WebRTC implementations of RTP are REQUIRED to support multiplexing of
a multimedia session onto a single RTP session according to (tbd).
If such RTP session multiplexing is to be used, this MUST be
negotiated during the signalling phase.  Support for multiple RTP
sessions over a single UDP flow as defined by
[I-D.westerlund-avtcore-transport-multiplexing] is RECOMMENDED/
(tbd: No consensus on the level of including support of Multiple RTP
sessions over a single UDP flow.)
4.5.  RTP and RTCP Multiplexing
Historically, RTP and RTCP have been run on separate transport layer
addresses (e.g., two UDP ports for each RTP session, one port for RTP
and one port for RTCP).  With the increased use of Network Address/
Port Translation (NAPT) this has become problematic, since
maintaining multiple NAT bindings can be costly.  It also complicates
firewall administration, since multiple ports need to be opened to
allow RTP traffic.  To reduce these costs and session set-up times,
support for multiplexing RTP data packets and RTCP control packets on
a single port for each RTP session is REQUIRED, as specified in
[RFC5761].  For backwards compatibility, implementations are also
REQUIRED to support sending of RTP and RTCP to separate destination
ports.
Note that the use of RTP and RTCP multiplexed onto a single transport
port ensures that there is occasional traffic sent on that port, even
if there is no active media traffic.  This can be useful to keep NAT
bindings alive, and is the recommend method for application level
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 sending
RTCP feedback packets as [RFC5506] non-compound packets is REQUIRED,
but MUST be negotiated using the signalling channel before use.  For
backwards compatibility, implementations are also REQUIRED to support
the use of compound RTCP feedback packets if the remote endpoint does
not agree to the use of non-compound RTCP in the signalling exchange.
4.7.  Symmetric RTP/RTCP
To ease traversal of NAT and firewall devices, implementations are
REQUIRED to implement and use Symmetric RTP [RFC4961].  This requires
that the IP address and port used for sending and receiving RTP and
RTCP packets are identical.  The reasons for using symmetric RTP is
primarily to avoid issues with NAT and Firewalls by ensuring that the
flow is actually bi-directional and thus kept alive and registered as
flow the intended recipient actually wants.  In addition, it saves
resources, specifically ports at the end-points, but also in the
network as NAT mappings or firewall state is not unnecessary bloated.
Also the amount of QoS state is reduced.
4.8.  Choice of RTP Synchronisation Source (SSRC)
Implementations are REQUIRED to support signalled RTP SSRC values,
using the "a=ssrc:" SDP attribute defined in Sections 4.1 and 5 of
[RFC5576], and MUST also support the "previous-ssrc" source attribute
defined in Section 6.2 of [RFC5576].  Other attributes defined in
[RFC5576] MAY be supported.
Use of the "a=ssrc:" attribute is OPTIONAL.  Implementations MUST
support random SSRC assignment, and MUST support SSRC collision
detection and resolution, both according to [RFC3550].
Perkins, et al.          Expires April 25, 2013                [Page 10]
4.9.  Generation of the RTCP Canonical Name (CNAME)
The RTCP Canonical Name (CNAME) provides a persistent transport-level
identifier for an RTP endpoint.  While the Synchronisation Source
(SSRC) identifier for an RTP endpoint can change if a collision is
detected, or when the RTP application is restarted, its RTCP CNAME is
meant to stay unchanged, so that RTP endpoints can be uniquely
identified and associated with their RTP media streams within a set
of related RTP sessions.  For proper functionality, each RTP endpoint
needs to have a unique RTCP CNAME value.
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.  Accordingly, support for
generating a short-term persistent RTCP CNAMEs following
[I-D.rescorla-avtcore-6222bis] is RECOMMENDED.
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.
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
RTP is inherently a group communication protocol.  Groups can be
implemented using a centralised server, multi-unicast, or using IP
multicast.  While IP multicast was popular in early deployments, in
today's practice, overlay-based conferencing dominates, typically
using one or more central servers to connect endpoints in a star or
flat tree topology.  These central servers can be implemented in a
number of ways as discussed in Appendix A, and in the memo on RTP
Topologies [I-D.westerlund-avtcore-rtp-topologies-update].
As discussed in Section 3.7 of
[I-D.westerlund-avtcore-rtp-topologies-update], the use of a video
switching MCU makes the use of RTCP for congestion control, or any
type of quality reports, very problematic.  Also, as discussed in
section 3.8 of [I-D.westerlund-avtcore-rtp-topologies-update], the
use of a content modifying MCU with RTCP termination breaks RTP loop
detection and removes the ability for receivers to identify active
senders.  RTP Transport Translators (Topo-Translator) are not of
immediate interest to WebRTC, although the main difference compared
to point to point is the possibility of seeing multiple different
transport paths in any RTCP feedback.  Accordingly, only Point to
Point (Topo-Point-to-Point), Multiple concurrent Point to Point
(Mesh) and RTP Mixers (Topo-Mixer) topologies are needed to achieve
the use-cases to be supported in WebRTC initially.  These RECOMMENDED
topologies are expected to be supported by all WebRTC end-points
(these topologies require no special RTP-layer support in the end-
point if the RTP features mandated in this memo are implemented).
The RTP extensions described below to be used with centralised
conferencing -- where one RTP Mixer (e.g., a conference bridge)
receives a participant's RTP media streams and distributes them to
the other participants -- are not necessary for interoperability; an
RTP endpoint that does not implement these extensions will work
correctly, but might offer poor performance.  Support for the listed
extensions will greatly improve the quality of experience and, to
provide a reasonable baseline quality, some these extensions are
mandatory to be supported by WebRTC end-points.
The RTCP conferencing extensions are defined in Extended RTP Profile
for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
5.1.1.  Full Intra Request (FIR)
The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of the
Codec Control Messages [RFC5104].  This message is used to make the
mixer request a new Intra picture from a participant in the session.
This is used when switching between sources to ensure that the
receivers can decode the video or other predictive media encoding
with long prediction chains.  It is REQUIRED that WebRTC senders
understand the react to this feedback message since it greatly
improves the user experience when using centralised mixer-based
conferencing; support for sending the FIR message is OPTIONAL.
5.1.2.  Picture Loss Indication (PLI)
The Picture Loss Indication 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 there could be multiple ways to fulfil the
request.  It is REQUIRED that WebRTC senders understand and react to
this feedback message as a loss tolerance mechanism; receivers MAY
send PLI messages.
5.1.3.  Slice Loss Indication (SLI)
The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF
profile [RFC4585].  It is used by a receiver to tell the encoder that
it has detected the loss or corruption of one or more consecutive
macro blocks, and would like to have these repaired somehow.  The use
of this feedback message is OPTIONAL as a loss tolerance mechanism.
5.1.4.  Reference Picture Selection Indication (RPSI)
Reference Picture Selection Indication (RPSI) is defined in Section
6.3.3 of the RTP/AVPF profile [RFC4585].  Some video coding standards
allow the use of older reference pictures than the most recent one
for predictive coding.  If such a codec is in used, and if the
encoder has learned about a loss of encoder-decoder synchronisation,
a known-as-correct reference picture can be used for future coding.
The RPSI message allows this to be signalled.  Support for RPSI
messages is OPTIONAL.
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)
This feedback message is defined in Sections 3.5.4 and 4.2.1 of the
Codec Control Messages [RFC5104].  This message and its notification
message are used by a media receiver to inform the sending party that
there is a current limitation on the amount of bandwidth available to
this receiver.  This can be various reasons for this: for example, an
RTP mixer can use this message to limit the media rate of the sender
being forwarded by the mixer (without doing media transcoding) to fit
the bottlenecks existing towards the other session participants.  It
is REQUIRED that this feedback message is supported.  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.
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
might include metadata that is additional to the usual RTP
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.  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 a client to inform a mixer about the level of audio
activity in the packet to which the header is attached.  This enables
a central node to make mixing or selection decisions without decoding
or detailed inspection of the payload, reducing the complexity in
some types of central RTP nodes.  It can also save decoding resources
in receivers, which can choose to decode only the most relevant RTP
media streams based on audio activity levels.
The Client-to-Mixer Audio Level [RFC6464] extension is RECOMMENDED to
be implemented.  If it is implemented, it is REQUIRED that the header
extensions are encrypted according to
[I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
contained in these header extensions can be considered sensitive.
5.2.3.  Mixer-to-Client Audio Level
The Mixer to Client Audio Level header extension [RFC6465] provides
the client with the audio level of the different sources mixed into a
common mix by a RTP mixer.  This enables a user interface to indicate
the relative activity level of each session participant, rather than
just being included or not based on the CSRC field.  This is a pure
optimisations of non critical functions, and is hence OPTIONAL to
implement.  If it is implemented, it is REQUIRED that the header
6.  WebRTC Use of RTP: Improving Transport Robustness
There are some tools that can make RTP flows robust against Packet
loss and reduce the impact on media quality.  However, they all add
extra bits compared to a non-robust stream.  These extra bits need to
be considered, and the aggregate bit-rate MUST be rate-controlled.
Thus, 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
will support 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, for example.
Senders are REQUIRED to understand the Generic NACK message defined
in Section 6.2.1 of [RFC4585], but MAY choose to ignore this feedback
(following Section 4.2 of [RFC4585]).  Receivers MAY send NACKs for
missing RTP packets; [RFC4585] provides some guidelines on when to
send NACKs.  It is not expected that a receiver will send a NACK for
every lost RTP packet, rather it needs to consider the cost of
sending NACK feedback, and the importance of the lost packet, to make
an informed decision on whether it is worth telling the sender about
a packet loss event.
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).  The use of retransmissions can also
increase the forward RTP bandwidth, and can potentially worsen the
problem if the packet loss was caused by network congestion.  We
note, however, that retransmission of an important lost packet to
repair decoder state can have lower cost than sending a full intra
frame.  It is not appropriate to blindly retransmit RTP packets in
response to a NACK.  The importance of lost packets and the
likelihood of them arriving in time to be useful needs to be
considered before RTP retransmission is used.
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 is not expected 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 WebRTC session
supports redundant transmission or FEC as a standard feature of that
payload format, then that support MAY be used in the WebRTC session,
subject to any appropriate signalling.
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 open one or more RTP sessions to
each participant in a WebRTC conference, and there can be several
participants.  Each of these RTP sessions can contain different types
of media, and the type of media, bit rate, and number of flows can be
highly asymmetric.  Non-RTP traffic can share the network paths RTP
flows.  Since the network environment is not predictable or stable,
WebRTC endpoints MUST ensure that the RTP traffic they generate can
adapt to match changes in the available network capacity.
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 flows.
Some requirements for congestion control algorithms for WebRTC
sessions are discussed in [I-D.jesup-rtp-congestion-reqs], and it is
expected that a future version of this memo will mandate the use of a
congestion control algorithm that satisfies these requirements.
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 in described [I-D.ietf-avtcore-rtp-circuit-breakers].  The circuit
breaker defines a conservative boundary condition for safe operation,
chosen such that applications that trigger the circuit breaker will
almost certainly be causing severe network congestion.  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 roughly once per
round trip time.  We note that the real-time media traffic might not
have to adapt to changing path conditions as rapidly as needed for
the elastic applications TCP was designed for, but frequent feedback
is still needed to allow the congestion control algorithm to track
the path dynamics.
The total RTCP bandwidth is limited in its transmission rate to a
fraction of the RTP traffic (by default 5%).  RTCP packets are larger
than, e.g., TCP ACKs (even when non-compound RTCP packets are used).
The RTP media stream bit rate thus limits the maximum feedback rate
as a function of the mean RTCP packet size.
Interactive communication might not be able to afford waiting for
packet losses to occur to indicate congestion, because an increase in
play out delay due to queuing (most prominent in wireless networks)
can easily lead to packets being dropped due to late arrival at the
receiver.  Therefore, more sophisticated cues might need to be
reported -- to be defined in a suitable congestion control framework
as noted above -- which, in turn, increase the report size again.
For example, different RTCP XR report blocks (jointly) provide the
necessary details to implement a variety of congestion control
algorithms, but the (compound) report size grows quickly.
In group communication, the share of RTCP bandwidth needs to be
shared by all group members, reducing the capacity and thus the
reporting frequency per node.
Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP
bandwidth, split across two entities in a point-to-point session.  An
endpoint could thus send a report of 100 bytes about every 70ms or
for every other frame in a 30 fps video.
7.3.  Congestion Control Interoperability With Legacy Systems
There are legacy implementations that do not implement RTCP, and
hence do not provide any congestion feedback.  Congestion control
cannot be performed with these end-points.  WebRTC implementations
that need to interwork with such end-points MUST limit their
transmission to a low rate, equivalent to a VoIP call using a low
bandwidth codec, that is unlikely to cause any significant
congestion.
When interworking with legacy implementations that support RTCP using
the RTP/AVP profile [RFC3551], congestion feedback is provided in
RTCP RR packets every few seconds.  Implementations that have to
interwork with such 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
8.  WebRTC Use of RTP: Performance Monitoring
RTCP does contains a basic set of RTP flow monitoring metrics like
packet loss and jitter.  There are a number of extensions that could
be included in the set to be supported.  However, in most cases which
RTP monitoring that is needed depends on the application, which makes
it difficult to select which to include when the set of applications
is very large.
Exposing some metrics in the WebRTC API needs to be considered
allowing the application to gather the measurements of interest.
However, security implications for the different data sets exposed
will need to be considered in this.
(tbd: If any RTCP XR metrics need to be added is still an open
question, but possible to extend at a later stage)
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] 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 of an external signalling channel
that can be used to configure the RTP sessions and their features.
The basic configuration of an RTP session consists of the following
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 in addition to any
cryptographic transformation of the packet content.  As WebRTC
requires the usage of the RTP/SAVPF profile this can be inferred
as there is only a single profile, but in SDP this is still
information that has to be signalled.  Interworking functions
might transform this into RTP/SAVP for a legacy use case by
indicating to the WebRTC end-point a RTP/SAVPF end-point and
limiting the usage of the a=rtcp 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
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 is used for RTP and RTCP flows, this MUST be
signalled (see Section 4.5).  If several RTP sessions are to be
multiplexed onto a single transport layer flow, this MUST also be
signalled (see Section 4.4).
RTP Payload Types, media formats, and media 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).
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 following sections describe how the WebRTC API features map onto
the RTP mechanisms described in this memo.
11.1.  API MediaStream to RTP Mapping
The WebRTC API and its media function have the concept of a WebRTC
MediaStream that consists of zero or more tracks.  A track 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 tracks within a WebRTC
MediaStream are expected to be synchronized.
A track correspond to the media received with one particular SSRC.
There might be additional SSRCs associated with that SSRC, like for
RTP retransmission or Forward Error Correction.  However, one SSRC
will identify an RTP media stream and its timing.
As a result, a WebRTC MediaStream is a collection of SSRCs carrying
the different media included in the synchronised aggregate.
Therefore, also the synchronization state associated with the
included SSRCs are part of concept.  It is important to consider that
there can be multiple different WebRTC MediaStreams containing a
given Track (SSRC).  To avoid unnecessary duplication of media at the
transport level in such cases, a need arises for a binding defining
which WebRTC MediaStreams a given SSRC is associated with at the
signalling level.
A proposal for how the binding between WebRTC MediaStreams and SSRC
can be done is specified in "Cross Session Stream Identification in
the Session Description Protocol" [I-D.alvestrand-rtcweb-msid].
(tbd: This text needs to be improved and achieved consensus on.
Interim meeting in June 2012 shows large differences in opinions.)
12.  RTP Implementation Considerations
The following provide some guidance on the implementation of the RTP
features described in this memo.
This section discusses RTP functionality that is part of the RTP
standard, needed by decisions made, or to enable use cases raised and
their motivations.  This discussion is from an WebRTC end-point
perspective.  It will occasionally talk about central nodes, but as
this specification is for an end-point, this is where the focus lies.
For more discussion on the central nodes and details about RTP
topologies please see Appendix A.
The section will touch on the relation with certain RTP/RTCP
extensions, but will focus on the RTP core functionality.  The
definition of what functionalities and the level of requirement on
implementing it is defined in Section 2.
12.1.  RTP Sessions and PeerConnection
An RTP session is an association among RTP nodes, which have one
common SSRC space.  An RTP session can include any number of end-
points and nodes sourcing, sinking, manipulating or reporting on the
RTP media streams being sent within the RTP session.  A
PeerConnection being a point-to-point association between an end-
point and another node.  That peer node can be both an end-point or
centralized processing node of some type; thus, the RTP session can
terminate immediately on the far end of the PeerConnection, but it
might also continue as further discussed below in Multiparty
(Section 12.3) and Multiple RTP End-points (Section 12.7).
A PeerConnection can contain one or more RTP session depending on how
it is setup and how many UDP flows it uses.  A common usage has been
to have one RTP session per media type, e.g. one for audio and one
for video, each sent over different UDP flows.  However, the default
usage in WebRTC will be to use one RTP session for all media types.
This usage then uses only one UDP flow, as also RTP and RTCP
multiplexing is mandated (Section 4.5).  However, for legacy
interworking and network prioritization (Section 12.9) based on
flows, a WebRTC end-point needs to support a mode of operation where
one RTP session per media type is used.  Currently, each RTP session
has to use its own UDP flow.  Discussions are ongoing if a solution
enabling multiple RTP sessions over a single UDP flow, see
The multi-unicast- or mesh-based multi-party topology (Figure 1) is a
good example for this section as it concerns the relation between RTP
sessions and PeerConnections.  In this topology, each participant
sends individual unicast RTP/UDP/IP flows to each of the other
participants using independent PeerConnections in a full mesh.  This
topology has the benefit of not requiring central nodes.  The
downside is that it increases the used bandwidth at each sender by
requiring one copy of the RTP media streams for each participant that
are part of the same session beyond the sender itself.  Hence, this
topology is limited to scenarios with few participants unless the
media is very low bandwidth.
+---+      +---+
| A |<---->| B |
^         ^
\       /
\     /
+---+
| C |
Figure 1: Multi-unicast
The multi-unicast topology could 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 PeerConnections we recommend that one implements this as
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 based on RTCP.  This has not been seen as a significant downside as
no one has yet seen a clear need for why A would need to know about
the B's and C's communication.  An advantage of using separate RTP
sessions is that it enables using different media bit-rates to 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.
12.2.  Multiple Sources
A WebRTC end-point might have multiple cameras, microphones or audio
inputs and thus a single end-point can source multiple RTP media
streams of the same media type concurrently.  Even if an end-point
does not have multiple media sources of the same media type it has to
support transmission using multiple SSRCs concurrently in the same
RTP session.  This is due to the requirement on an WebRTC end-point
to support multiple media types in one RTP session.  For example, one
audio and one video source can result in the end-point sending with
two different SSRCs in the same RTP session.  As multi-party
conferences are supported, as discussed below in Section 12.3, a
WebRTC end-point will need to be capable of receiving, decoding and
play out multiple RTP media streams of the same type concurrently.
tbd: Are any mechanism needed to signal limitations in the number of
active SSRC that an end-point can handle?
12.3.  Multiparty
There are numerous situations and clear use cases for WebRTC
supporting RTP sessions supporting multi-party.  This can be realized
in a number of ways using a number of different implementation
strategies.  In the following, the focus is on the different set of
WebRTC end-point requirements that arise from different sets of
multi-party topologies.
The multi-unicast mesh (Figure 1)-based multi-party topology
discussed above provides a non-centralized solution but can incur a
heavy tax on the end-points' outgoing paths.  It can also consume
large amount of encoding resources if each outgoing stream is
specifically encoded.  If an encoding is transmitted to multiple
parties, as in some implementations of the mesh case, a requirement
on the end-point becomes to be able to create RTP media streams
suitable for multiple destinations requirements.  These requirements
can both be dependent on transport path and the different end-points
preferences related to play out of the media.
+---+      +------------+      +---+
| A |<---->|            |<---->| B |
+---+      |            |      +---+
|   Mixer    |
| C |<---->|            |<---->| D |
Figure 2: RTP Mixer with Only Unicast Paths
A Mixer (Figure 2) is an RTP end-point that optimizes the
transmission of RTP media streams from certain perspectives, either
by only sending some of the received RTP media stream to any given
receiver or by providing a combined RTP media stream out of a set of
contributing streams.  There are various methods of implementation as
discussed in Appendix A.3.  A common aspect is that these central
nodes can use a number of tools to control the media encoding
provided by a WebRTC end-point.  This includes functions like
requesting breaking the encoding chain and have the encoder produce a
so called Intra frame.  Another is limiting the bit-rate of a given
stream to better suit the mixer view of the multiple down-streams.
Others are controlling the most suitable frame-rate, picture
resolution, the trade-off between frame-rate and spatial quality.
A mixer gets a significant responsibility to correctly perform
congestion control, source identification, manage synchronization
while providing the application with suitable media optimizations.
Mixers also need to be trusted nodes when it comes to security as it
manipulates either RTP or the media itself 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.
12.4.  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.
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].  Still if both
end-points start uses an new SSRC identifier prior to having
signalled it to the peer and received acknowledgement on the
signalling message, there can be collisions.  The Source-Specific SDP
Attributes [RFC5576] contains no mechanism to resolve SSRC collisions
or reject a end-points usage of an SSRC.
There could also appear SSRC values that are not signalled.  This is
more likely than it appears as certain RTP functions need extra SSRCs
to provide functionality related to another (the "main") SSRC, for
example, SSRC multiplexed RTP retransmission [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 PeerConnection level.
The more likely case for SSRC collision is that multiple end-points
in a multiparty conference create new sources and signals those
towards the central server.  In cases where the SSRC/CSRC are
propagated between the different end-points from the central node
collisions can occur.
Another scenario is when the central node manages to connect an end-
point's PeerConnection to another PeerConnection the end-point
already has, thus forming a loop where the end-point will receive its
own traffic.  While is is clearly considered a bug, it is important
that the end-point is able to recognise and handle the case when it
occurs.
12.5.  Contributing Sources
Contributing Sources (CSRC) is a functionality in the RTP header that
allows an RTP node to combine media packets from multiple sources
into one and to identify which sources yielded the result.  For
WebRTC end-points, supporting contributing sources is trivial.  The
set of CSRCs is provided in a given RTP packet.  This information can
then be exposed to the applications using some form of API, possibly
a mapping back into WebRTC MediaStream identities to avoid having to
expose two name spaces and the handling of SSRC collision handling to
the JavaScript.
(tbd: does the API need to provide the ability to add a CSRC list to
an outgoing packet? this is only useful if the sender is mixing
content)
There are also at least one extension that depends on the CSRC list
being used: the Mixer-to-client audio level [RFC6465], which enhances
the information provided by the CSRC to actual energy levels for
audio for each contributing source.
12.6.  Media Synchronization
When an end-point sends media from more than one media source, it
needs to consider if (and which of) these media sources are to be
synchronized.  In RTP/RTCP, synchronisation is provided by having a
set of RTP media streams be indicated as coming from the same
synchronisation context and logical end-point by using the same CNAME
The next provision is that the internal clocks of all media sources,
i.e., what drives the RTP timestamp, can be correlated to a system
clock that is provided in RTCP Sender Reports encoded in an NTP
format.  By correlating all RTP timestamps to a common system clock
for all sources, the timing relation of the different RTP media
streams, also across multiple RTP sessions can be derived at the
receiver and, if desired, the streams can be synchronized.  The
requirement is for the media sender to provide the correlation
information; it is up to the receiver to use it or not.
12.7.  Multiple RTP End-points
Some usages of RTP beyond the recommend topologies result in that an
WebRTC end-point sending media in an RTP session out over a single
PeerConnection will receive receiver reports from multiple RTP
receivers.  Note that receiving multiple receiver reports is expected
because any RTP node that has multiple SSRCs has to report to the
media sender.  The difference here is that they are multiple nodes,
and thus will likely have different path characteristics.
RTP Mixers can create a situation where an end-point experiences a
situation in-between a session with only two end-points and multiple
end-points.  Mixers are expected to not forward RTCP reports
regarding RTP media streams across themselves.  This is due to the
difference in the RTP media streams provided to the different end-
points.  The original media source lacks information about a mixer's
manipulations prior to sending it the different receivers.  This
scenario also results in that an end-point's feedback or requests
goes to the mixer.  When the mixer can't act on this by itself, it is
forced to go to the original media source to fulfil the receivers
request.  This will not necessarily be explicitly visible any RTP and
RTCP traffic, but the interactions and the time to complete them will
indicate such dependencies.
The topologies in which an end-point receives receiver reports from
multiple other end-points are the centralized relay, multicast and an
end-point forwarding an RTP media stream.  Having multiple RTP nodes
receive an RTP flow and send reports and feedback about it has
several impacts.  As previously discussed (Section 12.3) any codec
control and rate control needs to be capable of merging the
requirements and preferences to provide a single best encoding
according to the situation RTP media stream.  Specifically, when it
comes to congestion control it needs to be capable of identifying the
different end-points to form independent congestion state information
for each different path.
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, like TESLA for SRTP [RFC4383].
12.8.  Simulcast
This section discusses simulcast in the meaning of providing a node,
for example a Mixer, with multiple different encoded versions of the
same media source.  In the WebRTC context, this could be accomplished
in two ways.  One is to establish multiple PeerConnection all being
feed the same set of WebRTC MediaStreams.  Another method is to use
multiple WebRTC MediaStreams that are differently configured when it
comes to the media parameters.  This would result in that multiple
different RTP Media Streams (SSRCs) being in used with different
encoding based on the same media source (camera, microphone).
When intending to use simulcast it is important that this is made
explicit so that the end-points don't automatically try to optimize
away the different encodings and provide a single common version.
Thus, some explicit indications that the intent really is to have
different media encodings is likely needed.  It is to be noted that
it might be a central node, rather than an WebRTC end-point that
would benefit from receiving simulcast media sources.
tbd: How to perform simulcast needs to be determined and the
appropriate API or signalling for its usage needs to be defined.
12.9.  Differentiated Treatment of Flows
There are use cases for differentiated treatment of RTP media
streams.  Such differentiation can happen at several places in the
system.  First of all is the prioritization within the end-point
sending the media, which controls, both which RTP media streams that
will be sent, and their allocation of bit-rate out of the current
available aggregate as determined by the congestion control.
Secondly, the network can prioritize packet flows, including RTP
media streams.  Typically, differential treatment includes two steps,
the first being identifying whether an IP packet belongs to a class
that has to be treated differently, the second the actual mechanism
to prioritize packets.  This is done according to three methods;
DiffServ:  The end-point marks a packet with a DiffServ code point to
indicate to the network that the packet belongs to a particular
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.
With the exception of DiffServ both flow based and DPI have issues
with running multiple media types and flows on a single UDP flow,
especially when combined with data transport (SCTP/DTLS).  DPI has
issues because multiple types of flows are aggregated and thus it
becomes more difficult to analyse them.  The flow-based
differentiation will provide the same treatment to all packets within
the flow, i.e., relative prioritization is not possible.  Moreover,
if the resources are limited it might not be possible to provide
differential treatment compared to best-effort for all the flows in a
WebRTC application.
When flow-based differentiation is available the WebRTC application
needs to know about it so that it can provide the separation of the
RTP media streams onto different UDP flows to enable a more granular
usage of flow based differentiation.
DiffServ assumes that either the end-point or a classifier can mark
the packets with an appropriate DSCP so that the packets are treated
according to that marking.  If the end-point is to mark the traffic
two requirements arise in the WebRTC context: 1) The WebRTC
application or browser has to know which DSCP to use and that it can
use them on some set of RTP media streams. 2) The information needs
to be propagated to the operating system when transmitting the
packet.  These issues are discussed in DSCP and other packet markings
for RTCWeb QoS [I-D.ietf-rtcweb-qos].
tbd: The model for providing differentiated treatment needs to be
evolved.  Most of this is not the responsibility of this memo.
However, this memo could include:
1.  How can the application can prioritize MediaStreamTracks
differently in the API?
2.  How MediaStreamTrack prioritization maps to the RTP level, and
what type of marking behaviour can occur on the RTP media stream
and its datagram?
13.  Open Issues
This section contains a summary of the open issues or to be done
things noted in the document:
1.   Need to add references to the RTP payload format for the Video
Codec chosen in Section 4.3.
2.   The methods and solutions for RTP multiplexing over a single
transport is not yet finalized in Section 4.4.
3.   RTP congestion control algorithms will probably require some
feedback information to be conveyed in RTCP.  Are the tools that
are mandated by this memo sufficient, or do we need additional
4.   RTP congestion control could be implementing using either a
sender-based algorithm or a receiver-based algorithm.  To ensure
interoperability, does this memo need to mandate which end is in
charge of congestion control for a path?
5.   Still open if any RTCP XR performance metrics are needed, as
discussed in Section 8.
6.   The API mapping to RTP level concepts has to be agreed and
documented in Section 11.
7.   An open question if any requirements are needed to agree and
limit the number of simultaneously used media sources (SSRCs)
within an RTP session.  See Section 12.2.
8.   Is an API needed for expressing any application level media
mixing of an RTP media stream so that the correct CSRC list can
be set as discussed in Section 12.5?
9.   The method for achieving simulcast of a media source has to be
decided as discussed in Section 12.8.
10.  Possible documentation of what support for differentiated
treatment that are needed on RTP level as the API and the
network level specification matures as discussed in
Section 12.9.
11.  Editing of Appendix A to remove redundancy between this and the
update of RTP Topologies
[I-D.westerlund-avtcore-rtp-topologies-update].
This memo makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as
an RFC.
The security considerations for the WebRTC framework are described in
[I-D.ietf-rtcweb-security].  The overall security architecture for
WebRTC is described in [I-D.ietf-rtcweb-security-arch].
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 (tbd).
tbd: Privacy concerns, and the generation of untraceable CNAMEs, are
under discussion.
The guidelines in [RFC6562] apply when using variable bit rate (VBR)
audio codecs, e.g., Opus or the Mixer audio level header extensions.
The authors would like to thank Harald Alvestrand, Cary Bran, Charles
Eckel and Cullen Jennings for valuable feedback.
[I-D.holmberg-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011.
[I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
Breakers for Unicast Sessions",
draft-ietf-avtcore-rtp-circuit-breakers-00 (work in
[I-D.ietf-avtcore-srtp-encrypted-header-ext]
Lennox, J., "Encryption of Header Extensions in the Secure
Real-Time Transport Protocol (SRTP)",
draft-ietf-avtcore-srtp-encrypted-header-ext-02 (work in
progress), July 2012.
[I-D.ietf-avtext-multiple-clock-rates]
Petit-Huguenin, M. and G. Zorn, "Support for Multiple
Clock Rates in an RTP Session",
draft-ietf-avtext-multiple-clock-rates-06 (work in
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-00 (work in
progress), September 2012.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-04 (work
in progress), June 2012.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web",
draft-ietf-rtcweb-security-03 (work in progress),
June 2012.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "RTCWEB Security Architecture",
draft-ietf-rtcweb-security-arch-05 (work in progress),
[I-D.lennox-rtcweb-rtp-media-type-mux]
Rosenberg, J. and J. Lennox, "Multiplexing Multiple Media
Types In a Single Real-Time Transport Protocol (RTP)
Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work
in progress), October 2011.
[I-D.rescorla-avtcore-6222bis]
Rescorla, E. and A. Begen, "Guidelines for Choosing RTP
Control Protocol (RTCP) Canonical Names (CNAMEs)",
draft-rescorla-avtcore-6222bis-00 (work in progress),
[I-D.terriberry-avp-codecs]
Terriberry, T., "Update to Recommended Codecs for the AVP
RTP Profile", draft-terriberry-avp-codecs-00 (work in
progress), August 2012.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
Single Lower-Layer Transport",
draft-westerlund-avtcore-transport-multiplexing-04 (work
[RFC2736]  Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736,
[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,
[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)",
[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,
[RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
[RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, July 2007.
[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.
[I-D.alvestrand-rtcweb-msid]
Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol",
draft-alvestrand-rtcweb-msid-02 (work in progress),
[I-D.ietf-avt-srtp-ekt]
Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
(work in progress), October 2011.
[I-D.ietf-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS",
draft-ietf-rtcweb-qos-00 (work in progress), October 2012.
[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-09 (work in
progress), June 2012.
[I-D.jesup-rtp-congestion-reqs]
Jesup, R. and H. Alvestrand, "Congestion Control
Requirements For Real Time Media",
draft-jesup-rtp-congestion-reqs-00 (work in progress),
[I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., Perkins, C., and H.
Alvestrand, "Guidelines for using the Multiplexing
Features of RTP",
draft-westerlund-avtcore-multiplex-architecture-02 (work
in progress), July 2012.
[I-D.westerlund-avtcore-rtp-topologies-update]
Westerlund, M. and S. Wenger, "RTP Topologies",
draft-westerlund-avtcore-rtp-topologies-update-01 (work in
[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.
[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,
[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,
[RFC4828]  Floyd, S. and E. Kohler, "TCP Friendly Rate Control
(TFRC): The Small-Packet (SP) Variant", RFC 4828,
[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.
Appendix A.  Supported RTP Topologies
RTP supports both unicast and group communication, with participants
being connected using wide range of transport-layer topologies.  Some
of these topologies involve only the end-points, while others use RTP
translators and mixers to provide in-network processing.  Properties
of some RTP topologies are discussed in
[I-D.westerlund-avtcore-rtp-topologies-update], and we further
describe those expected to be useful for WebRTC in the following.  We
also goes into important RTP session aspects that the topology or
implementation variant can place on a WebRTC end-point.
This section includes RTP topologies beyond the RECOMMENDED ones.
This in an attempt to highlight the differences and the in many case
small differences in implementation to support a larger set of
possible topologies.
(tbd: This section needs reworking and clearer relation to
[I-D.westerlund-avtcore-rtp-topologies-update].)
A.1.  Point to Point
The point-to-point RTP topology (Figure 3) is the simplest scenario
for WebRTC applications.  This is going to be very common for user to
user calls.
+---+         +---+
| A |<------->| B |
Figure 3: Point to Point
This being the basic one lets use the topology to high-light a couple
of details that are common for all RTP usage in the WebRTC context.
First is the intention to multiplex RTP and RTCP over the same UDP-
flow.  Secondly is the question of using only a single RTP session or
one per media type for legacy interoperability.  Thirdly is the
question of using multiple sender sources (SSRCs) per end-point.
Historically, RTP and RTCP have been run on separate UDP ports.  With
the increased use of Network Address/Port Translation (NAPT) this has
become problematic, since maintaining multiple NAT bindings can be
costly.  It also complicates firewall administration, since multiple
ports need to be opened to allow RTP traffic.  To reduce these costs
and session set-up times, support for multiplexing RTP data packets
and RTCP control packets on a single port [RFC5761] will be
In cases where there is only one type of media (e.g., a voice-only
call) this topology will be implemented as a single RTP session, with
bidirectional flows of RTP and RTCP packets, all then multiplexed
onto a single 5-tuple.  If multiple types of media are to be used
(e.g., audio and video), then each type media can be sent as a
separate RTP session using a different 5-tuple, allowing for separate
transport level treatment of each type of media.  Alternatively, all
types of media can be multiplexed onto a single 5-tuple as a single
RTP session, or as several RTP sessions if using a demultiplexing
shim.  Multiplexing different types of media onto a single 5-tuple
places some limitations on how RTP is used, as described in "RTP
Multiplexing Architecture"
[I-D.westerlund-avtcore-multiplex-architecture].  It is not expected
that these limitations will significantly affect the scenarios
targeted by WebRTC, but they can impact interoperability with legacy
An RTP session have good support for simultaneously transport
multiple media sources.  Each media source uses an unique SSRC
identifier and each SSRC has independent RTP sequence number and
timestamp spaces.  This is being utilized in WebRTC for several
cases.  One is to enable multiple media sources of the same type, an
end-point that has two video cameras can potentially transmit video
from both to its peer(s).  Another usage is when a single RTP session
is being used for both multiple media types, thus an end-point can
transmit both audio and video to the peer(s).  Thirdly to support
multi-party cases as will be discussed below support for multiple
SSRC of the same media type is needed.
Thus we can introduce a couple of different notations in the below
two alternate figures of a single peer connection in a point to point
set-up.  The first depicting a setup where the peer connection
established has two different RTP sessions, one for audio and one for
video.  The second one using a single RTP session.  In both cases A
has two video streams to send and one audio stream.  B has only one
audio and video stream.  These are used to illustrate the relation
between a peerConnection, the UDP flow(s), the RTP session(s) and the
SSRCs that will be used in the later cases also.  In the below
figures RTCP flows are not included.  They will flow bi-directionally
between any RTP session instances in the different nodes.
+-A-------------+                 +-B-------------+
| +-PeerC1------|                 |-PeerC1------+ |
| | +-UDP1------|                 |-UDP1------+ | |
| | | +-RTP1----|                 |-RTP1----+ | | |
| | | | +-Audio-|                 |-Audio-+ | | | |
| | | | |    AA1|---------------->|       | | | | |
| | | | |       |<----------------|BA1    | | | | |
| | | | +-------|                 |-------+ | | | |
| | | +---------|                 |---------+ | | |
| | +-----------|                 |-----------+ | |
| |             |                 |             | |
| | | | +-Video-|                 |-Video-+ | | | |
| | | | |    AV1|---------------->|       | | | | |
| | | | |       |<----------------|BV1    | | | | |
| +-------------|                 |-------------+ |
+---------------+                 +---------------+
Figure 4: Point to Point: Multiple RTP sessions
As can be seen above in the Point to Point: Multiple RTP sessions
(Figure 4) the single Peer Connection contains two RTP sessions over
different UDP flows UDP 1 and UDP 2, i.e. their 5-tuples will be
different, normally on source and destination ports.  The first RTP
session (RTP1) carries audio, one stream in each direction AA1 and
BA1.  The second RTP session contains two video streams from A (AV1
and AV2) and one from B to A (BV1).
| | | |         |                 |         | | | |
Figure 5: Point to Point: Single RTP session.
In (Figure 5) there is only a single UDP flow and RTP session (RTP1).
This RTP session carries a total of five (5) RTP media streams
(SSRCs).  From A to B there is Audio (AA1) and two video (AV1 and
AV2).  From B to A there is Audio (BA1) and Video (BV1).
A.2.  Multi-Unicast (Mesh)
For small multiparty calls, it is practical to set up a multi-unicast
topology (Figure 6).  In this topology, each participant sends
individual unicast RTP/UDP/IP flows to each of the other participants
using independent PeerConnections in a full mesh.
This topology has the benefit of not requiring central nodes.  The
media is very low bandwidth.  The multi-unicast topology could 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 PeerConnections we recommend
that one implements this as 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 based on RTCP.  This has not
been seen as a significant downside as now one has yet seen a need
for why A would need to know about the B's and C's communication.  An
advantage of using separate RTP sessions is that it enables using
different media bit-rates to 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.
+-A------------------------+              +-B-------------+
|+---+       +-PeerC1------|              |-PeerC1------+ |
||MIC|       | +-UDP1------|              |-UDP1------+ | |
|+---+       | | +-RTP1----|              |-RTP1----+ | | |
| |  +----+  | | | +-Audio-|              |-Audio-+ | | | |
| +->|ENC1|--+-+-+-+--->AA1|------------->|       | | | | |
| |  +----+  | | | |       |<-------------|BA1    | | | | |
| |          | | | +-------|              |-------+ | | | |
| |          | | +---------|              |---------+ | | |
| |          | +-----------|              |-----------+ | |
| |          +-------------|              |-------------+ |
| |                        |              |---------------+
| |                        |
| |                        |              +-C-------------+
| |          +-PeerC2------|              |-PeerC2------+ |
| |          | +-UDP2------|              |-UDP2------+ | |
| |          | | +-RTP2----|              |-RTP2----+ | | |
|    +----+  | | | |       |<-------------|CA1    | | | | |
|            | | | +-------|              |-------+ | | | |
|            | | +---------|              |---------+ | | |
|            | +-----------|              |-----------+ | |
|            +-------------|              |-------------+ |
+--------------------------+              +---------------+
Figure 7: Session structure for Multi-Unicast Setup
Lets review how the RTP sessions looks from A's perspective by
considering both how the media is a handled and what PeerConnections
and RTP sessions that are set-up in Figure 7.  A's microphone is
captured and the digital audio can then be feed into two different
encoder instances each beeing associated with two different
PeerConnections (PeerC1 and PeerC2) each containing independent RTP
sessions (RTP1 and RTP2).  The SSRCs in each RTP session will be
completely independent and the media bit-rate produced by the encoder
can also be tuned to address any congestion control requirements
between A and B differently then for the path A to C.
For media encodings which are more resource consuming, like video,
one could expect that it will be common that end-points that are
resource constrained will use a different implementation strategy
where the encoder is shared between the different PeerConnections as
shown below Figure 8.
+-A----------------------+                 +-B-------------+
|+---+                   |                 |               |
||CAM|     +-PeerC1------|                 |-PeerC1------+ |
|+---+     | +-UDP1------|                 |-UDP1------+ | |
|  |       | | +-RTP1----|                 |-RTP1----+ | | |
|  V       | | | +-Video-|                 |-Video-+ | | | |
|+----+    | | | |       |<----------------|BV1    | | | | |
||ENC |----+-+-+-+--->AV1|---------------->|       | | | | |
|+----+    | | | +-------|                 |-------+ | | | |
|  |       | | +---------|                 |---------+ | | |
|  |       | +-----------|                 |-----------+ | |
|  |       +-------------|                 |-------------+ |
|  |                     |                 |---------------+
|  |                     |
|  |                     |                 +-C-------------+
|  |       +-PeerC2------|                 |-PeerC2------+ |
|  |       | +-UDP2------|                 |-UDP2------+ | |
|  |       | | | +-Video-|                 |-Video-+ | | | |
|  +-------+-+-+-+--->AV2|---------------->|       | | | | |
|          | | | |       |<----------------|CV1    | | | | |
|          | | | +-------|                 |-------+ | | | |
|          | | +---------|                 |---------+ | | |
|          | +-----------|                 |-----------+ | |
|          +-------------|                 |-------------+ |
+------------------------+                 +---------------+
Figure 8: Single Encoder Multi-Unicast Setup
This will clearly save resources consumed by encoding but does
introduce the need for the end-point A to make decisions on how it
encodes the media so it suites delivery to both B and C. This is not
limited to congestion control, also preferred resolution to receive
based on dispaly area available is another aspect requiring
consideration.  The need for this type of decision logic does arise
in several different topologies and implementation.
A.3.  Mixer Based
An mixer (Figure 9) is a centralised point that selects or mixes
content in a conference to optimise the RTP session so that each end-
point only needs connect to one entity, the mixer.  The mixer can
also reduce the bit-rate needed from the mixer down to a conference
participants as the media sent from the mixer to the end-point can be
optimised in different ways.  These optimisations include methods
like only choosing media from the currently most active speaker or
mixing together audio so that only one audio stream is needed instead
of 3 in the depicted scenario (Figure 9).
Mixers have two downsides, the first is that the mixer has to be a
trusted node as they either performs media operations or at least re-
packetize the media.  Both type of operations requires when using
SRTP that the mixer verifies integrity, decrypts the content, perform
its operation and form new RTP packets, encrypts and integrity
protect them.  This applies to all types of mixers described below.
The second downside is that all these operations and optimization of
the session requires processing.  How much depends on the
implementation as will become evident below.
The implementation of an mixer can take several different forms and
we will discuss the main themes available that doesn't break RTP.
Please note that a Mixer could also contain translator
functionalities, like a media transcoder to adjust the media bit-rate
or codec used on a particular RTP media stream.
A.3.1.  Media Mixing
This type of mixer is one which clearly can be called RTP mixer is
likely the one that most thinks of when they hear the term mixer.
Its basic patter of operation is that it will receive the different
participants RTP media stream.  Select which that are to be included
in a media domain mix of the incoming RTP media streams.  Then create
a single outgoing stream from this mix.
Audio mixing is straight forward and commonly possible to do for a
number of participants.  Lets assume that you want to mix N number of
streams from different participants.  Then the mixer need to perform
decoding N times.  Then it needs to produce N or N+1 mixes, the
reasons that different mixes are needed are so that each contributing
source get a mix which don't contain themselves, as this would result
in an echo.  When N is lower than the number of all participants one
can produce a Mix of all N streams for the group that are curently
not included in the mix, thus N+1 mixes.  These audio streams are
then encoded again, RTP packetized and sent out.
Video can't really be "mixed" and produce something particular useful
for the users, however creating an composition out of the contributed
video streams can be done.  In fact it can be done in a number of
ways, tiling the different streams creating a chessboard, selecting
someone as more important and showing them large and a number of
other sources as smaller is another.  Also here one commonly need to
produce a number of different compositions so that the contributing
part doesn't need to see themselves.  Then the mixer re-encodes the
created video stream, RTP packetize it and send it out
The problem with media mixing is that it both consume large amount of
media processing and encoding resources.  The second is the quality
degradation created by decoding and re-encoding the RTP media stream.
Its advantage is that it is quite simplistic for the clients to
handle as they don't need to handle local mixing and composition.
+-A-------------+             +-MIXER--------------------------+
| +-PeerC1------|             |-PeerC1--------+                |
| | +-UDP1------|             |-UDP1--------+ |                |
| | | +-RTP1----|             |-RTP1------+ | |        +-----+ |
| | | | +-Audio-|             |-Audio---+ | | | +---+  |     | |
| | | | |    AA1|------------>|---------+-+-+-+-|DEC|->|     | |
| | | | |       |<------------|MA1 <----+ | | | +---+  |     | |
| | | | |       |             |(BA1+CA1)|\| | | +---+  |     | |
| | | | +-------|             |---------+ +-+-+-|ENC|<-| B+C | |
| | | +---------|             |-----------+ | | +---+  |     | |
| | +-----------|             |-------------+ |        |  M  | |
| +-------------|             |---------------+        |  E  | |
+---------------+             |                        |  D  | |
|                        |  I  | |
+-B-------------+             |                        |  A  | |
| +-PeerC2------|             |-PeerC2--------+        |     | |
| | +-UDP2------|             |-UDP2--------+ |        |  M  | |
| | | +-RTP2----|             |-RTP2------+ | |        |  I  | |
| | | | +-Audio-|             |-Audio---+ | | | +---+  |  X  | |
| | | | |    BA1|------------>|---------+-+-+-+-|DEC|->|  E  | |
| | | | |       |<------------|MA2 <----+ | | | +---+  |  R  | |
| | | | +-------|             |(BA1+CA1)|\| | | +---+  |     | |
| | | +---------|             |---------+ +-+-+-|ENC|<-| A+C | |
| | +-----------|             |-----------+ | | +---+  |     | |
| +-------------|             |-------------+ |        |     | |
+---------------+             |---------------+        |     | |
|                        |     | |
+-C-------------+             |                        |     | |
| | +-UDP3------|             |-UDP3--------+ |        |     | |
| | | +-RTP3----|             |-RTP3------+ | |        |     | |
| | | | |    CA1|------------>|---------+-+-+-+-|DEC|->|     | |
| | | +---------|             |---------+ +-+-+-|ENC|<-| A+B | |
| +-------------|             |-------------+ |        +-----+ |
+---------------+             |---------------+                |
+--------------------------------+
Figure 10: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be very straight forward as
can be seen in Figure 10.  The mixer present one SSRC towards the
peer client, e.g.  MA1 to Peer A, which is the media mix of the other
participants.  As each peer receives a different version produced by
the mixer there are no actual relation between the different RTP
sessions in the actual media or the transport level information.
There is however one connection between RTP1-RTP3 in this figure.  It
has to do with the SSRC space and the identity information.  When A
receives the MA1 stream which is a combination of BA1 and CA1 streams
in the other PeerConnections RTP could enable the mixer to include
CSRC information in the MA1 stream to identify the contributing
source BA1 and CA1.
The CSRC has in its turn utility in RTP extensions, like the in
Section 5.2.3 discussed Mixer to Client audio levels RTP header
extension [RFC6465].  If the SSRC from one PeerConnection are used as
CSRC in another PeerConnection then RTP1, RTP2 and RTP3 becomes one
joint session as they have a common SSRC space.  At this stage one
also need to consider which RTCP information one need to expose in
the different legs.  For the above situation commonly nothing more
than the Source Description (SDES) information and RTCP BYE for CSRC
need to be exposed.  The main goal would be to enable the correct
binding against the application logic and other information sources.
This also enables loop detection in the RTP session.
A.3.1.1.  RTP Session Termination
There exist an possible implementation choice to have the RTP
sessions being separated between the different legs in the multi-
party communication session and only generate RTP media streams in
each without carrying on RTP/RTCP level any identity information
about the contributing sources.  This removes both the functionality
that CSRC can provide and the possibility to use any extensions that
build on CSRC and the loop detection.  It might appear a
simplification if SSRC collision would occur between two different
end-points as they can be avoided to be resolved and instead remapped
between the independent sessions if at all exposed.  However, SSRC/
CSRC remapping requires that SSRC/CSRC are never exposed to the
WebRTC JavaScript client to use as reference.  This as they only have
local importance if they are used on a multi-party session scope the
result would be mis-referencing.  Also SSRC collision handling will
still be needed as it can occur between the mixer and the end-point.
Session termination might appear to resolve some issues, it however
creates other issues that needs resolving, like loop detection,
identification of contributing sources and the need to handle mapped
identities and ensure that the right one is used towards the right
identities and never used directly between multiple end-points.
A.3.2.  Media Switching
An RTP Mixer based on media switching avoids the media decoding and
encoding cycle in the mixer, but not the decryption and re-encryption
cycle as one rewrites RTP headers.  This both reduces the amount of
computational resources needed in the mixer and increases the media
quality per transmitted bit.  This is achieve by letting the mixer
have a number of SSRCs that represents conceptual or functional
streams the mixer produces.  These streams are created by selecting
media from one of the by the mixer received RTP media streams and
forward the media using the mixers own SSRCs.  The mixer can then
switch between available sources if that is needed by the concept for
the source, like currently active speaker.
To achieve a coherent RTP media stream from the mixer's SSRC the
mixer is forced to rewrite the incoming RTP packet's header.  First
the SSRC field has to be set to the value of the Mixer's SSRC.
Secondly, the sequence number is set to the next in the sequence of
outgoing packets it sent.  Thirdly the RTP timestamp value needs to
be adjusted using an offset that changes each time one switch media
source.  Finally depending on the negotiation the RTP payload type
value representing this particular RTP payload configuration might
have to be changed if the different PeerConnections have not arrived
on the same numbering for a given configuration.  This also requires
that the different end-points do support a common set of codecs,
otherwise media transcoding for codec compatibility is still needed.
Lets consider the operation of media switching mixer that supports a
video conference with six participants (A-F) where the two latest
speakers in the conference are shown to each participants.  Thus the
mixer has two SSRCs sending video to each peer.
| | | | +-Video-|             |-Video---+ | | |        |     | |
| | | | |    AV1|------------>|---------+-+-+-+------->|     | |
| | | | |       |<------------|MV1 <----+-+-+-+-BV1----|     | |
| | | | |       |<------------|MV2 <----+-+-+-+-EV1----|     | |
| | | | +-------|             |---------+ | | |        |     | |
| | | +---------|             |-----------+ | |        |     | |
| | +-----------|             |-------------+ |        |  S  | |
| +-------------|             |---------------+        |  W  | |
+---------------+             |                        |  I  | |
|                        |  T  | |
+-B-------------+             |                        |  C  | |
| +-PeerC2------|             |-PeerC2--------+        |  H  | |
| | | +-RTP2----|             |-RTP2------+ | |        |  M  | |
| | | | +-Video-|             |-Video---+ | | |        |  A  | |
| | | | |    BV1|------------>|---------+-+-+-+------->|  T  | |
| | | | |       |<------------|MV3 <----+-+-+-+-AV1----|  R  | |
| | | | |       |<------------|MV4 <----+-+-+-+-EV1----|  I  | |
| | | | +-------|             |---------+ | | |        |  X  | |
| | +-----------|             |-------------+ |        |     | |
| +-------------|             |---------------+        |     | |
+---------------+             |                        |     | |
:                        :     : :
+-F-------------+             |                        |     | |
| | | | |    CV1|------------>|---------+-+-+-+------->|     | |
| | | | |       |<------------|MV11 <---+-+-+-+-AV1----|     | |
| | | | |       |<------------|MV12 <---+-+-+-+-EV1----|     | |
| | +-----------|             |-------------+ |        +-----+ |
| +-------------|             |---------------+                |
+---------------+             +--------------------------------+
Figure 11: Media Switching RTP Mixer
The Media Switching RTP mixer can similar to the Media Mixing one
reduce the bit-rate needed towards the different peers by selecting
and switching in a sub-set of RTP media streams out of the ones it
receives from the conference participations.
To ensure that a media receiver can correctly decode the RTP media
stream after a switch, it becomes necessary to ensure for state
saving codecs that they start from default state at the point of
switching.  Thus one common tool for video is to request that the
encoding creates an intra picture, something that isn't dependent on
earlier state.  This can be done using Full Intra Request RTCP codec
control message as discussed in Section 5.1.1.
Also in this type of mixer one could consider to terminate the RTP
sessions fully between the different PeerConnection.  The same
arguments and considerations as discussed in Appendix A.3.1.1 applies
A.3.3.  Media Projecting
Another method for handling media in the RTP mixer is to project all
potential sources (SSRCs) into a per end-point independent RTP
session.  The mixer can then select which of the potential sources
that are currently actively transmitting media, despite that the
mixer in another RTP session receives media from that end-point.
This is similar to the media switching Mixer but have some important
differences in RTP details.
| | | | |       |<------------|BV1 <----+-+-+-+--------|     | |
| | | | |       |<------------|CV1 <----+-+-+-+--------|     | |
| | | | |       |<------------|DV1 <----+-+-+-+--------|     | |
| | | | |       |<------------|EV1 <----+-+-+-+--------|     | |
| | | | |       |<------------|FV1 <----+-+-+-+--------|     | |
| | | | |       |<------------|AV1 <----+-+-+-+--------|  R  | |
| | | | |       |<------------|CV1 <----+-+-+-+--------|  I  | |
| | | | |       | :    :    : |: :  : : : : : :  :  : :|  X  | |
| | | | |       |<------------|AV1 <----+-+-+-+--------|     | |
| | | | |       | :    :    : |: :  : : : : : :  :  : :|     | |
Figure 12: Media Projecting Mixer
So in this six participant conference depicted above in (Figure 12)
one can see that end-point A will in this case be aware of 5 incoming
SSRCs, BV1-FV1.  If this mixer intend to have the same behavior as in
Appendix A.3.2 where the mixer provides the end-points with the two
latest speaking end-points, then only two out of these five SSRCs
will concurrently transmit media to A. As the mixer selects which
source in the different RTP sessions that transmit media to the end-
points each RTP media stream will require some rewriting when being
projected from one session into another.  The main thing is that the
sequence number will need to be consecutively incremented based on
the packet actually being transmitted in each RTP session.  Thus the
RTP sequence number offset will change each time a source is turned
on in RTP session.
As the RTP sessions are independent the SSRC numbers used can be
handled independently also thus working around any SSRC collisions by
having remapping tables between the RTP sessions.  However the
related WebRTC MediaStream signalling need to be correspondingly
changed to ensure consistent WebRTC MediaStream to SSRC mappings
between the different PeerConnections and the same comment that
higher functions MUST NOT use SSRC as references to RTP media streams
applies also here.
The mixer will also be responsible to act on any RTCP codec control
requests coming from an end-point and decide if it can act on it
locally or needs to translate the request into the RTP session that
contains the media source.  Both end-points and the mixer will need
to implement conference related codec control functionalities to
provide a good experience.  Full Intra Request to request from the
media source to provide switching points between the sources,
Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
to aggregate congestion control response towards the media source and
have it adjust its bit-rate in case the limitation is not in the
source to mixer link.
This version of the mixer also puts different requirements on the
end-point when it comes to decoder instances and handling of the RTP
media streams providing media.  As each projected SSRC can at any
time provide media the end-point either needs to handle having thus
many allocated decoder instances or have efficient switching of
decoder contexts in a more limited set of actual decoder instances to
cope with the switches.  The WebRTC application also gets more
responsibility to update how the media provides is to be presented to
the user.
A.4.  Translator Based
There is also a variety of translators.  The core commonality is that
they do not need to make themselves visible in the RTP level by
having an SSRC themselves.  Instead they sit between one or more end-
point and perform translation at some level.  It can be media
transcoding, protocol translation or covering missing functionality
for a legacy end-point or simply relay packets between transport
domains or to realize multi-party.  We will go in details below.
A.4.1.  Transcoder
A transcoder operates on media level and really used for two
purposes, the first is to allow two end-points that doesn't have a
common set of media codecs to communicate by translating from one
codec to another.  The second is to change the bit-rate to a lower
one.  For WebRTC end-points communicating with each other only the
first one is relevant.  In certain legacy deployment media transcoder
will be necessary to ensure both codecs and bit-rate falls within the
envelope the legacy end-point supports.
As transcoding requires access to the media, the transcoder has to be
within the security context and access any media encryption and
integrity keys.  On the RTP plane a media transcoder will in practice
fork the RTP session into two different domains that are highly
decoupled when it comes to media parameters and reporting, but not
identities.  To maintain signalling bindings to SSRCs a transcoder is
likely needing to use the SSRC of one end-point to represent the
transcoded RTP media stream to the other end-point(s).  The
congestion control loop can be terminated in the transcoder as the
media bit-rate being sent by the transcoder can be adjusted
independently of the incoming bit-rate.  However, for optimizing
performance and resource consumption the translator needs to consider
what signals or bit-rate reductions it needs to send towards the
source end-point.  For example receiving a 2.5 Mbps video stream and
then send out a 250 kbps video stream after transcoding is a waste of
resources.  In most cases a 500 kbps video stream from the source in
the right resolution is likely to provide equal quality after
transcoding as the 2.5 Mbps source stream.  At the same time
increasing media bit-rate further than what is needed to represent
the incoming quality accurate is also wasted resources.
+-A-------------+             +-Translator------------------+
| +-PeerC1------|             |-PeerC1--------+             |
| | +-UDP1------|             |-UDP1--------+ |             |
| | | +-RTP1----|             |-RTP1------+ | |             |
| | | | +-Audio-|             |-Audio---+ | | | +---+       |
| | | | |    AA1|------------>|---------+-+-+-+-|DEC|----+  |
| | | | |       |<------------|BA1 <----+ | | | +---+    |  |
| | | | |       |             |         |\| | | +---+    |  |
| | | | +-------|             |---------+ +-+-+-|ENC|<-+ |  |
| | | +---------|             |-----------+ | | +---+  | |  |
| | +-----------|             |-------------+ |        | |  |
| +-------------|             |---------------+        | |  |
+---------------+             |                        | |  |
|                        | |  |
+-B-------------+             |                        | |  |
| +-PeerC2------|             |-PeerC2--------+        | |  |
| | +-UDP2------|             |-UDP2--------+ |        | |  |
| | | +-RTP1----|             |-RTP1------+ | |        | |  |
| | | | +-Audio-|             |-Audio---+ | | | +---+  | |  |
| | | | |    BA1|------------>|---------+-+-+-+-|DEC|--+ |  |
| | | | |       |<------------|AA1 <----+ | | | +---+    |  |
| | | | +-------|             |---------+ +-+-+-|ENC|<---+  |
| | | +---------|             |-----------+ | | +---+       |
| | +-----------|             |-------------+ |             |
| +-------------|             |---------------+             |
+---------------+             +-----------------------------+
Figure 13: Media Transcoder
Figure 13 exposes some important details.  First of all you can see
the SSRC identifiers used by the translator are the corresponding
end-points.  Secondly, there is a relation between the RTP sessions
in the two different PeerConnections that are represented by having
both parts be identified by the same level and they need to share
certain contexts.  Also certain type of RTCP messages will need to be
bridged between the two parts.  Certain RTCP feedback messages are
likely needed to be sourced by the translator in response to actions
by the translator and its media encoder.
A.4.2.  Gateway / Protocol Translator
Gateways are used when some protocol feature that are needed are not
supported by an end-point wants to participate in session.  This RTP
translator in Figure 14 takes on the role of ensuring that from the
perspective of participant A, participant B appears as a fully
compliant WebRTC end-point (that is, it is the combination of the
Translator and participant B that looks like a WebRTC end point).
+------------+
+---+      | Translator |      +---+
| A |<---->| to legacy  |<---->| B |
+---+      | end-point  |      +---+
WebRTC     |            |     Legacy
Figure 14: Gateway (RTP translator) towards legacy end-point
For WebRTC there are a number of requirements that could force the
need for a gateway if a WebRTC end-point is to communicate with a
legacy end-point, such as support of ICE and DTLS-SRTP for key
management.  On RTP level the main functions that might be missing in
a legacy implementation that otherwise support RTP are RTCP in
general, SRTP implementation, congestion control and feedback
messages needed to make it work.
| +-PeerC1------|             |-PeerC1------+               |
| | +-UDP1------|             |-UDP1------+ |               |
| | | +-RTP1----|             |-RTP1-----------------------+|
| | | | +-Audio-|             |-Audio---+                  ||
| | | | |    AA1|------------>|---------+----------------+ ||
| | | | |       |<------------|BA1 <----+--------------+ | ||
| | | | |       |<---RTCP---->|<--------+----------+   | | ||
| | | | +-------|             |---------+      +---+-+ | | ||
| | | +---------|             |---------------+| T   | | | ||
| | +-----------|             |-----------+ | || R   | | | ||
| +-------------|             |-------------+ || A   | | | ||
+---------------+             |               || N   | | | ||
|               || S   | | | ||
+-B-(Legacy)----+             |               || L   | | | ||
|               |             |               || A   | | | ||
|   +-UDP2------|             |-UDP2------+   || T   | | | ||
|   | +-RTP1----|             |-RTP1----------+| E   | | | ||
|   | | +-Audio-|             |-Audio---+      +---+-+ | | ||
|   | | |       |<---RTCP---->|<--------+----------+   | | ||
|   | | |    BA1|------------>|---------+--------------+ | ||
|   | | |       |<------------|AA1 <----+----------------+ ||
|   | | +-------|             |---------+                  ||
|   | +---------|             |----------------------------+|
|   +-----------|             |-----------+                 |
|               |             |                             |
Figure 15: RTP/RTCP Protocol Translator
The legacy gateway can be implemented in several ways and what it
need to change is highly dependent on what functions it need to proxy
for the legacy end-point.  One possibility is depicted in Figure 15
where the RTP media streams are compatible and forward without
changes.  However, their RTP header values are captured to enable the
RTCP translator to create RTCP reception information related to the
leg between the end-point and the translator.  This can then be
combined with the more basic RTCP reports that the legacy endpoint
(B) provides to give compatible and expected RTCP reporting to A.
Thus enabling at least full congestion control on the path between A
and the translator.  If B has limited possibilities for congestion
response for the media then the translator might need the capability
to perform media transcoding to address cases where it otherwise
would need to terminate media transmission.
As the translator are generating RTP/RTCP traffic on behalf of B to A
it will need to be able to correctly protect these packets that it
translates or generates.  Thus security context information are
needed in this type of translator if it operates on the RTP/RTCP
packet content or media.  In fact one of the more likely scenario is
that the translator (gateway) will need to have two different
security contexts one towards A and one towards B and for each RTP/
RTCP packet do a authenticity verification, decryption followed by a
encryption and integrity protection operation to resolve mismatch in
security systems.
A.4.3.  Relay
There exist a class of translators that operates on transport level
below RTP and thus do not effect RTP/RTCP packets directly.  They
come in two distinct flavours, the one used to bridge between two
different transport or address domains to more function as a gateway
and the second one which is to to provide a group communication
feature as depicted below in Figure 16.
| Translator |
Figure 16: RTP Translator (Relay) with Only Unicast Paths
The first kind is straight forward and is likely to exist in WebRTC
context when an legacy end-point is compatible with the exception for
ICE, and thus needs a gateway that terminates the ICE and then
forwards all the RTP/RTCP traffic and key management to the end-point
only rewriting the IP/UDP to forward the packet to the legacy node.
The second type is useful if one wants a less complex central node or
a central node that is outside of the security context and thus do
not have access to the media.  This relay takes on the role of
forwarding the media (RTP and RTCP) packets to the other end-points
but doesn't perform any RTP or media processing.  Such a device
simply forwards the media from each sender to all of the other
participants, and is sometimes called a transport-layer translator.
In Figure 16, participant A will only need to send a media once to
the relay, which will redistribute it by sending a copy of the stream
to participants B, C, and D. Participant A will still receive three
RTP streams with the media from B, C and D if they transmit
simultaneously.  This is from an RTP perspective resulting in an RTP
session that behaves equivalent to one transporter over an IP Any
Source Multicast (ASM).
This results in one common RTP session between all participants
despite that there will be independent PeerConnections created to the
translator as depicted below Figure 17.
+-A-------------+             +-RELAY--------------------------+
| | | +-RTP1----|             |-RTP1-------------------------+ |
| | | | +-Video-|             |-Video---+                    | |
| | | | |    AV1|------------>|---------------------------+  | |
| | | | |       |<------------|BV1 <--------------------+ |  | |
| | | | |       |<------------|CV1 <------------------+ | |  | |
| | | | +-------|             |---------+             | | |  | |
| | | +---------|             |-------------------+   ^ ^ V  | |
| | +-----------|             |-------------+ |   |   | | |  | |
| +-------------|             |---------------+   |   | | |  | |
+---------------+             |                   |   | | |  | |
|                   |   | | |  | |
+-B-------------+             |                   |   | | |  | |
| +-PeerC2------|             |-PeerC2--------+   |   | | |  | |
| | +-UDP2------|             |-UDP2--------+ |   |   | | |  | |
| | | +-RTP2----|             |-RTP1--------------+   | | |  | |
| | | | +-Video-|             |-Video---+             | | |  | |
| | | | |    BV1|------------>|-----------------------+ | |  | |
| | | | |       |<------------|AV1 <----------------------+  | |
| | | | |       |<------------|CV1 <--------------------+ |  | |
| | | +---------|             |-------------------+   | | |  | |
| | +-----------|             |-------------+ |   |   V ^ V  | |
:                   |   | | |  | |
+-C-------------+             |                   |   | | |  | |
| | | | |    CV1|------------>|-------------------------+ |  | |
| | | | |       |<------------|BV1 <------------------+      | |
| | | | +-------|             |---------+                    | |
| | | +---------|             |------------------------------+ |
| | +-----------|             |-------------+ |                |
Figure 17: Transport Multi-party Relay
As the Relay RTP and RTCP packets between the UDP flows as indicated
by the arrows for the media flow a given WebRTC end-point, like A
will see the remote sources BV1 and CV1.  There will be also two
different network paths between A, and B or C. This results in that
the client A has to be capable of handling that when determining
congestion state that there might exist multiple destinations on the
far side of a PeerConnection and that these paths have to be treated
differently.  It also results in a requirement to combine the
different congestion states into a decision to transmit a particular
RTP media stream suitable to all participants.
It is also important to note that the relay can not perform selective
relaying of some sources and not others.  The reason is that the RTCP
reporting in that case becomes inconsistent and without explicit
information about it being blocked has to be interpreted as severe
In this usage it is also necessary that the session management has
configured a common set of RTP configuration including RTP payload
formats as when A sends a packet with pt=97 it will arrive at both B
and C carrying pt=97 and having the same packetization and encoding,
no entity will have manipulated the packet.
When it comes to security there exist some additional requirements to
ensure that the property that the relay can't read the media traffic
is enforced.  First of all the key to be used has to be agreed such
so that the relay doesn't get it, e.g. no DTLS-SRTP handshake with
the relay, instead some other method needs to be used.  Secondly, the
keying structure has to be capable of handling multiple end-points in
the same RTP session.
The second problem can basically be solved in two ways.  Either a
common master key from which all derive their per source key for
SRTP.  The second alternative which might be more practical is that
each end-point has its own key used to protects all RTP/RTCP packets
it sends.  Each participants key are then distributed to the other
participants.  This second method could be implemented using DTLS-
SRTP to a special key server and then use Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] to distribute the actual used key to the
other participants in the RTP session Figure 18.  The first one could
be achieved using MIKEY messages in SDP.
+---+                               +---+
|   |         +-----------+         |   |
| A |<------->| DTLS-SRTP |<------->| C |
|   |<--   -->|   HOST    |<--   -->|   |
+---+   \ /   +-----------+   \ /   +---+
X                     X
+---+   / \   +-----------+   / \   +---+
|   |<--   -->|    RTP    |<--   -->|   |
| B |<------->|   RELAY   |<------->| D |
Figure 18: DTLS-SRTP host and RTP Relay Separated
The relay can still verify that a given SSRC isn't used or spoofed by
another participant within the multi-party session by binding SSRCs
on their first usage to a given source address and port pair.
Packets carrying that source SSRC from other addresses can be
suppressed to prevent spoofing.  This is possible as long as SRTP is
used which leaves the SSRC of the packet originator in RTP and RTCP
packets in the clear.  If such packet level method for enforcing
source authentication within the group, then there exist
cryptographic methods such as TESLA [RFC4383] that could be used for
true source authentication.
A.5.  End-point Forwarding
An WebRTC end-point (B in Figure 19) will receive a WebRTC
MediaStream (set of SSRCs) over a PeerConnection (from A).  For the
moment is not decided if the end-point is allowed or not to in its
turn send that WebRTC MediaStream over another PeerConnection to C.
This section discusses the RTP and end-point implications of allowing
such functionality, which on the API level is extremely simplistic to
+---+    +---+    +---+
| A |--->| B |--->| C |
Figure 19: MediaStream Forwarding
There exist two main approaches to how B forwards the media from A to
C. The first one is to simply relay the RTP media stream.  The second
one is for B to act as a transcoder.  Lets consider both approaches.
A relay approach will result in that the WebRTC end-points will have
to have the same capabilities as being discussed in Relay
(Appendix A.4.3).  Thus A will see an RTP session that is extended
beyond the PeerConnection and see two different receiving end-points
with different path characteristics (B and C).  Thus A's congestion
control needs to be capable of handling this.  The security solution
can either support mechanism that allows A to inform C about the key
A is using despite B and C having agreed on another set of keys.
Alternatively B will decrypt and then re-encrypt using a new key.
The relay based approach has the advantage that B does not need to
transcode the media thus both maintaining the quality of the encoding
and reducing B's complexity requirements.  If the right security
solutions are supported then also C will be able to verify the
authenticity of the media coming from A. As downside A are forced to
take both B and C into consideration when delivering content.
The media transcoder approach is similar to having B act as Mixer
terminating the RTP session combined with the transcoder as discussed
in Appendix A.4.1.  A will only see B as receiver of its media.  B
will responsible to produce a RTP media stream suitable for the B to
C PeerConnection.  This might require media transcoding for
congestion control purpose to produce a suitable bit-rate.  Thus
loosing media quality in the transcoding and forcing B to spend the
resource on the transcoding.  The media transcoding does result in a
separation of the two different legs removing almost all
dependencies.  B could choice to implement logic to optimize its
media transcoding operation, by for example requesting media
properties that are suitable for C also, thus trying to avoid it
having to transcode the content and only forward the media payloads
between the two sides.  For that optimization to be practical WebRTC
end-points have to support sufficiently good tools for codec control.
A.6.  Simulcast
for example a stream switching Mixer, with multiple different encoded
version of the same media source.  In the WebRTC context that appears
to be most easily accomplished by establishing multiple
PeerConnection all being feed the same set of WebRTC MediaStreams.
Each PeerConnection is then configured to deliver a particular media
quality and thus media bit-rate.  This will work well as long as the
end-point implements media encoding according to Figure 7.  Then each
PeerConnection will receive an independently encoded version and the
codec parameters can be agreed specifically in the context of this
PeerConnection.
For simulcast to work one needs to prevent that the end-point deliver
content encoded as depicted in Figure 8.  If a single encoder
instance is feed to multiple PeerConnections the intention of
performing simulcast will fail.
Thus it needs to be considered to explicitly signal which of the two
implementation strategies that are desired and which will be done.
At least making the application and possible the central node
interested in receiving simulcast of an end-points RTP media streams
to be aware if it will function or not.
Colin Perkins
School of Computing Science
Glasgow  G12 8QQ
Email: csp@csperkins.org
Magnus Westerlund
Farogatan 6
SE-164 80 Kista
Email: magnus.westerlund@ericsson.com
Joerg Ott
Espoo  02150
Email: jorg.ott@aalto.fi