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In: MissingRef
RTCWEB Working Group                                          C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Standards Track                           M. Westerlund
Expires: October 25, 2014                                       Ericsson
                                                                  J. Ott
                                                        Aalto University
                                                          April 23, 2014


  Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
                     draft-ietf-rtcweb-rtp-usage-13

Abstract

   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc. between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on October 25, 2014.

Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents



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   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Rationale . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   4.  WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . .   5
     4.1.  RTP and RTCP  . . . . . . . . . . . . . . . . . . . . . .   5
     4.2.  Choice of the RTP Profile . . . . . . . . . . . . . . . .   7
     4.3.  Choice of RTP Payload Formats . . . . . . . . . . . . . .   7
     4.4.  Use of RTP Sessions . . . . . . . . . . . . . . . . . . .   9
     4.5.  RTP and RTCP Multiplexing . . . . . . . . . . . . . . . .   9
     4.6.  Reduced Size RTCP . . . . . . . . . . . . . . . . . . . .  10
     4.7.  Symmetric RTP/RTCP  . . . . . . . . . . . . . . . . . . .  10
     4.8.  Choice of RTP Synchronisation Source (SSRC) . . . . . . .  11
     4.9.  Generation of the RTCP Canonical Name (CNAME) . . . . . .  11
     4.10. Handling of Leap Seconds  . . . . . . . . . . . . . . . .  12
   5.  WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . .  12
     5.1.  Conferencing Extensions and Topologies  . . . . . . . . .  12
       5.1.1.  Full Intra Request (FIR)  . . . . . . . . . . . . . .  14
       5.1.2.  Picture Loss Indication (PLI) . . . . . . . . . . . .  14
       5.1.3.  Slice Loss Indication (SLI) . . . . . . . . . . . . .  14
       5.1.4.  Reference Picture Selection Indication (RPSI) . . . .  15
       5.1.5.  Temporal-Spatial Trade-off Request (TSTR) . . . . . .  15
       5.1.6.  Temporary Maximum Media Stream Bit Rate Request
               (TMMBR) . . . . . . . . . . . . . . . . . . . . . . .  15
     5.2.  Header Extensions . . . . . . . . . . . . . . . . . . . .  16
       5.2.1.  Rapid Synchronisation . . . . . . . . . . . . . . . .  16
       5.2.2.  Client-to-Mixer Audio Level . . . . . . . . . . . . .  16
       5.2.3.  Mixer-to-Client Audio Level . . . . . . . . . . . . .  17
   6.  WebRTC Use of RTP: Improving Transport Robustness . . . . . .  17
     6.1.  Negative Acknowledgements and RTP Retransmission  . . . .  17
     6.2.  Forward Error Correction (FEC)  . . . . . . . . . . . . .  18
   7.  WebRTC Use of RTP: Rate Control and Media Adaptation  . . . .  19
     7.1.  Boundary Conditions and Circuit Breakers  . . . . . . . .  20
     7.2.  RTCP Limitations for Congestion Control . . . . . . . . .  20
     7.3.  Congestion Control Interoperability and Legacy Systems  .  22
   8.  WebRTC Use of RTP: Performance Monitoring . . . . . . . . . .  23
   9.  WebRTC Use of RTP: Future Extensions  . . . . . . . . . . . .  24
   10. Signalling Considerations . . . . . . . . . . . . . . . . . .  24
   11. WebRTC API Considerations . . . . . . . . . . . . . . . . . .  25
   12. RTP Implementation Considerations . . . . . . . . . . . . . .  28
     12.1.  Configuration and Use of RTP Sessions  . . . . . . . . .  28



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       12.1.1.  Use of Multiple Media Sources Within an RTP Session   28
       12.1.2.  Use of Multiple RTP Sessions . . . . . . . . . . . .  29
       12.1.3.  Differentiated Treatment of RTP Packet Streams . . .  34
     12.2.  Media Source, RTP Packet Streams, and Participant
            Identification . . . . . . . . . . . . . . . . . . . . .  35
       12.2.1.  Media Source . . . . . . . . . . . . . . . . . . . .  36
       12.2.2.  SSRC Collision Detection . . . . . . . . . . . . . .  36
       12.2.3.  Media Synchronisation Context  . . . . . . . . . . .  37
   13. Security Considerations . . . . . . . . . . . . . . . . . . .  38
   14. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  39
   15. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  39
   16. References  . . . . . . . . . . . . . . . . . . . . . . . . .  39
     16.1.  Normative References . . . . . . . . . . . . . . . . . .  39
     16.2.  Informative References . . . . . . . . . . . . . . . . .  42
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  44

1.  Introduction

   The Real-time Transport Protocol (RTP) [RFC3550] provides a framework
   for delivery of audio and video teleconferencing data and other real-
   time media applications.  Previous work has defined the RTP protocol,
   along with numerous profiles, payload formats, and other extensions.
   When combined with appropriate signalling, these form the basis for
   many teleconferencing systems.

   The Web Real-Time communication (WebRTC) framework provides the
   protocol building blocks to support direct, interactive, real-time
   communication using audio, video, collaboration, games, etc., between
   two peers' web-browsers.  This memo describes how the RTP framework
   is to be used in the WebRTC context.  It proposes a baseline set of
   RTP features that are to be implemented by all WebRTC-aware end-
   points, along with suggested extensions for enhanced functionality.

   This memo specifies a protocol intended for use within the WebRTC
   framework, but is not restricted to that context.  An overview of the
   WebRTC framework is given in [I-D.ietf-rtcweb-overview].

   The structure of this memo is as follows.  Section 2 outlines our
   rationale in preparing this memo and choosing these RTP features.
   Section 3 defines terminology.  Requirements for core RTP protocols
   are described in Section 4 and suggested RTP extensions are described
   in Section 5.  Section 6 outlines mechanisms that can increase
   robustness to network problems, while Section 7 describes congestion
   control and rate adaptation mechanisms.  The discussion of mandated
   RTP mechanisms concludes in Section 8 with a review of performance
   monitoring and network management tools that can be used in the
   WebRTC context.  Section 9 gives some guidelines for future
   incorporation of other RTP and RTP Control Protocol (RTCP) extensions



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   into this framework.  Section 10 describes requirements placed on the
   signalling channel.  Section 11 discusses the relationship between
   features of the RTP framework and the WebRTC application programming
   interface (API), and Section 12 discusses RTP implementation
   considerations.  The memo concludes with security considerations
   (Section 13) and IANA considerations (Section 14).

2.  Rationale

   The RTP framework comprises the RTP data transfer protocol, the RTP
   control protocol, and numerous RTP payload formats, profiles, and
   extensions.  This range of add-ons has allowed RTP to meet various
   needs that were not envisaged by the original protocol designers, and
   to support many new media encodings, but raises the question of what
   extensions are to be supported by new implementations.  The
   development of the WebRTC framework provides an opportunity to review
   the available RTP features and extensions, and to define a common
   baseline feature set for all WebRTC implementations of RTP.  This
   builds on the past 20 years development of RTP to mandate the use of
   extensions that have shown widespread utility, while still remaining
   compatible with the wide installed base of RTP implementations where
   possible.

   RTP and RTCP extensions that are not discussed in this document can
   be implemented by WebRTC end-points if they are beneficial for new
   use cases.  However, they are not necessary to address the WebRTC use
   cases and requirements identified in
   [I-D.ietf-rtcweb-use-cases-and-requirements].

   While the baseline set of RTP features and extensions defined in this
   memo is targeted at the requirements of the WebRTC framework, it is
   expected to be broadly useful for other conferencing-related uses of
   RTP.  In particular, it is likely that this set of RTP features and
   extensions will be appropriate for other desktop or mobile video
   conferencing systems, or for room-based high-quality telepresence
   applications.

3.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].  The RFC
   2119 interpretation of these key words applies only when written in
   ALL CAPS.  Lower- or mixed-case uses of these key words are not to be
   interpreted as carrying special significance in this memo.

   We define the following additional terms:




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   WebRTC MediaStream:  The MediaStream concept defined by the W3C in
      the WebRTC API [W3C.WD-mediacapture-streams-20130903].

   Transport-layer Flow:  A uni-directional flow of transport packets
      that are identified by having a particular 5-tuple of source IP
      address, source port, destination IP address, destination port,
      and transport protocol used.

   Bi-directional Transport-layer Flow:  A bi-directional transport-
      layer flow is a transport-layer flow that is symmetric.  That is,
      the transport-layer flow in the reverse direction has a 5-tuple
      where the source and destination address and ports are swapped
      compared to the forward path transport-layer flow, and the
      transport protocol is the same.

   This document uses the terminology from
   [I-D.ietf-avtext-rtp-grouping-taxonomy].  Other terms are used
   according to their definitions from the RTP Specification [RFC3550].
   We especially note the following frequently used terms: RTP Packet
   Stream, RTP Session, and End-point.

4.  WebRTC Use of RTP: Core Protocols

   The following sections describe the core features of RTP and RTCP
   that need to be implemented, along with the mandated RTP profiles.
   Also described are the core extensions providing essential features
   that all WebRTC implementations need to implement to function
   effectively on today's networks.

4.1.  RTP and RTCP

   The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be
   implemented as the media transport protocol for WebRTC.  RTP itself
   comprises two parts: the RTP data transfer protocol, and the RTP
   control protocol (RTCP).  RTCP is a fundamental and integral part of
   RTP, and MUST be implemented in all WebRTC applications.

   The following RTP and RTCP features are sometimes omitted in limited
   functionality implementations of RTP, but are REQUIRED in all WebRTC
   implementations:

   o  Support for use of multiple simultaneous SSRC values in a single
      RTP session, including support for RTP end-points that send many
      SSRC values simultaneously, following [RFC3550] and
      [I-D.ietf-avtcore-rtp-multi-stream].  Support for the RTCP
      optimisations for multi-SSRC sessions defined in
      [I-D.ietf-avtcore-rtp-multi-stream-optimisation] is RECOMMENDED.




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   o  Random choice of SSRC on joining a session; collision detection
      and resolution for SSRC values (see also Section 4.8).

   o  Support for reception of RTP data packets containing CSRC lists,
      as generated by RTP mixers, and RTCP packets relating to CSRCs.

   o  Sending correct synchronisation information in the RTCP Sender
      Reports, to allow receivers to implement lip-synchronisation;
      support for the rapid RTP synchronisation extensions (see
      Section 5.2.1) is RECOMMENDED.

   o  Support for multiple synchronisation contexts.  Participants that
      send multiple simultaneous RTP packet streams SHOULD do so as part
      of a single synchronisation context, using a single RTCP CNAME for
      all streams and allowing receivers to play the streams out in a
      synchronised manner.  For compatibility with potential future
      versions of this specification, or for interoperability with non-
      WebRTC devices through a gateway, receivers MUST support multiple
      synchronisation contexts, indicated by the use of multiple RTCP
      CNAMEs in an RTP session.  This specification requires the usage
      of a single CNAME when sending RTP Packet Streams in some
      circumstances, see Section 4.9.

   o  Support for sending and receiving RTCP SR, RR, SDES, and BYE
      packet types, with OPTIONAL support for other RTCP packet types
      unless mandated by other parts of this specification;
      implementations MUST ignore unknown RTCP packet types.  Note that
      additional RTCP Packet types are used by the RTP/SAVPF Profile
      (Section 4.2) and the other RTCP extensions (Section 5).

   o  Support for multiple end-points in a single RTP session, and for
      scaling the RTCP transmission interval according to the number of
      participants in the session; support for randomised RTCP
      transmission intervals to avoid synchronisation of RTCP reports;
      support for RTCP timer reconsideration.

   o  Support for configuring the RTCP bandwidth as a fraction of the
      media bandwidth, and for configuring the fraction of the RTCP
      bandwidth allocated to senders, e.g., using the SDP "b=" line
      [RFC4566][RFC3556].  Support for the reduced minimum RTCP
      reporting interval described in Section 6.2 of [RFC3550] is
      RECOMMENDED.

   It is known that a significant number of legacy RTP implementations,
   especially those targeted at VoIP-only systems, do not support all of
   the above features, and in some cases do not support RTCP at all.
   Implementers are advised to consider the requirements for graceful
   degradation when interoperating with legacy implementations.



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   Other implementation considerations are discussed in Section 12.

4.2.  Choice of the RTP Profile

   The complete specification of RTP for a particular application domain
   requires the choice of an RTP Profile.  For WebRTC use, the Extended
   Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as
   extended by [RFC7007], MUST be implemented.  The RTP/SAVPF profile is
   the combination of basic RTP/AVP profile [RFC3551], the RTP profile
   for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP
   profile (RTP/SAVP) [RFC3711].

   The RTCP-based feedback extensions [RFC4585] are needed for the
   improved RTCP timer model.  This allows more flexible transmission of
   RTCP packets in response to events, rather than strictly according to
   bandwidth, and is vital for being able to report congestion signals
   as well as media events.  These extensions also allow saving RTCP
   bandwidth, and an end-point will commonly only use the full RTCP
   bandwidth allocation if there are many events that require feedback.
   The timer rules are also needed to make use of the RTP conferencing
   extensions discussed in Section 5.1.

      Note: The enhanced RTCP timer model defined in the RTP/AVPF
      profile is backwards compatible with legacy systems that implement
      only the RTP/AVP or RTP/SAVP profile, given some constraints on
      parameter configuration such as the RTCP bandwidth value and "trr-
      int" (the most important factor for interworking with RTP/(S)AVP
      end-points via a gateway is to set the trr-int parameter to a
      value representing 4 seconds).

   The secure RTP (SRTP) profile extensions [RFC3711] are needed to
   provide media encryption, integrity protection, replay protection and
   a limited form of source authentication.  WebRTC implementations MUST
   NOT send packets using the basic RTP/AVP profile or the RTP/AVPF
   profile; they MUST employ the full RTP/SAVPF profile to protect all
   RTP and RTCP packets that are generated (i.e., implementations MUST
   use SRTP and SRTCP).  The RTP/SAVPF profile MUST be configured using
   the cipher suites, DTLS-SRTP protection profiles, keying mechanisms,
   and other parameters described in [I-D.ietf-rtcweb-security-arch].

4.3.  Choice of RTP Payload Formats

   The set of mandatory to implement codecs and RTP payload formats for
   WebRTC is not specified in this memo, instead they are defined in
   separate specifications, such as [I-D.ietf-rtcweb-audio].
   Implementations can support any codec for which an RTP payload format
   and associated signalling is defined.  Implementation cannot assume
   that the other participants in an RTP session understand any RTP



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   payload format, no matter how common; the mapping between RTP payload
   type numbers and specific configurations of particular RTP payload
   formats MUST be agreed before those payload types/formats can be
   used.  In an SDP context, this can be done using the "a=rtpmap:" and
   "a=fmtp:" attributes associated with an "m=" line, along with any
   other SDP attributes needed to configure the RTP payload format.

   End-points can signal support for multiple RTP payload formats, or
   multiple configurations of a single RTP payload format, as long as
   each unique RTP payload format configuration uses a different RTP
   payload type number.  As outlined in Section 4.8, the RTP payload
   type number is sometimes used to associate an RTP packet stream with
   a signalling context.  This association is possible provided unique
   RTP payload type numbers are used in each context.  For example, an
   RTP packet stream can be associated with an SDP "m=" line by
   comparing the RTP payload type numbers used by the RTP packet stream
   with payload types signalled in the "a=rtpmap:" lines in the media
   sections of the SDP.  If RTP packet streams are being associated with
   signalling contexts based on the RTP payload type, then the
   assignment of RTP payload type numbers MUST be unique across
   signalling contexts; if the same RTP payload format configuration is
   used in multiple contexts, then a different RTP payload type number
   has to be assigned in each context to ensure uniqueness.  If the RTP
   payload type number is not being used to associate RTP packet streams
   with a signalling context, then the same RTP payload type number can
   be used to indicate the exact same RTP payload format configuration
   in multiple contexts.  A single RTP payload type number MUST NOT be
   assigned to different RTP payload formats, or different
   configurations of the same RTP payload format, within a single RTP
   session (note that the different "m=" lines in an SDP bundle group
   [I-D.ietf-mmusic-sdp-bundle-negotiation] form a single RTP session).

   An end-point that has signalled support for multiple RTP payload
   formats SHOULD be able to accept data in any of those payload formats
   at any time, unless it has previously signalled limitations on its
   decoding capability.  This requirement is constrained if several
   types of media (e.g., audio and video) are sent in the same RTP
   session.  In such a case, a source (SSRC) is restricted to switching
   only between the RTP payload formats signalled for the type of media
   that is being sent by that source; see Section 4.4.  To support rapid
   rate adaptation by changing codec, RTP does not require advance
   signalling for changes between RTP payload formats used by a single
   SSRC that were signalled during session set-up.

   An RTP sender that changes between two RTP payload types that use
   different RTP clock rates MUST follow the recommendations in
   Section 4.1 of [RFC7160].  RTP receivers MUST follow the
   recommendations in Section 4.3 of [RFC7160] in order to support



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   sources that switch between clock rates in an RTP session (these
   recommendations for receivers are backwards compatible with the case
   where senders use only a single clock rate).

4.4.  Use of RTP Sessions

   An association amongst a set of end-points communicating using RTP is
   known as an RTP session [RFC3550].  An end-point can be involved in
   several RTP sessions at the same time.  In a multimedia session, each
   type of media has typically been carried in a separate RTP session
   (e.g., using one RTP session for the audio, and a separate RTP
   session using a different transport-layer flow for the video).
   WebRTC implementations of RTP are REQUIRED to implement support for
   multimedia sessions in this way, separating each session using
   different transport-layer flows for compatibility with legacy
   systems.

   In modern day networks, however, with the widespread use of network
   address/port translators (NAT/NAPT) and firewalls, it is desirable to
   reduce the number of transport-layer flows used by RTP applications.
   This can be done by sending all the RTP packet streams in a single
   RTP session, which will comprise a single transport-layer flow (this
   will prevent the use of some quality-of-service mechanisms, as
   discussed in Section 12.1.3).  Implementations are therefore also
   REQUIRED to support transport of all RTP packet streams, independent
   of media type, in a single RTP session using a single transport layer
   flow, according to [I-D.ietf-avtcore-multi-media-rtp-session].  If
   multiple types of media are to be used in a single RTP session, all
   participants in that RTP session MUST agree to this usage.  In an SDP
   context, [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to
   signal such a bundle of RTP packet streams forming a single RTP
   session.

   Further discussion about the suitability of different RTP session
   structures and multiplexing methods to different scenarios are
   suitable can be found in [I-D.ietf-avtcore-multiplex-guidelines].

4.5.  RTP and RTCP Multiplexing

   Historically, RTP and RTCP have been run on separate transport layer
   flows (e.g., two UDP ports for each RTP session, one port for RTP and
   one port for RTCP).  With the increased use of Network Address/Port
   Translation (NAT/NAPT) this has become problematic, since maintaining
   multiple NAT bindings can be costly.  It also complicates firewall
   administration, since multiple ports need to be opened to allow RTP
   traffic.  To reduce these costs and session set-up times, support for
   multiplexing RTP data packets and RTCP control packets on a single
   transport-layer flow for each RTP session is REQUIRED, provided it is



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   negotiated in the signalling channel before use as specified in
   [RFC5761].  For backwards compatibility, implementations are also
   REQUIRED to support RTP and RTCP sent on separate transport-layer
   flows.

   Note that the use of RTP and RTCP multiplexed onto a single
   transport-layer flow ensures that there is occasional traffic sent on
   that port, even if there is no active media traffic.  This can be
   useful to keep NAT bindings alive, and is the recommend method for
   application level keep-alives of RTP sessions [RFC6263].

4.6.  Reduced Size RTCP

   RTCP packets are usually sent as compound RTCP packets, and [RFC3550]
   requires that those compound packets start with an Sender Report (SR)
   or Receiver Report (RR) packet.  When using frequent RTCP feedback
   messages under the RTP/AVPF Profile [RFC4585] these statistics are
   not needed in every packet, and unnecessarily increase the mean RTCP
   packet size.  This can limit the frequency at which RTCP packets can
   be sent within the RTCP bandwidth share.

   To avoid this problem, [RFC5506] specifies how to reduce the mean
   RTCP message size and allow for more frequent feedback.  Frequent
   feedback, in turn, is essential to make real-time applications
   quickly aware of changing network conditions, and to allow them to
   adapt their transmission and encoding behaviour.  Support for non-
   compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be
   negotiated using the signalling channel before use.  For backwards
   compatibility, implementations are also REQUIRED to support the use
   of compound RTCP feedback packets if the remote end-point does not
   agree to the use of non-compound RTCP in the signalling exchange.

4.7.  Symmetric RTP/RTCP

   To ease traversal of NAT and firewall devices, implementations are
   REQUIRED to implement and use Symmetric RTP [RFC4961].  The reason
   for using symmetric RTP is primarily to avoid issues with NATs and
   Firewalls by ensuring that the send and receive RTP packet streams,
   as well as RTCP, are actually bi-directional transport-layer flows.
   This will keep alive the NAT and firewall pinholes, and help indicate
   consent that the receive direction is a transport-layer flow the
   intended recipient actually wants.  In addition, it saves resources,
   specifically ports at the end-points, but also in the network as NAT
   mappings or firewall state is not unnecessary bloated.  The amount of
   per flow QoS state kept in the network is also reduced.






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4.8.  Choice of RTP Synchronisation Source (SSRC)

   Implementations are REQUIRED to support signalled RTP synchronisation
   source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined
   in Section 4.1 and Section 5 of [RFC5576].  Implementations MUST also
   support the "previous-ssrc" source attribute defined in Section 6.2
   of [RFC5576].  Other per-SSRC attributes defined in [RFC5576] MAY be
   supported.

   Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP
   session is OPTIONAL.  Implementations MUST be prepared to accept RTP
   and RTCP packets using SSRCs that have not been explicitly signalled
   ahead of time.  Implementations MUST support random SSRC assignment,
   and MUST support SSRC collision detection and resolution, according
   to [RFC3550].  When using signalled SSRC values, collision detection
   MUST be performed as described in Section 5 of [RFC5576].

   It is often desirable to associate an RTP packet stream with a non-
   RTP context.  For users of the WebRTC API a mapping between SSRCs and
   MediaStreamTracks are provided per Section 11.  For gateways or other
   usages it is possible to associate an RTP packet stream with an "m="
   line in a session description formatted using SDP.  If SSRCs are
   signalled this is straightforward (in SDP the "a=ssrc:" line will be
   at the media level, allowing a direct association with an "m=" line).
   If SSRCs are not signalled, the RTP payload type numbers used in an
   RTP packet stream are often sufficient to associate that packet
   stream with a signalling context (e.g., if RTP payload type numbers
   are assigned as described in Section 4.3 of this memo, the RTP
   payload types used by an RTP packet stream can be compared with
   values in SDP "a=rtpmap:" lines, which are at the media level in SDP,
   and so map to an "m=" line).

4.9.  Generation of the RTCP Canonical Name (CNAME)

   The RTCP Canonical Name (CNAME) provides a persistent transport-level
   identifier for an RTP end-point.  While the Synchronisation Source
   (SSRC) identifier for an RTP end-point can change if a collision is
   detected, or when the RTP application is restarted, its RTCP CNAME is
   meant to stay unchanged for the duration of a RTCPeerConnection
   [W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely
   identified and associated with their RTP packet streams within a set
   of related RTP sessions.

   Each RTP end-point MUST have at least one RTCP CNAME, and that RTCP
   CNAME MUST be unique within the RTCPeerConnection.  RTCP CNAMEs
   identify a particular synchronisation context, i.e., all SSRCs
   associated with a single RTCP CNAME share a common reference clock.
   If an end-point has SSRCs that are associated with several



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   unsynchronised reference clocks, and hence different synchronisation
   contexts, it will need to use multiple RTCP CNAMEs, one for each
   synchronisation context.

   Taking the discussion in Section 11 into account, a WebRTC end-point
   MUST NOT use more than one RTCP CNAME in the RTP sessions belonging
   to single RTCPeerConnection (that is, an RTCPeerConnection forms a
   synchronisation context).  RTP middleboxes MAY generate RTP packet
   streams associated with more than one RTCP CNAME, to allow them to
   avoid having to resynchronize media from multiple different end-
   points part of a multi-party RTP session.

   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME, but these are not sufficient in the presence of NAT
   devices.  In addition, long-term persistent identifiers can be
   problematic from a privacy viewpoint (Section 13).  Accordingly, a
   WebRTC endpoint MUST generate a new, unique, short-term persistent
   RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a
   single exception; if explicitly requested at creation an
   RTCPeerConnection MAY use the same CNAME as as an existing
   RTCPeerConnection within their common same-origin context.

   An WebRTC end-point MUST support reception of any CNAME that matches
   the syntax limitations specified by the RTP specification [RFC3550]
   and cannot assume that any CNAME will be chosen according to the form
   suggested above.

4.10.  Handling of Leap Seconds

   The guidelines regarding handling of leap seconds to limit their
   impact on RTP media playout and synchronization given in [RFC7164]
   SHOULD be followed.

5.  WebRTC Use of RTP: Extensions

   There are a number of RTP extensions that are either needed to obtain
   full functionality, or extremely useful to improve on the baseline
   performance, in the WebRTC application context.  One set of these
   extensions is related to conferencing, while others are more generic
   in nature.  The following subsections describe the various RTP
   extensions mandated or suggested for use within the WebRTC context.

5.1.  Conferencing Extensions and Topologies

   RTP is a protocol that inherently supports group communication.
   Groups can be implemented by having each endpoint send its RTP packet
   streams to an RTP middlebox that redistributes the traffic, by using
   a mesh of unicast RTP packet streams between endpoints, or by using



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   an IP multicast group to distribute the RTP packet streams.  These
   topologies can be implemented in a number of ways as discussed in
   [I-D.ietf-avtcore-rtp-topologies-update].

   While the use of IP multicast groups is popular in IPTV systems, the
   topologies based on RTP middleboxes are dominant in interactive video
   conferencing environments.  Topologies based on a mesh of unicast
   transport-layer flows to create a common RTP session have not seen
   widespread deployment to date.  Accordingly, WebRTC implementations
   are not expected to support topologies based on IP multicast groups
   or to support mesh-based topologies, such as a point-to-multipoint
   mesh configured as a single RTP session (Topo-Mesh in the terminology
   of [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
   multipoint mesh constructed using several RTP sessions, implemented
   in the WebRTC context using independent RTCPeerConnections, can be
   expected to be utilised by WebRTC applications and needs to be
   supported.

   WebRTC implementations of RTP endpoints implemented according to this
   memo are expected to support all the topologies described in
   [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
   and receive unicast RTP packet streams to and from some peer device,
   provided that peer can participate in performing congestion control
   on the RTP packet streams.  The peer device could be another RTP
   endpoint, or it could be an RTP middlebox that redistributes the RTP
   packet streams to other RTP endpoints.  This limitation means that
   some of the RTP middlebox-based topologies are not suitable for use
   in the WebRTC environment.  Specifically:

   o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
      since they make the use of RTCP for congestion control and quality
      of service reports problematic (see Section 3.8 of
      [I-D.ietf-avtcore-rtp-topologies-update]).

   o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
      SHOULD NOT be used because its safe use requires a congestion
      control algorithm or RTP circuit breaker that handles point to
      multipoint, which has not yet been standardised.

   The following topology can be used, however it has some issues worth
   noting:

   o  Content modifying MCUs with RTCP termination (Topo-RTCP-
      terminating-MCU) MAY be used.  Note that in this RTP Topology, RTP
      loop detection and identification of active senders is the
      responsibility of the WebRTC application; since the clients are
      isolated from each other at the RTP layer, RTP cannot assist with




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      these functions (see section 3.9 of
      [I-D.ietf-avtcore-rtp-topologies-update]).

   The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
   designed to be used with centralised conferencing, where an RTP
   middlebox (e.g., a conference bridge) receives a participant's RTP
   packet streams and distributes them to the other participants.  These
   extensions are not necessary for interoperability; an RTP end-point
   that does not implement these extensions will work correctly, but
   might offer poor performance.  Support for the listed extensions will
   greatly improve the quality of experience and, to provide a
   reasonable baseline quality, some of these extensions are mandatory
   to be supported by WebRTC end-points.

   The RTCP conferencing extensions are defined in Extended RTP Profile
   for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
   AVPF) [RFC4585] and the memo on Codec Control Messages (CCM) in RTP/
   AVPF [RFC5104]; they are fully usable by the Secure variant of this
   profile (RTP/SAVPF) [RFC5124].

5.1.1.  Full Intra Request (FIR)

   The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1
   of the Codec Control Messages [RFC5104].  It is used to make the
   mixer request a new Intra picture from a participant in the session.
   This is used when switching between sources to ensure that the
   receivers can decode the video or other predictive media encoding
   with long prediction chains.  WebRTC senders MUST understand and
   react to FIR feedback messages they receiver, since this greatly
   improves the user experience when using centralised mixer-based
   conferencing.  Support for sending FIR messages is OPTIONAL.

5.1.2.  Picture Loss Indication (PLI)

   The Picture Loss Indication message is defined in Section 6.3.1 of
   the RTP/AVPF profile [RFC4585].  It is used by a receiver to tell the
   sending encoder that it lost the decoder context and would like to
   have it repaired somehow.  This is semantically different from the
   Full Intra Request above as there could be multiple ways to fulfil
   the request.  WebRTC senders MUST understand and react to PLI
   feedback messages as a loss tolerance mechanism.  Receivers MAY send
   PLI messages.

5.1.3.  Slice Loss Indication (SLI)

   The Slice Loss Indication message is defined in Section 6.3.2 of the
   RTP/AVPF profile [RFC4585].  It is used by a receiver to tell the
   encoder that it has detected the loss or corruption of one or more



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   consecutive macro blocks, and would like to have these repaired
   somehow.  It is RECOMMENDED that receivers generate SLI feedback
   messages if slices are lost when using a codec that supports the
   concept of macro blocks.  A sender that receives an SLI feedback
   message SHOULD attempt to repair the lost slice(s).

5.1.4.  Reference Picture Selection Indication (RPSI)

   Reference Picture Selection Indication (RPSI) messages are defined in
   Section 6.3.3 of the RTP/AVPF profile [RFC4585].  Some video encoding
   standards allow the use of older reference pictures than the most
   recent one for predictive coding.  If such a codec is in use, and if
   the encoder has learnt that encoder-decoder synchronisation has been
   lost, then a known as correct reference picture can be used as a base
   for future coding.  The RPSI message allows this to be signalled.
   Receivers that detect that encoder-decoder synchronisation has been
   lost SHOULD generate an RPSI feedback message if codec being used
   supports reference picture selection.  A RTP packet stream sender
   that receives such an RPSI message SHOULD act on that messages to
   change the reference picture, if it is possible to do so within the
   available bandwidth constraints, and with the codec being used.

5.1.5.  Temporal-Spatial Trade-off Request (TSTR)

   The temporal-spatial trade-off request and notification are defined
   in Sections 3.5.2 and 4.3.2 of [RFC5104].  This request can be used
   to ask the video encoder to change the trade-off it makes between
   temporal and spatial resolution, for example to prefer high spatial
   image quality but low frame rate.  Support for TSTR requests and
   notifications is OPTIONAL.

5.1.6.  Temporary Maximum Media Stream Bit Rate Request (TMMBR)

   The TMMBR feedback message is defined in Sections 3.5.4 and 4.2.1 of
   the Codec Control Messages [RFC5104].  This request and its
   notification message are used by a media receiver to inform the
   sending party that there is a current limitation on the amount of
   bandwidth available to this receiver.  This can be various reasons
   for this: for example, an RTP mixer can use this message to limit the
   media rate of the sender being forwarded by the mixer (without doing
   media transcoding) to fit the bottlenecks existing towards the other
   session participants.  WebRTC senders are REQUIRED to implement
   support for TMMBR messages, and MUST follow bandwidth limitations set
   by a TMMBR message received for their SSRC.  The sending of TMMBR
   requests is OPTIONAL.






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5.2.  Header Extensions

   The RTP specification [RFC3550] provides the capability to include
   RTP header extensions containing in-band data, but the format and
   semantics of the extensions are poorly specified.  The use of header
   extensions is OPTIONAL in the WebRTC context, but if they are used,
   they MUST be formatted and signalled following the general mechanism
   for RTP header extensions defined in [RFC5285], since this gives
   well-defined semantics to RTP header extensions.

   As noted in [RFC5285], the requirement from the RTP specification
   that header extensions are "designed so that the header extension may
   be ignored" [RFC3550] stands.  To be specific, header extensions MUST
   only be used for data that can safely be ignored by the recipient
   without affecting interoperability, and MUST NOT be used when the
   presence of the extension has changed the form or nature of the rest
   of the packet in a way that is not compatible with the way the stream
   is signalled (e.g., as defined by the payload type).  Valid examples
   of RTP header extensions might include metadata that is additional to
   the usual RTP information, but that can safely be ignored without
   compromising interoperability.

5.2.1.  Rapid Synchronisation

   Many RTP sessions require synchronisation between audio, video, and
   other content.  This synchronisation is performed by receivers, using
   information contained in RTCP SR packets, as described in the RTP
   specification [RFC3550].  This basic mechanism can be slow, however,
   so it is RECOMMENDED that the rapid RTP synchronisation extensions
   described in [RFC6051] be implemented in addition to RTCP SR-based
   synchronisation.  The rapid synchronisation extensions use the
   general RTP header extension mechanism [RFC5285], which requires
   signalling, but are otherwise backwards compatible.

5.2.2.  Client-to-Mixer Audio Level

   The Client to Mixer Audio Level extension [RFC6464] is an RTP header
   extension used by an endpoint to inform a mixer about the level of
   audio activity in the packet to which the header is attached.  This
   enables an RTP middlebox to make mixing or selection decisions
   without decoding or detailed inspection of the payload, reducing the
   complexity in some types of mixer.  It can also save decoding
   resources in receivers, which can choose to decode only the most
   relevant RTP packet streams based on audio activity levels.

   The Client-to-Mixer Audio Level [RFC6464] header extension is
   RECOMMENDED to be implemented.  If this header extension is
   implemented, it is REQUIRED that implementations are capable of



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   encrypting the header extension according to [RFC6904] since the
   information contained in these header extensions can be considered
   sensitive.  It is further RECOMMENDED that this encryption is used,
   unless the encryption has been explicitly disabled through API or
   signalling.

5.2.3.  Mixer-to-Client Audio Level

   The Mixer to Client Audio Level header extension [RFC6465] provides
   an endpoint with the audio level of the different sources mixed into
   a common mix by a RTP mixer.  This enables a user interface to
   indicate the relative activity level of each session participant,
   rather than just being included or not based on the CSRC field.  This
   is a pure optimisations of non critical functions, and is hence
   OPTIONAL to implement.  If this header extension is implemented, it
   is REQUIRED that implementations are capable of encrypting the header
   extension according to [RFC6904] since the information contained in
   these header extensions can be considered sensitive.  It is further
   RECOMMENDED that this encryption is used, unless the encryption has
   been explicitly disabled through API or signalling.

6.  WebRTC Use of RTP: Improving Transport Robustness

   There are tools that can make RTP packet streams robust against
   packet loss and reduce the impact of loss on media quality.  However,
   they all add overhead compared to a non-robust stream.  The overhead
   needs to be considered, and the aggregate bit-rate MUST be rate
   controlled to avoid causing network congestion (see Section 7).  As a
   result, improving robustness might require a lower base encoding
   quality, but has the potential to deliver that quality with fewer
   errors.  The mechanisms described in the following sub-sections can
   be used to improve tolerance to packet loss.

6.1.  Negative Acknowledgements and RTP Retransmission

   As a consequence of supporting the RTP/SAVPF profile, implementations
   can send negative acknowledgements (NACKs) for RTP data packets
   [RFC4585].  This feedback can be used to inform a sender of the loss
   of particular RTP packets, subject to the capacity limitations of the
   RTCP feedback channel.  A sender can use this information to optimise
   the user experience by adapting the media encoding to compensate for
   known lost packets.

   RTP packet stream Senders are REQUIRED to understand the Generic NACK
   message defined in Section 6.2.1 of [RFC4585], but MAY choose to
   ignore some or all of this feedback (following Section 4.2 of
   [RFC4585]).  Receivers MAY send NACKs for missing RTP packets.
   Guidelines on when to send NACKs are provided in [RFC4585].  It is



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   not expected that a receiver will send a NACK for every lost RTP
   packet, rather it needs to consider the cost of sending NACK
   feedback, and the importance of the lost packet, to make an informed
   decision on whether it is worth telling the sender about a packet
   loss event.

   The RTP Retransmission Payload Format [RFC4588] offers the ability to
   retransmit lost packets based on NACK feedback.  Retransmission needs
   to be used with care in interactive real-time applications to ensure
   that the retransmitted packet arrives in time to be useful, but can
   be effective in environments with relatively low network RTT (an RTP
   sender can estimate the RTT to the receivers using the information in
   RTCP SR and RR packets, as described at the end of Section 6.4.1 of
   [RFC3550]).  The use of retransmissions can also increase the forward
   RTP bandwidth, and can potentially caused increased packet loss if
   the original packet loss was caused by network congestion.  We note,
   however, that retransmission of an important lost packet to repair
   decoder state can have lower cost than sending a full intra frame.
   It is not appropriate to blindly retransmit RTP packets in response
   to a NACK.  The importance of lost packets and the likelihood of them
   arriving in time to be useful needs to be considered before RTP
   retransmission is used.

   Receivers are REQUIRED to implement support for RTP retransmission
   packets [RFC4588].  Senders MAY send RTP retransmission packets in
   response to NACKs if the RTP retransmission payload format has been
   negotiated for the session, and if the sender believes it is useful
   to send a retransmission of the packet(s) referenced in the NACK.  An
   RTP sender does not need to retransmit every NACKed packet.

6.2.  Forward Error Correction (FEC)

   The use of Forward Error Correction (FEC) can provide an effective
   protection against some degree of packet loss, at the cost of steady
   bandwidth overhead.  There are several FEC schemes that are defined
   for use with RTP.  Some of these schemes are specific to a particular
   RTP payload format, others operate across RTP packets and can be used
   with any payload format.  It needs to be noted that using redundant
   encoding or FEC will lead to increased play out delay, which needs to
   be considered when choosing the redundancy or FEC formats and their
   respective parameters.

   If an RTP payload format negotiated for use in a RTCPeerConnection
   supports redundant transmission or FEC as a standard feature of that
   payload format, then that support MAY be used in the
   RTCPeerConnection, subject to any appropriate signalling.





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   There are several block-based FEC schemes that are designed for use
   with RTP independent of the chosen RTP payload format.  At the time
   of this writing there is no consensus on which, if any, of these FEC
   schemes is appropriate for use in the WebRTC context.  Accordingly,
   this memo makes no recommendation on the choice of block-based FEC
   for WebRTC use.

7.  WebRTC Use of RTP: Rate Control and Media Adaptation

   WebRTC will be used in heterogeneous network environments using a
   variety set of link technologies, including both wired and wireless
   links, to interconnect potentially large groups of users around the
   world.  As a result, the network paths between users can have widely
   varying one-way delays, available bit-rates, load levels, and traffic
   mixtures.  Individual end-points can send one or more RTP packet
   streams to each participant in a WebRTC conference, and there can be
   several participants.  Each of these RTP packet streams can contain
   different types of media, and the type of media, bit rate, and number
   of RTP packet streams as well as transport-layer flows can be highly
   asymmetric.  Non-RTP traffic can share the network paths with RTP
   transport-layer flows.  Since the network environment is not
   predictable or stable, WebRTC end-points MUST ensure that the RTP
   traffic they generate can adapt to match changes in the available
   network capacity.

   The quality of experience for users of WebRTC implementation is very
   dependent on effective adaptation of the media to the limitations of
   the network.  End-points have to be designed so they do not transmit
   significantly more data than the network path can support, except for
   very short time periods, otherwise high levels of network packet loss
   or delay spikes will occur, causing media quality degradation.  The
   limiting factor on the capacity of the network path might be the link
   bandwidth, or it might be competition with other traffic on the link
   (this can be non-WebRTC traffic, traffic due to other WebRTC flows,
   or even competition with other WebRTC flows in the same session).

   An effective media congestion control algorithm is therefore an
   essential part of the WebRTC framework.  However, at the time of this
   writing, there is no standard congestion control algorithm that can
   be used for interactive media applications such as WebRTC's flows.
   Some requirements for congestion control algorithms for
   RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements].
   It is expected that a future version of this memo will mandate the
   use of a congestion control algorithm that satisfies these
   requirements.






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7.1.  Boundary Conditions and Circuit Breakers

   In the absence of a concrete congestion control algorithm, all WebRTC
   implementations MUST implement the RTP circuit breaker algorithm that
   is described in [I-D.ietf-avtcore-rtp-circuit-breakers].  The RTP
   circuit breaker is designed to enable applications to recognise and
   react to situations of extreme network congestion.  However, since
   the RTP circuit breaker might not be triggered until congestion
   becomes extreme, it cannot be considered a substitute for congestion
   control, and applications MUST also implement congestion control to
   allow them to adapt to changes in network capacity.  Any future RTP
   congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.

   The session establishment signalling will also necessarily establish
   boundaries to which the media bit-rate will conform.  The choice of
   media codecs provides upper- and lower-bounds on the supported bit-
   rates that the application can utilise to provide useful quality, and
   the packetization choices that exist.  In addition, the signalling
   channel can establish maximum media bit-rate boundaries using the SDP
   "b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media
   Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo).
   The combination of media codec choice and signalled bandwidth limits
   SHOULD be used to limit traffic based on known bandwidth limitations,
   for example the capacity of the edge links, to the extent possible.

7.2.  RTCP Limitations for Congestion Control

   Experience with the congestion control algorithms of TCP [RFC5681],
   TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
   that feedback on packet arrivals needs to be sent frequently (roughly
   once per round trip time is common).  We note that the real-time
   media traffic might not be able to adapt to changing path conditions
   as rapidly as elastic applications using TCP, but frequent feedback,
   perhaps on the order of once per video frame, is still needed to
   allow the congestion control algorithm to track the path dynamics.

   As an example of the type of RTCP congestion control feedback that is
   possible, consider one of the simplest scenarios for WebRTC: a point
   to point video call between two end systems.  There will be four RTP
   flows in this scenario, two audio and two video, with all four flows
   being active for essentially all the time (the audio flows will
   likely use voice activity detection and comfort noise to reduce the
   packet rate during silent periods, but doesn't cause transmissions to
   stop).  Assume all four flows are sent in a single RTP session, each
   using a separate SSRC.  Further, assume each SSRC sends RTCP reports
   for all other SSRCs in the session (i.e., the optimisations in
   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] are not used, giving



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   the worst case for the RTCP overhead).  When all members are senders
   like this, the RTCP timing rules in Sections 6.2 and 6.3 of [RFC3550]
   and [RFC4585] reduce to:

               rtcp_interval = avg_rtcp_size * n / rtcp_bw

   where avg_rtcp_size is measured in octets, and the rtcp_bw is the
   bandwidth available for RTCP.  The average RTCP size will depend on
   the amount of feedback that is sent in each RTCP packet, on the
   number of members in the session, and on the size of source
   description (RTCP SDES) information sent.  As a baseline, each RTCP
   packet will be a compound RTCP packet that contains an RTCP SR and an
   RTCP SDES packet.  In the scenario above, each RTCP SR packet will
   contain three report blocks, once for each of the other RTP SSRCs
   sending data, for a total of 100 octets (this is 8 octets header, 20
   octets sender info, and 3 * 24 octets report blocks).  The RTCP SDES
   packet will comprise a header (4 octets), an originating SSRC (4
   octets), a CNAME chunk, and padding.  If the CNAME follows [RFC7022]
   and it will be 19 octets in size, and require 1 octet of padding.
   The resulting compound RTCP packet will be 128 octets in size.  If
   sent in UDP/IPv4 with no IP options and using Secure RTP, which adds
   20 (IPv4) + 8 (UDP) + 14 (SRTP with 80 bit Authentication tag), the
   avg_rtcp_size will therefore be 170 octets, including the header
   overhead.  The value n is this scenario is 4, and the rtcp_bw is
   assumed to be 5% of the session bandwidth.

   If it is desired to send RTCP feedback packets on average 30 times
   per second, to correspond to one RTCP report every frame for 30fps
   video, we can invert the above rtcp_interval calculation to get an
   rtcp_bw that gives an interval of 1/30th of a second or lower.  This
   corresponds to an rtcp_bw of 20400 octets per second (since 1/30 =
   170 * 4 / 20400).  This is 163200 bits per second, which if 5% of the
   session bandwidth, gives a session bandwidth of approximately 3.3Mbps
   (i.e., 3.3Mbps media rate, plus an additional 5% for RTCP, to give a
   total data rate of approximately 3.4Mbps).  That is, RTCP can report
   on every frame of video provided the session bandwidth is 3.3Mbps or
   larger, when every SSRC sends a report for every video frame.  Please
   note that the actual RTCP transmission intervals will be within the
   interval [0.0135, 0.0406]s, but maintaining an average RTCP
   transmission interval of 0.033s.

      Note: To achieve the RTCP transmission intervals above the RTP/
      SAVPF profile with T_rr_interval=0 is used, since even when using
      the reduced minimal transmission interval, the RTP/SAVP profile
      would only allow sending RTCP at most every 0.11s (every third
      frame of video).  Using RTP/SAVPF with T_rr_interval=0 however is
      capable of fully utilizing the configured 5% RTCP bandwidth
      fraction.



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   If additional feedback beyond the standard report block is needed,
   the session bandwidth needed will increase.  For example, with an
   additional 20 octets data being reported in each RTCP packet, the
   session bandwidth needed increases to 3.5Mbps for every SSRC to be
   able to report on every frame.  However, the above baseline might not
   be the most appropriate usage of the RTCP bandwidth.  Depending on
   needs, a less frequent usage of regular RTCP compound packets,
   controlled by T_rr_interval combined with using the reduced size RTCP
   packets, can achieve more frequent and useful reporting.  Also the
   reporting requirements defined in
   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] will reduced the
   amount of bandwidth consumed for reporting when each endpoint has
   multiple SSRCs.

   Calculations such as these show that RTCP cannot be used to send per-
   packet congestion feedback.  RTCP can, however, be used to send
   congestion feedback on each frame of video sent in an interactive
   video conferencing scenario, provided the RTCP parameters are
   correctly configured and the overall session bandwidth exceeds a
   couple of megabits per second (the exact rate depending on the number
   of session participants, the RTCP bandwidth fraction, and whether
   audio and video are sent in one or two RTP sessions).  Using similar
   calculations, it can be shown that RTCP can likely also be used to
   send feedback on a per-RTT basis, provided the RTT is not too low.

   Interactive communication might not be able to afford to wait for
   packet losses to occur to indicate congestion, because an increase in
   play out delay due to queuing (most prominent in wireless networks)
   can easily lead to packets being dropped due to late arrival at the
   receiver.  Therefore, more sophisticated cues might need to be
   reported -- to be defined in a suitable congestion control framework
   as noted above -- which, in turn, increase the report size again.
   For example, different RTCP XR report blocks (jointly) provide the
   necessary details to implement a variety of congestion control
   algorithms, but the (compound) report size grows quickly.

7.3.  Congestion Control Interoperability and Legacy Systems

   There are legacy RTP implementations that do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC implementations
   that need to interwork with such end-points MUST limit their
   transmission to a low rate, equivalent to a VoIP call using a low
   bandwidth codec, that is unlikely to cause any significant
   congestion.

   When interworking with legacy implementations that support RTCP using
   the RTP/AVP profile [RFC3551], congestion feedback is provided in



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   RTCP RR packets every few seconds.  Implementations that have to
   interwork with such end-points MUST ensure that they keep within the
   RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers]
   constraints to limit the congestion they can cause.

   If a legacy end-point supports RTP/AVPF, this enables negotiation of
   important parameters for frequent reporting, such as the "trr-int"
   parameter, and the possibility that the end-point supports some
   useful feedback format for congestion control purpose such as TMMBR
   [RFC5104].  Implementations that have to interwork with such end-
   points MUST ensure that they stay within the RTP circuit breaker
   [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the
   congestion they can cause, but might find that they can achieve
   better congestion response depending on the amount of feedback that
   is available.

   With proprietary congestion control algorithms issues can arise when
   different algorithms and implementations interact in a communication
   session.  If the different implementations have made different
   choices in regards to the type of adaptation, for example one sender
   based, and one receiver based, then one could end up in situation
   where one direction is dual controlled, when the other direction is
   not controlled.  This memo cannot mandate behaviour for proprietary
   congestion control algorithms, but implementations that use such
   algorithms ought to be aware of this issue, and try to ensure that
   both effective congestion control is negotiated for media flowing in
   both directions.  If the IETF were to standardise both sender- and
   receiver-based congestion control algorithms for WebRTC traffic in
   the future, the issues of interoperability, control, and ensuring
   that both directions of media flow are congestion controlled would
   also need to be considered.

8.  WebRTC Use of RTP: Performance Monitoring

   As described in Section 4.1, implementations are REQUIRED to generate
   RTCP Sender Report (SR) and Reception Report (RR) packets relating to
   the RTP packet streams they send and receive.  These RTCP reports can
   be used for performance monitoring purposes, since they include basic
   packet loss and jitter statistics.

   A large number of additional performance metrics are supported by the
   RTCP Extended Reports (XR) framework [RFC3611][RFC6792].  At the time
   of this writing, it is not clear what extended metrics are suitable
   for use in the WebRTC context, so there is no requirement that
   implementations generate RTCP XR packets.  However, implementations
   that can use detailed performance monitoring data MAY generate RTCP
   XR packets as appropriate; the use of such packets SHOULD be
   signalled in advance.



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   All WebRTC implementations MUST be prepared to receive RTP XR report
   packets, whether or not they were signalled.  There is no requirement
   that the data contained in such reports be used, or exposed to the
   Javascript application, however.

9.  WebRTC Use of RTP: Future Extensions

   It is possible that the core set of RTP protocols and RTP extensions
   specified in this memo will prove insufficient for the future needs
   of WebRTC applications.  In this case, future updates to this memo
   MUST be made following the Guidelines for Writers of RTP Payload
   Format Specifications [RFC2736], How to Write an RTP Payload Format
   [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP
   Control Protocol [RFC5968], and SHOULD take into account any future
   guidelines for extending RTP and related protocols that have been
   developed.

   Authors of future extensions are urged to consider the wide range of
   environments in which RTP is used when recommending extensions, since
   extensions that are applicable in some scenarios can be problematic
   in others.  Where possible, the WebRTC framework will adopt RTP
   extensions that are of general utility, to enable easy implementation
   of a gateway to other applications using RTP, rather than adopt
   mechanisms that are narrowly targeted at specific WebRTC use cases.

10.  Signalling Considerations

   RTP is built with the assumption that an external signalling channel
   exists, and can be used to configure RTP sessions and their features.
   The basic configuration of an RTP session consists of the following
   parameters:

   RTP Profile:  The name of the RTP profile to be used in session.  The
      RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
      on basic level, as can their secure variants RTP/SAVP [RFC3711]
      and RTP/SAVPF [RFC5124].  The secure variants of the profiles do
      not directly interoperate with the non-secure variants, due to the
      presence of additional header fields for authentication in SRTP
      packets and cryptographic transformation of the payload.  WebRTC
      requires the use of the RTP/SAVPF profile, and this MUST be
      signalled if SDP is used.  Interworking functions might transform
      this into the RTP/SAVP profile for a legacy use case, by
      indicating to the WebRTC end-point that the RTP/SAVPF is used, and
      limiting the usage of the "a=rtcp-fb:" attribute to indicate a
      trr-int value of 4 seconds.

   Transport Information:  Source and destination IP address(s) and
      ports for RTP and RTCP MUST be signalled for each RTP session.  In



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      WebRTC these transport addresses will be provided by ICE that
      signals candidates and arrives at nominated candidate address
      pairs.  If RTP and RTCP multiplexing [RFC5761] is to be used, such
      that a single port, i.e. transport-layer flow, is used for RTP and
      RTCP flows, this MUST be signalled (see Section 4.5).

   RTP Payload Types, media formats, and format parameters:  The mapping
      between media type names (and hence the RTP payload formats to be
      used), and the RTP payload type numbers MUST be signalled.  Each
      media type MAY also have a number of media type parameters that
      MUST also be signalled to configure the codec and RTP payload
      format (the "a=fmtp:" line from SDP).  Section 4.3 of this memo
      discusses requirements for uniqueness of payload types.

   RTP Extensions:  The RTP extensions to be used SHOULD be agreed upon,
      including any parameters for each respective extension.  At the
      very least, this will help avoiding using bandwidth for features
      that the other end-point will ignore.  But for certain mechanisms
      there is requirement for this to happen as interoperability
      failure otherwise happens.

   RTCP Bandwidth:  Support for exchanging RTCP Bandwidth values to the
      end-points will be necessary.  This SHALL be done as described in
      "Session Description Protocol (SDP) Bandwidth Modifiers for RTP
      Control Protocol (RTCP) Bandwidth" [RFC3556], or something
      semantically equivalent.  This also ensures that the end-points
      have a common view of the RTCP bandwidth, this is important as too
      different view of the bandwidths can lead to failure to
      interoperate.

   These parameters are often expressed in SDP messages conveyed within
   an offer/answer exchange.  RTP does not depend on SDP or on the offer
   /answer model, but does require all the necessary parameters to be
   agreed upon, and provided to the RTP implementation.  We note that in
   the WebRTC context it will depend on the signalling model and API how
   these parameters need to be configured but they will be need to
   either set in the API or explicitly signalled between the peers.

11.  WebRTC API Considerations

   The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and
   Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses
   the concept of a MediaStream that consists of zero or more
   MediaStreamTracks.  A MediaStreamTrack is an individual stream of
   media from any type of media source like a microphone or a camera,
   but also conceptual sources, like a audio mix or a video composition,
   are possible.  The MediaStreamTracks within a MediaStream need to be
   possible to play out synchronised.



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   A MediaStreamTrack's realisation in RTP in the context of an
   RTCPeerConnection consists of a source packet stream identified with
   an SSRC within an RTP session part of the RTCPeerConnection.  The
   MediaStreamTrack can also result in additional packet streams, and
   thus SSRCs, in the same RTP session.  These can be dependent packet
   streams from scalable encoding of the source stream associated with
   the MediaStreamTrack, if such a media encoder is used.  They can also
   be redundancy packet streams, these are created when applying Forward
   Error Correction (Section 6.2) or RTP retransmission (Section 6.1) to
   the source packet stream.

   It is important to note that the same media source can be feeding
   multiple MediaStreamTracks.  As different sets of constraints or
   other parameters can be applied to the MediaStreamTrack, each
   MediaStreamTrack instance added to a RTCPeerConnection SHALL result
   in an independent source packet stream, with its own set of
   associated packet streams, and thus different SSRC(s).  It will
   depend on applied constraints and parameters if the source stream and
   the encoding configuration will be identical between different
   MediaStreamTracks sharing the same media source.  Thus it is possible
   for multiple source packet streams to share encoded streams (but not
   packet streams), but this is an implementation choice to try to
   utilise such optimisations.  Note that such optimizations would need
   to take into account that the constraints for one of the
   MediaStreamTracks can at any moment change, meaning that the encoding
   configurations might no longer be identical.

   The same MediaStreamTrack can also be included in multiple
   MediaStreams, thus multiple sets of MediaStreams can implicitly need
   to use the same synchronisation base.  To ensure that this works in
   all cases, and don't forces a end-point to change synchronisation
   base and CNAME in the middle of a ongoing delivery of any packet
   streams, which would cause media disruption; all MediaStreamTracks
   and their associated SSRCs originating from the same end-point needs
   to be sent using the same CNAME within one RTCPeerConnection.  This
   is motivating the strong recommendation in Section 4.9 to only use a
   single CNAME.

      The requirement on using the same CNAME for all SSRCs that
      originates from the same end-point, does not require middleboxes
      that forwards traffic from multiple end-points to only use a
      single CNAME.

   Different CNAMEs normally need to be used for different
   RTCPeerConnection instances, as specified in Section 4.9.  Having two
   communication sessions with the same CNAME could enable tracking of a
   user or device across different services (see Section 4.4.1 of
   [I-D.ietf-rtcweb-security] for details).  A web application can



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   request that the CNAMEs used in different RTCPeerConnection within a
   same-orign context to be the same, this allow for synchronization of
   the endpoint's RTP packet streams across the different
   RTCPeerConnections.

      Note: this doesn't result in a tracking issue, since the creation
      of matching CNAMEs depends on existing tracking.

   The above will currently force a WebRTC end-point that receives an
   MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing
   on any RTCPeerConnection to perform resynchronisation of the stream.
   This, as the sending party needs to change the CNAME, which implies
   that it has to use a locally available system clock as timebase for
   the synchronisation.  Thus, the relative relation between the
   timebase of the incoming stream and the system sending out needs to
   defined.  This relation also needs monitoring for clock drift and
   likely adjustments of the synchronisation.  The sending entity is
   also responsible for congestion control for its the sent streams.  In
   cases of packet loss the loss of incoming data also needs to be
   handled.  This leads to the observation that the method that is least
   likely to cause issues or interruptions in the outgoing source packet
   stream is a model of full decoding, including repair etc followed by
   encoding of the media again into the outgoing packet stream.
   Optimisations of this method is clearly possible and implementation
   specific.

   A WebRTC end-point MUST support receiving multiple MediaStreamTracks,
   where each of different MediaStreamTracks (and their sets of
   associated packet streams) uses different CNAMEs.  However,
   MediaStreamTracks that are received with different CNAMEs have no
   defined synchronisation.

      Note: The motivation for supporting reception of multiple CNAMEs
      are to allow for forward compatibility with any future changes
      that enables more efficient stream handling when end-points relay/
      forward streams.  It also ensures that end-points can interoperate
      with certain types of multi-stream middleboxes or end-points that
      are not WebRTC.

   The binding between the WebRTC MediaStreams, MediaStreamTracks and
   the SSRC is done as specified in "Cross Session Stream Identification
   in the Session Description Protocol" [I-D.ietf-mmusic-msid].  This
   document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to
   map unknown source packet stream SSRCs to MediaStreamTracks and
   MediaStreams.  Commonly the RTP Payload Type of any incoming packets
   will reveal if the packet stream is a source stream or a redundancy
   or dependent packet stream.  The association to the correct source




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   packet stream depends on the payload format in use for the packet
   stream.

   Finally this specification puts a requirement on the WebRTC API to
   realize a method for determining the CSRC list (Section 4.1) as well
   as the Mixer-to-Client audio levels (Section 5.2.3) (when supported)
   and the basic requirements for this is further discussed in
   Section 12.2.1.

12.  RTP Implementation Considerations

   The following discussion provides some guidance on the implementation
   of the RTP features described in this memo.  The focus is on a WebRTC
   end-point implementation perspective, and while some mention is made
   of the behaviour of middleboxes, that is not the focus of this memo.

12.1.  Configuration and Use of RTP Sessions

   A WebRTC end-point will be a simultaneous participant in one or more
   RTP sessions.  Each RTP session can convey multiple media sources,
   and can include media data from multiple end-points.  In the
   following, we outline some ways in which WebRTC end-points can
   configure and use RTP sessions.

12.1.1.  Use of Multiple Media Sources Within an RTP Session

   RTP is a group communication protocol, and every RTP session can
   potentially contain multiple RTP packet streams.  There are several
   reasons why this might be desirable:

   Multiple media types:  Outside of WebRTC, it is common to use one RTP
      session for each type of media sources (e.g., one RTP session for
      audio sources and one for video sources, each sent over different
      transport layer flows).  However, to reduce the number of UDP
      ports used, the default in WebRTC is to send all types of media in
      a single RTP session, as described in Section 4.4, using RTP and
      RTCP multiplexing (Section 4.5) to further reduce the number of
      UDP ports needed.  This RTP session then uses only one bi-
      directional transport-layer flow, but will contain multiple RTP
      packet streams, each containing a different type of media.  A
      common example might be an end-point with a camera and microphone
      that sends two RTP packet streams, one video and one audio, into a
      single RTP session.

   Multiple Capture Devices:  A WebRTC end-point might have multiple
      cameras, microphones, or other media capture devices, and so might
      want to generate several RTP packet streams of the same media
      type.  Alternatively, it might want to send media from a single



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      capture device in several different formats or quality settings at
      once.  Both can result in a single end-point sending multiple RTP
      packet streams of the same media type into a single RTP session at
      the same time.

   Associated Repair Data:  An end-point might send a RTP packet stream
      that is somehow associated with another stream.  For example, it
      might send an RTP packet stream that contains FEC or
      retransmission data relating to another stream.  Some RTP payload
      formats send this sort of associated repair data as part of the
      source packet stream, while others send it as a separate packet
      stream.

   Layered or Multiple Description Coding:  An end-point can use a
      layered media codec, for example H.264 SVC, or a multiple
      description codec, that generates multiple RTP packet streams,
      each with a distinct RTP SSRC, within a single RTP session.

   RTP Mixers, Translators, and Other Middleboxes:  An RTP session, in
      the WebRTC context, is a point-to-point association between an
      end-point and some other peer device, where those devices share a
      common SSRC space.  The peer device might be another WebRTC end-
      point, or it might be an RTP mixer, translator, or some other form
      of media processing middlebox.  In the latter cases, the middlebox
      might send mixed or relayed RTP streams from several participants,
      that the WebRTC end-point will need to render.  Thus, even though
      a WebRTC end-point might only be a member of a single RTP session,
      the peer device might be extending that RTP session to incorporate
      other end-points.  WebRTC is a group communication environment and
      end-points need to be capable of receiving, decoding, and playing
      out multiple RTP packet streams at once, even in a single RTP
      session.

12.1.2.  Use of Multiple RTP Sessions

   In addition to sending and receiving multiple RTP packet streams
   within a single RTP session, a WebRTC end-point might participate in
   multiple RTP sessions.  There are several reasons why a WebRTC end-
   point might choose to do this:

   To interoperate with legacy devices:  The common practice in the non-
      WebRTC world is to send different types of media in separate RTP
      sessions, for example using one RTP session for audio and another
      RTP session, on a separate transport layer flow, for video.  All
      WebRTC end-points need to support the option of sending different
      types of media on different RTP sessions, so they can interwork
      with such legacy devices.  This is discussed further in
      Section 4.4.



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   To provide enhanced quality of service:  Some network-based quality
      of service mechanisms operate on the granularity of transport
      layer flows.  If it is desired to use these mechanisms to provide
      differentiated quality of service for some RTP packet streams,
      then those RTP packet streams need to be sent in a separate RTP
      session using a different transport-layer flow, and with
      appropriate quality of service marking.  This is discussed further
      in Section 12.1.3.

   To separate media with different purposes:  An end-point might want
      to send RTP packet streams that have different purposes on
      different RTP sessions, to make it easy for the peer device to
      distinguish them.  For example, some centralised multiparty
      conferencing systems display the active speaker in high
      resolution, but show low resolution "thumbnails" of other
      participants.  Such systems might configure the end-points to send
      simulcast high- and low-resolution versions of their video using
      separate RTP sessions, to simplify the operation of the RTP
      middlebox.  In the WebRTC context this is currently possible to
      accomplished by establishing multiple WebRTC MediaStreamTracks
      that have the same media source in one (or more)
      RTCPeerConnection.  Each MediaStreamTrack is then configured to
      deliver a particular media quality and thus media bit-rate, and
      will produce an independently encoded version with the codec
      parameters agreed specifically in the context of that
      RTCPeerConnection.  The RTP middlebox can distinguish packets
      corresponding to the low- and high-resolution streams by
      inspecting their SSRC, RTP payload type, or some other information
      contained in RTP payload, RTP header extension or RTCP packets,
      but it can be easier to distinguish the RTP packet streams if they
      arrive on separate RTP sessions on separate transport-layer flows.

   To directly connect with multiple peers:  A multi-party conference
      does not need to use an RTP middlebox.  Rather, a multi-unicast
      mesh can be created, comprising several distinct RTP sessions,
      with each participant sending RTP traffic over a separate RTP
      session (that is, using an independent RTCPeerConnection object)
      to every other participant, as shown in Figure 1.  This topology
      has the benefit of not requiring an RTP middlebox node that is
      trusted to access and manipulate the media data.  The downside is
      that it increases the used bandwidth at each sender by requiring
      one copy of the RTP packet streams for each participant that are
      part of the same session beyond the sender itself.








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   +---+     +---+
   | A |<--->| B |
   +---+     +---+
     ^         ^
      \       /
       \     /
        v   v
        +---+
        | C |
        +---+


            Figure 1: Multi-unicast using several RTP sessions

      The multi-unicast topology could also be implemented as a single
      RTP session, spanning multiple peer-to-peer transport layer
      connections, or as several pairwise RTP sessions, one between each
      pair of peers.  To maintain a coherent mapping between the
      relation between RTP sessions and RTCPeerConnection objects we
      recommend that this is implemented as several individual RTP
      sessions.  The only downside is that end-point A will not learn of
      the quality of any transmission happening between B and C, since
      it will not see RTCP reports for the RTP session between B and C,
      whereas it would it all three participants were part of a single
      RTP session.  Experience with the Mbone tools (experimental RTP-
      based multicast conferencing tools from the late 1990s) has showed
      that RTCP reception quality reports for third parties can usefully
      be presented to the users in a way that helps them understand
      asymmetric network problems, and the approach of using separate
      RTP sessions prevents this.  However, an advantage of using
      separate RTP sessions is that it enables using different media
      bit-rates and RTP session configurations between the different
      peers, thus not forcing B to endure the same quality reductions if
      there are limitations in the transport from A to C as C will.  It
      it believed that these advantages outweigh the limitations in
      debugging power.

   To indirectly connect with multiple peers:  A common scenario in
      multi-party conferencing is to create indirect connections to
      multiple peers, using an RTP mixer, translator, or some other type
      of RTP middlebox.  Figure 2 outlines a simple topology that might
      be used in a four-person centralised conference.  The middlebox
      acts to optimise the transmission of RTP packet streams from
      certain perspectives, either by only sending some of the received
      RTP packet stream to any given receiver, or by providing a
      combined RTP packet stream out of a set of contributing streams.





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   +---+      +-------------+      +---+
   | A |<---->|             |<---->| B |
   +---+      | RTP mixer,  |      +---+
              | translator, |
              | or other    |
   +---+      | middlebox   |      +---+
   | C |<---->|             |<---->| D |
   +---+      +-------------+      +---+


                Figure 2: RTP mixer with only unicast paths

      There are various methods of implementation for the middlebox.  If
      implemented as a standard RTP mixer or translator, a single RTP
      session will extend across the middlebox and encompass all the
      end-points in one multi-party session.  Other types of middlebox
      might use separate RTP sessions between each end-point and the
      middlebox.  A common aspect is that these RTP middleboxes can use
      a number of tools to control the media encoding provided by a
      WebRTC end-point.  This includes functions like requesting
      breaking the encoding chain and have the encoder produce a so
      called Intra frame.  Another is limiting the bit-rate of a given
      stream to better suit the mixer view of the multiple down-streams.
      Others are controlling the most suitable frame-rate, picture
      resolution, the trade-off between frame-rate and spatial quality.
      The middlebox gets the significant responsibility to correctly
      perform congestion control, source identification, manage
      synchronisation while providing the application with suitable
      media optimizations.  The middlebox is also has to be a trusted
      node when it comes to security, since it manipulates either the
      RTP header or the media itself (or both) received from one end-
      point, before sending it on towards the end-point(s), thus they
      need to be able to decrypt and then encrypt it before sending it
      out.

      RTP Mixers can create a situation where an end-point experiences a
      situation in-between a session with only two end-points and
      multiple RTP sessions.  Mixers are expected to not forward RTCP
      reports regarding RTP packet streams across themselves.  This is
      due to the difference in the RTP packet streams provided to the
      different end-points.  The original media source lacks information
      about a mixer's manipulations prior to sending it the different
      receivers.  This scenario also results in that an end-point's
      feedback or requests goes to the mixer.  When the mixer can't act
      on this by itself, it is forced to go to the original media source
      to fulfil the receivers request.  This will not necessarily be
      explicitly visible any RTP and RTCP traffic, but the interactions
      and the time to complete them will indicate such dependencies.



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      Providing source authentication in multi-party scenarios is a
      challenge.  In the mixer-based topologies, end-points source
      authentication is based on, firstly, verifying that media comes
      from the mixer by cryptographic verification and, secondly, trust
      in the mixer to correctly identify any source towards the end-
      point.  In RTP sessions where multiple end-points are directly
      visible to an end-point, all end-points will have knowledge about
      each others' master keys, and can thus inject packets claimed to
      come from another end-point in the session.  Any node performing
      relay can perform non-cryptographic mitigation by preventing
      forwarding of packets that have SSRC fields that came from other
      end-points before.  For cryptographic verification of the source
      SRTP would require additional security mechanisms, for example
      TESLA for SRTP [RFC4383], that are not part of the base WebRTC
      standards.

   To forward media between multiple peers:  It is sometimes desirable
      for an end-point that receives an RTP packet stream to be able to
      forward that RTP packet stream to a third party.  The are some
      obvious security and privacy implications in supporting this, but
      also potential uses.  This is supported in the W3C API by taking
      the received and decoded media and using it as media source that
      is re-encoding and transmitted as a new stream.

      At the RTP layer, media forwarding acts as a back-to-back RTP
      receiver and RTP sender.  The receiving side terminates the RTP
      session and decodes the media, while the sender side re-encodes
      and transmits the media using an entirely separate RTP session.
      The original sender will only see a single receiver of the media,
      and will not be able to tell that forwarding is happening based on
      RTP-layer information since the RTP session that is used to send
      the forwarded media is not connected to the RTP session on which
      the media was received by the node doing the forwarding.

      The end-point that is performing the forwarding is responsible for
      producing an RTP packet stream suitable for onwards transmission.
      The outgoing RTP session that is used to send the forwarded media
      is entirely separate to the RTP session on which the media was
      received.  This will require media transcoding for congestion
      control purpose to produce a suitable bit-rate for the outgoing
      RTP session, reducing media quality and forcing the forwarding
      end-point to spend the resource on the transcoding.  The media
      transcoding does result in a separation of the two different legs
      removing almost all dependencies, and allowing the forwarding end-
      point to optimize its media transcoding operation.  The cost is
      greatly increased computational complexity on the forwarding node.
      Receivers of the forwarded stream will see the forwarding device
      as the sender of the stream, and will not be able to tell from the



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      RTP layer that they are receiving a forwarded stream rather than
      an entirely new RTP packet stream generated by the forwarding
      device.

12.1.3.  Differentiated Treatment of RTP Packet Streams

   There are use cases for differentiated treatment of RTP packet
   streams.  Such differentiation can happen at several places in the
   system.  First of all is the prioritization within the end-point
   sending the media, which controls, both which RTP packet streams that
   will be sent, and their allocation of bit-rate out of the current
   available aggregate as determined by the congestion control.

   It is expected that the WebRTC API [W3C.WD-webrtc-20130910] will
   allow the application to indicate relative priorities for different
   MediaStreamTracks.  These priorities can then be used to influence
   the local RTP processing, especially when it comes to congestion
   control response in how to divide the available bandwidth between the
   RTP packet streams.  Any changes in relative priority will also need
   to be considered for RTP packet streams that are associated with the
   main RTP packet streams, such as redundant streams for RTP
   retransmission and FEC.  The importance of such redundant RTP packet
   streams is dependent on the media type and codec used, in regards to
   how robust that codec is to packet loss.  However, a default policy
   might to be to use the same priority for redundant RTP packet stream
   as for the source RTP packet stream.

   Secondly, the network can prioritize transport-layer flows and sub-
   flows, including RTP packet streams.  Typically, differential
   treatment includes two steps, the first being identifying whether an
   IP packet belongs to a class that has to be treated differently, the
   second the actual mechanism to prioritize packets.  This is done
   according to three methods:

   DiffServ:  The end-point marks a packet with a DiffServ code point to
      indicate to the network that the packet belongs to a particular
      class.

   Flow based:  Packets that need to be given a particular treatment are
      identified using a combination of IP and port address.

   Deep Packet Inspection:  A network classifier (DPI) inspects the
      packet and tries to determine if the packet represents a
      particular application and type that is to be prioritized.

   Flow-based differentiation will provide the same treatment to all
   packets within a transport-layer flow, i.e., relative prioritization
   is not possible.  Moreover, if the resources are limited it might not



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   be possible to provide differential treatment compared to best-effort
   for all the RTP packet streams in a WebRTC application.  When flow-
   based differentiation is available the WebRTC application needs to
   know about it so that it can provide the separation of the RTP packet
   streams onto different UDP flows to enable a more granular usage of
   flow based differentiation.  That way at least providing different
   prioritization of audio and video if desired by application.

   DiffServ assumes that either the end-point or a classifier can mark
   the packets with an appropriate DSCP so that the packets are treated
   according to that marking.  If the end-point is to mark the traffic
   two requirements arise in the WebRTC context: 1) The WebRTC
   application or browser has to know which DSCP to use and that it can
   use them on some set of RTP packet streams. 2) The information needs
   to be propagated to the operating system when transmitting the
   packet.  Details of this process are outside the scope of this memo
   and are further discussed in "DSCP and other packet markings for
   RTCWeb QoS" [I-D.ietf-tsvwg-rtcweb-qos].

   For packet based marking schemes it might be possible to mark
   individual RTP packets differently based on the relative priority of
   the RTP payload.  For example video codecs that have I, P, and B
   pictures could prioritise any payloads carrying only B frames less,
   as these are less damaging to loose.  However, depending on the QoS
   mechanism and what markings that are applied, this can result in not
   only different packet drop probabilities but also packet reordering,
   see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion.  As default
   policy all RTP packets related to a RTP packet stream ought to be
   provided with the same prioritization; per-packet prioritization is
   outside the scope of this memo, but might be specified elsewhere in
   future.

   It is also important to consider how RTCP packets associated with a
   particular RTP packet stream need to be marked.  RTCP compound
   packets with Sender Reports (SR), ought to be marked with the same
   priority as the RTP packet stream itself, so the RTCP-based round-
   trip time (RTT) measurements are done using the same transport-layer
   flow priority as the RTP packet stream experiences.  RTCP compound
   packets containing RR packet ought to be sent with the priority used
   by the majority of the RTP packet streams reported on.  RTCP packets
   containing time-critical feedback packets can use higher priority to
   improve the timeliness and likelihood of delivery of such feedback.

12.2.  Media Source, RTP Packet Streams, and Participant Identification







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12.2.1.  Media Source

   Each RTP packet stream is identified by a unique synchronisation
   source (SSRC) identifier.  The SSRC identifier is carried in each of
   the RTP packets comprising a RTP packet stream, and is also used to
   identify that stream in the corresponding RTCP reports.  The SSRC is
   chosen as discussed in Section 4.8.  The first stage in
   demultiplexing RTP and RTCP packets received on a single transport
   layer flow at a WebRTC end-point is to separate the RTP packet
   streams based on their SSRC value; once that is done, additional
   demultiplexing steps can determine how and where to render the media.

   RTP allows a mixer, or other RTP-layer middlebox, to combine encoded
   streams from multiple media sources to form a new encoded stream from
   a new media source (the mixer).  The RTP packets in that new RTP
   packet stream can include a Contributing Source (CSRC) list,
   indicating which original SSRCs contributed to the combined source
   stream.  As described in Section 4.1, implementations need to support
   reception of RTP data packets containing a CSRC list and RTCP packets
   that relate to sources present in the CSRC list.  The CSRC list can
   change on a packet-by-packet basis, depending on the mixing operation
   being performed.  Knowledge of what media sources contributed to a
   particular RTP packet can be important if the user interface
   indicates which participants are active in the session.  Changes in
   the CSRC list included in packets needs to be exposed to the WebRTC
   application using some API, if the application is to be able to track
   changes in session participation.  It is desirable to map CSRC values
   back into WebRTC MediaStream identities as they cross this API, to
   avoid exposing the SSRC/CSRC name space to JavaScript applications.

   If the mixer-to-client audio level extension [RFC6465] is being used
   in the session (see Section 5.2.3), the information in the CSRC list
   is augmented by audio level information for each contributing source.
   This information can usefully be exposed in the user interface.

12.2.2.  SSRC Collision Detection

   The RTP standard [RFC3550] requires any RTP implementation to have
   support for detecting and handling SSRC collisions, i.e., resolve the
   conflict when two different end-points use the same SSRC value.  This
   requirement also applies to WebRTC end-points.  There are several
   scenarios where SSRC collisions can occur:

   o  In a point-to-point session where each SSRC is associated with
      either of the two end-points and where the main media carrying
      SSRC identifier will be announced in the signalling channel, a
      collision is less likely to occur due to the information about
      used SSRCs provided by Source-Specific SDP Attributes [RFC5576].



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      Still, collisions can occur if both end-points start uses an new
      SSRC identifier prior to having signalled it to the peer and
      received acknowledgement on the signalling message.  The Source-
      Specific SDP Attributes [RFC5576] contains no mechanism to resolve
      SSRC collisions or reject a end-points usage of an SSRC.

   o  SSRC values that have not been signalled could also appear in an
      RTP session.  This is more likely than it appears, since some RTP
      functions use extra SSRCs to provide their functionality.  For
      example, retransmission data might be transmitted using a separate
      RTP packet stream that requires its own SSRC, separate to the SSRC
      of the source RTP packet stream [RFC4588].  In those cases, an
      end-point can create a new SSRC that strictly doesn't need to be
      announced over the signalling channel to function correctly on
      both RTP and RTCPeerConnection level.

   o  Multiple end-points in a multiparty conference can create new
      sources and signal those towards the RTP middlebox.  In cases
      where the SSRC/CSRC are propagated between the different end-
      points from the RTP middlebox collisions can occur.

   o  An RTP middlebox could connect an end-point's RTCPeerConnection to
      another RTCPeerConnection from the same end-point, thus forming a
      loop where the end-point will receive its own traffic.  While is
      is clearly considered a bug, it is important that the end-point is
      able to recognise and handle the case when it occurs.  This case
      becomes even more problematic when media mixers, and so on, are
      involved, where the stream received is a different stream but
      still contains this client's input.

   These SSRC/CSRC collisions can only be handled on RTP level as long
   as the same RTP session is extended across multiple
   RTCPeerConnections by a RTP middlebox.  To resolve the more generic
   case where multiple RTCPeerConnections are interconnected, then
   identification of the media source(s) part of a MediaStreamTrack
   being propagated across multiple interconnected RTCPeerConnection
   needs to be preserved across these interconnections.

12.2.3.  Media Synchronisation Context

   When an end-point sends media from more than one media source, it
   needs to consider if (and which of) these media sources are to be
   synchronized.  In RTP/RTCP, synchronisation is provided by having a
   set of RTP packet streams be indicated as coming from the same
   synchronisation context and logical end-point by using the same RTCP
   CNAME identifier.





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   The next provision is that the internal clocks of all media sources,
   i.e., what drives the RTP timestamp, can be correlated to a system
   clock that is provided in RTCP Sender Reports encoded in an NTP
   format.  By correlating all RTP timestamps to a common system clock
   for all sources, the timing relation of the different RTP packet
   streams, also across multiple RTP sessions can be derived at the
   receiver and, if desired, the streams can be synchronized.  The
   requirement is for the media sender to provide the correlation
   information; it is up to the receiver to use it or not.

13.  Security Considerations

   The overall security architecture for WebRTC is described in
   [I-D.ietf-rtcweb-security-arch], and security considerations for the
   WebRTC framework are described in [I-D.ietf-rtcweb-security].  These
   considerations also apply to this memo.

   The security considerations of the RTP specification, the RTP/SAVPF
   profile, and the various RTP/RTCP extensions and RTP payload formats
   that form the complete protocol suite described in this memo apply.
   We do not believe there are any new security considerations resulting
   from the combination of these various protocol extensions.

   The Extended Secure RTP Profile for Real-time Transport Control
   Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
   handling of fundamental issues by offering confidentiality, integrity
   and partial source authentication.  A mandatory to implement media
   security solution is created by combing this secured RTP profile and
   DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of
   [I-D.ietf-rtcweb-security-arch].

   RTCP packets convey a Canonical Name (CNAME) identifier that is used
   to associate RTP packet streams that need to be synchronised across
   related RTP sessions.  Inappropriate choice of CNAME values can be a
   privacy concern, since long-term persistent CNAME identifiers can be
   used to track users across multiple WebRTC calls.  Section 4.9 of
   this memo provides guidelines for generation of untraceable CNAME
   values that alleviate this risk.

   The guidelines in [RFC6562] apply when using variable bit rate (VBR)
   audio codecs such as Opus (see Section 4.3 for discussion of mandated
   audio codecs).  The guidelines in [RFC6562] also apply, but are of
   lesser importance, when using the client-to-mixer audio level header
   extensions (Section 5.2.2) or the mixer-to-client audio level header
   extensions (Section 5.2.3).  The use of the encryption of the header
   extensions are RECOMMENDED, unless there are known reasons, like RTP
   middleboxes or third party monitoring that will greatly benefit from
   the information, and this has been expressed using API or signalling.



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   If further evidence are produced to show that information leakage is
   significant from audio level indications, then use of encryption
   needs to be mandated at that time.

14.  IANA Considerations

   This memo makes no request of IANA.

   Note to RFC Editor: this section is to be removed on publication as
   an RFC.

15.  Acknowledgements

   The authors would like to thank Bernard Aboba, Harald Alvestrand,
   Cary Bran, Charles Eckel, Christian Groves, Cullen Jennings, Dan
   Romascanu, Martin Thomson, and the other members of the IETF RTCWEB
   working group for their valuable feedback.

16.  References

16.1.  Normative References

   [I-D.ietf-avtcore-multi-media-rtp-session]
              Westerlund, M., Perkins, C., and J. Lennox, "Sending
              Multiple Types of Media in a Single RTP Session", draft-
              ietf-avtcore-multi-media-rtp-session-05 (work in
              progress), February 2014.

   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", draft-ietf-
              avtcore-rtp-circuit-breakers-05 (work in progress),
              February 2014.

   [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session:
              Grouping RTCP Reception Statistics and Other Feedback",
              draft-ietf-avtcore-rtp-multi-stream-optimisation-02 (work
              in progress), February 2014.

   [I-D.ietf-avtcore-rtp-multi-stream]
              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-03 (work in progress),
              February 2014.





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   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-09 (work in progress), February 2014.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-06 (work in progress), January 2014.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC2736]  Handley, M. and C. Perkins, "Guidelines for Writers of RTP
              Payload Format Specifications", BCP 36, RFC 2736, December
              1999.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
              3556, July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
              2006.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, July 2007.






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   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, February 2008.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.

   [RFC6051]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", RFC 6051, November 2010.

   [RFC6464]  Lennox, J., Ivov, E., and E. Marocco, "A Real-time
              Transport Protocol (RTP) Header Extension for Client-to-
              Mixer Audio Level Indication", RFC 6464, December 2011.

   [RFC6465]  Ivov, E., Marocco, E., and J. Lennox, "A Real-time
              Transport Protocol (RTP) Header Extension for Mixer-to-
              Client Audio Level Indication", RFC 6465, December 2011.

   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of
              Variable Bit Rate Audio with Secure RTP", RFC 6562, March
              2012.

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", RFC 6904, April
              2013.

   [RFC7007]  Terriberry, T., "Update to Remove DVI4 from the
              Recommended Codecs for the RTP Profile for Audio and Video
              Conferences with Minimal Control (RTP/AVP)", RFC 7007,
              August 2013.






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   [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", RFC 7022, September 2013.

   [RFC7160]  Petit-Huguenin, M. and G. Zorn, "Support for Multiple
              Clock Rates in an RTP Session", RFC 7160, April 2014.

   [RFC7164]  Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC
              7164, March 2014.

   [W3C.WD-mediacapture-streams-20130903]
              Burnett, D., Bergkvist, A., Jennings, C., and A.
              Narayanan, "Media Capture and Streams", World Wide Web
              Consortium WD WD-mediacapture-streams-20130903, September
              2013, <http://www.w3.org/TR/2013/
              WD-mediacapture-streams-20130903>.

   [W3C.WD-webrtc-20130910]
              Bergkvist, A., Burnett, D., Jennings, C., and A.
              Narayanan, "WebRTC 1.0: Real-time Communication Between
              Browsers", World Wide Web Consortium WD WD-
              webrtc-20130910, September 2013,
              <http://www.w3.org/TR/2013/WD-webrtc-20130910>.

16.2.  Informative References

   [I-D.ietf-avtcore-multiplex-guidelines]
              Westerlund, M., Perkins, C., and H. Alvestrand,
              "Guidelines for using the Multiplexing Features of RTP to
              Support Multiple Media Streams", draft-ietf-avtcore-
              multiplex-guidelines-02 (work in progress), January 2014.

   [I-D.ietf-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              ietf-avtcore-rtp-topologies-update-01 (work in progress),
              October 2013.

   [I-D.ietf-avtext-rtp-grouping-taxonomy]
              Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,
              "A Taxonomy of Grouping Semantics and Mechanisms for Real-
              Time Transport Protocol (RTP) Sources", draft-ietf-avtext-
              rtp-grouping-taxonomy-01 (work in progress), February
              2014.

   [I-D.ietf-mmusic-msid]
              Alvestrand, H., "WebRTC MediaStream Identification in the
              Session Description Protocol", draft-ietf-mmusic-msid-05
              (work in progress), March 2014.



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   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-07 (work in progress), April 2014.

   [I-D.ietf-payload-rtp-howto]
              Westerlund, M., "How to Write an RTP Payload Format",
              draft-ietf-payload-rtp-howto-13 (work in progress),
              January 2014.

   [I-D.ietf-rmcat-cc-requirements]
              Jesup, R., "Congestion Control Requirements For RMCAT",
              draft-ietf-rmcat-cc-requirements-04 (work in progress),
              April 2014.

   [I-D.ietf-rtcweb-audio]
              Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", draft-ietf-rtcweb-audio-05 (work in
              progress), February 2014.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-09 (work
              in progress), February 2014.

   [I-D.ietf-rtcweb-use-cases-and-requirements]
              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements", draft-
              ietf-rtcweb-use-cases-and-requirements-14 (work in
              progress), February 2014.

   [I-D.ietf-tsvwg-rtcweb-qos]
              Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
              other packet markings for RTCWeb QoS", draft-ietf-tsvwg-
              rtcweb-qos-00 (work in progress), April 2014.

   [RFC3611]  Friedman, T., Caceres, R., and A. Clark, "RTP Control
              Protocol Extended Reports (RTCP XR)", RFC 3611, November
              2003.

   [RFC4341]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion Control ID 2: TCP-like
              Congestion Control", RFC 4341, March 2006.







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   [RFC4342]  Floyd, S., Kohler, E., and J. Padhye, "Profile for
              Datagram Congestion Control Protocol (DCCP) Congestion
              Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
              March 2006.

   [RFC4383]  Baugher, M. and E. Carrara, "The Use of Timed Efficient
              Stream Loss-Tolerant Authentication (TESLA) in the Secure
              Real-time Transport Protocol (SRTP)", RFC 4383, February
              2006.

   [RFC4828]  Floyd, S. and E. Kohler, "TCP Friendly Rate Control
              (TFRC): The Small-Packet (SP) Variant", RFC 4828, April
              2007.

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification", RFC
              5348, September 2008.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, September 2009.

   [RFC5968]  Ott, J. and C. Perkins, "Guidelines for Extending the RTP
              Control Protocol (RTCP)", RFC 5968, September 2010.

   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", RFC 6263, June 2011.

   [RFC6792]  Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the
              RTP Monitoring Framework", RFC 6792, November 2012.

Authors' Addresses

   Colin Perkins
   University of Glasgow
   School of Computing Science
   Glasgow  G12 8QQ
   United Kingdom

   Email: csp@csperkins.org
   URI:   http://csperkins.org/






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Internet-Draft               RTP for WebRTC                   April 2014


   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com


   Joerg Ott
   Aalto University
   School of Electrical Engineering
   Espoo  02150
   Finland

   Email: jorg.ott@aalto.fi


































Perkins, et al.         Expires October 25, 2014               [Page 45]


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