[Docs] [txt|pdf] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits]

Versions: (draft-perkins-rtcweb-rtp-usage) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 Draft is active
In: MissingRef
RTCWEB Working Group                                       C. S. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Standards Track                           M. Westerlund
Expires: January 16, 2014                                       Ericsson
                                                                  J. Ott
                                                        Aalto University
                                                           July 15, 2013

  Web Real-Time Communication (WebRTC): Media Transport and Use of RTP


   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 January 16, 2014.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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

Perkins, et al.         Expires January 16, 2014                [Page 1]

Internet-Draft               RTP for WebRTC                    July 2013

   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 . . . . . . . . . . . . . . . . . . . . . . . . .   5
   4.  WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . .   5
     4.1.  RTP and RTCP  . . . . . . . . . . . . . . . . . . . . . .   6
     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) . . . . . . .  10
     4.9.  Generation of the RTCP Canonical Name (CNAME) . . . . . .  11
   5.  WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . .  12
     5.1.  Conferencing Extensions . . . . . . . . . . . . . . . . .  12
       5.1.1.  Full Intra Request (FIR)  . . . . . . . . . . . . . .  13
       5.1.2.  Picture Loss Indication (PLI) . . . . . . . . . . . .  13
       5.1.3.  Slice Loss Indication (SLI) . . . . . . . . . . . . .  13
       5.1.4.  Reference Picture Selection Indication (RPSI) . . . .  13
       5.1.5.  Temporal-Spatial Trade-off Request (TSTR) . . . . . .  14
       5.1.6.  Temporary Maximum Media Stream Bit Rate Request
               (TMMBR) . . . . . . . . . . . . . . . . . . . . . . .  14
     5.2.  Header Extensions . . . . . . . . . . . . . . . . . . . .  14
       5.2.1.  Rapid Synchronisation . . . . . . . . . . . . . . . .  15
       5.2.2.  Client-to-Mixer Audio Level . . . . . . . . . . . . .  15
       5.2.3.  Mixer-to-Client Audio Level . . . . . . . . . . . . .  15
   6.  WebRTC Use of RTP: Improving Transport Robustness . . . . . .  16
     6.1.  Negative Acknowledgements and RTP Retransmission  . . . .  16
     6.2.  Forward Error Correction (FEC)  . . . . . . . . . . . . .  17
   7.  WebRTC Use of RTP: Rate Control and Media Adaptation  . . . .  17
     7.1.  Boundary Conditions and Circuit Breakers  . . . . . . . .  18
     7.2.  RTCP Limitations for Congestion Control . . . . . . . . .  19
     7.3.  Congestion Control Interoperability and Legacy Systems  .  19
   8.  WebRTC Use of RTP: Performance Monitoring . . . . . . . . . .  20
   9.  WebRTC Use of RTP: Future Extensions  . . . . . . . . . . . .  21
   10. Signalling Considerations . . . . . . . . . . . . . . . . . .  21
   11. WebRTC API Considerations . . . . . . . . . . . . . . . . . .  23
   12. RTP Implementation Considerations . . . . . . . . . . . . . .  23
     12.1.  RTP Sessions and PeerConnections . . . . . . . . . . . .  24
     12.2.  Multiple Sources . . . . . . . . . . . . . . . . . . . .  25

Perkins, et al.         Expires January 16, 2014                [Page 2]

Internet-Draft               RTP for WebRTC                    July 2013

     12.3.  Multiparty . . . . . . . . . . . . . . . . . . . . . . .  25
     12.4.  SSRC Collision Detection . . . . . . . . . . . . . . . .  27
     12.5.  Contributing Sources and the CSRC List . . . . . . . . .  28
     12.6.  Media Synchronization  . . . . . . . . . . . . . . . . .  28
     12.7.  Multiple RTP End-points  . . . . . . . . . . . . . . . .  29
     12.8.  Simulcast  . . . . . . . . . . . . . . . . . . . . . . .  30
     12.9.  Differentiated Treatment of Flows  . . . . . . . . . . .  30
   13. Security Considerations . . . . . . . . . . . . . . . . . . .  32
   14. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  33
   15. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . .  33
   16. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  33
   17. References  . . . . . . . . . . . . . . . . . . . . . . . . .  33
     17.1.  Normative References . . . . . . . . . . . . . . . . . .  34
     17.2.  Informative References . . . . . . . . . . . . . . . . .  37
   Appendix A.  Supported RTP Topologies . . . . . . . . . . . . . .  38
     A.1.  Point to Point  . . . . . . . . . . . . . . . . . . . . .  39
     A.2.  Multi-Unicast (Mesh)  . . . . . . . . . . . . . . . . . .  41
     A.3.  Mixer Based . . . . . . . . . . . . . . . . . . . . . . .  44
       A.3.1.  Media Mixing  . . . . . . . . . . . . . . . . . . . .  45
       A.3.2.  Media Switching . . . . . . . . . . . . . . . . . . .  47
       A.3.3.  Media Projecting  . . . . . . . . . . . . . . . . . .  50
     A.4.  Translator Based  . . . . . . . . . . . . . . . . . . . .  52
       A.4.1.  Transcoder  . . . . . . . . . . . . . . . . . . . . .  52
       A.4.2.  Gateway / Protocol Translator . . . . . . . . . . . .  53
       A.4.3.  Relay . . . . . . . . . . . . . . . . . . . . . . . .  55
     A.5.  End-point Forwarding  . . . . . . . . . . . . . . . . . .  58
     A.6.  Simulcast . . . . . . . . . . . . . . . . . . . . . . . .  60
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  60

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.

   The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete
   WebRTC framework, of which this memo is a part.

Perkins, et al.         Expires January 16, 2014                [Page 3]

Internet-Draft               RTP for WebRTC                    July 2013

   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
   into this framework.  Section 10 describes requirements placed on the
   signalling channel.  Section 11 discusses the relationship between
   features of the RTP framework and the WebRTC application programming
   interface (API), and Section 12 discusses RTP implementation
   considerations.  This memo concludes with an appendix discussing
   several different RTP Topologies, and how they affect the RTP
   session(s) and various implementation details of possible realization
   of central nodes.

2.  Rationale

   The RTP framework comprises the RTP data transfer protocol, the RTP
   control protocol, and numerous RTP payload formats, profiles, and
   extensions.  This range of add-ons has allowed RTP to meet various
   needs that were not envisaged by the original protocol designers, and
   to support many new media encodings, but raises the question of what
   extensions are to be supported by new implementations.  The
   development of the WebRTC framework provides an opportunity for us to
   review the available RTP features and extensions, and to define a
   common baseline feature set for all WebRTC implementations of RTP.
   This builds on the past 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.

   Other RTP and RTCP extensions not discussed in this document can be
   implemented by WebRTC end-points if they are beneficial for new use
   cases.  However, they are not necessary to address the WebRTC use
   cases and requirements identified to date

Perkins, et al.         Expires January 16, 2014                [Page 4]

Internet-Draft               RTP for WebRTC                    July 2013

   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

3.  Terminology

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

   We define the following terms:

   RTP Media Stream:  A sequence of RTP packets, and associated RTCP
      packets, using a single synchronisation source (SSRC) that
      together carries part or all of the content of a specific Media
      Type from a specific sender source within a given RTP session.

   RTP Session:  As defined by [RFC3550], the endpoints belonging to the
      same RTP Session are those that share a single SSRC space.  That
      is, those endpoints can see an SSRC identifier transmitted by any
      one of the other endpoints.  An endpoint can see an SSRC either
      directly in RTP and RTCP packets, or as a contributing source
      (CSRC) in RTP packets from a mixer.  The RTP Session scope is
      hence decided by the endpoints' network interconnection topology,
      in combination with RTP and RTCP forwarding strategies deployed by
      endpoints and any interconnecting middle nodes.

   WebRTC MediaStream:  The MediaStream concept defined by the W3C in
      the API.

   Other terms are used according to their definitions from the RTP
   Specification [RFC3550] and WebRTC overview
   [I-D.ietf-rtcweb-overview] documents.

4.  WebRTC Use of RTP: Core Protocols

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

Perkins, et al.         Expires January 16, 2014                [Page 5]

Internet-Draft               RTP for WebRTC                    July 2013

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

   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.

      *  (tbd: is draft-westerlund-mmusic-max-ssrc-01 needed?)

   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  Support for sending correct synchronization information in the
      RTCP Sender Reports, to allow a receiver to implement lip-sync,
      with RECOMMENDED support for the rapid RTP synchronisation
      extensions (see Section 5.2.1).

   o  Support for sending and receiving RTCP SR, RR, SDES, and BYE
      packet types, with OPTIONAL support for other RTCP packet types;
      implementations MUST ignore unknown RTCP packet types.

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

   o  Support for configuring the RTCP bandwidth as a fraction of the
      media bandwidth, and for configuring the fraction of the RTCP
      bandwidth allocated to senders, e.g., using the SDP "b=" line.

   It is known that a significant number of legacy RTP implementations,
   especially those targeted at VoIP-only systems, do not support all of

Perkins, et al.         Expires January 16, 2014                [Page 6]

Internet-Draft               RTP for WebRTC                    July 2013

   the above features, and in some cases do not support RTCP at all.
   Implementers are advised to consider the requirements for graceful
   degradation when interoperating with legacy implementations.

   Other implementation considerations are discussed in Section 12.

4.2.  Choice of the RTP Profile

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

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

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

   The secure RTP profile [RFC3711] is needed to provide media
   encryption, integrity protection, replay protection and a limited
   form of source authentication.  WebRTC implementations MUST NOT send
   packets using the basic RTP/AVP profile or the RTP/AVPF profile; they
   MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP
   packets that are generated.  The default and mandatory to implement
   transforms listed in Section 5 of [RFC3711] SHALL apply.

   Implementations MUST support DTLS-SRTP [RFC5764] for key-management.
   Other key management schemes MAY be supported.

4.3.  Choice of RTP Payload Formats

   The set of mandatory to implement codecs and RTP payload formats for
   WebRTC is not specified in this memo.  Implementations can support

Perkins, et al.         Expires January 16, 2014                [Page 7]

Internet-Draft               RTP for WebRTC                    July 2013

   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 payload format, no matter how
   common; support for all RTP payload formats MUST be negotiated before
   they are used.

   Endpoints can signal support for multiple RTP payload formats, or
   multiple configurations of a single RTP payload format, as long as
   each unique RTP payload format configuration uses a different RTP
   payload type number.  As outlined in Section 4.8, the RTP payload
   type number is sometimes used to associate an RTP media stream with a
   signalling context.  This association is possible provided unique RTP
   payload type numbers are used in each context.  For example, an RTP
   media stream can be associated with an SDP "m=" line by comparing the
   RTP payload type numbers used by the media stream with payload types
   signalled in the "a=rtpmap:" lines in the media sections of the SDP.
   If RTP media 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 associated RTP media 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.

   An endpoint that has signalled support for multiple RTP payload
   formats SHOULD accept data in any of those payload formats at any
   time, unless it has previously signalled limitations on its decoding
   capability.  This requirement is constrained if several 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 that were signalled during
   session set-up.

   An RTP sender that changes between two RTP payload types that use
   different RTP clock rates MUST follow the recommendations in
   Section 4.1 of [I-D.ietf-avtext-multiple-clock-rates].  RTP receivers
   MUST follow the recommendations in Section 4.3 of
   [I-D.ietf-avtext-multiple-clock-rates], in order to support sources
   that switch between clock rates in an RTP session (these
   recommendations for receivers are backwards compatible with the case
   where senders use only a single clock rate).

Perkins, et al.         Expires January 16, 2014                [Page 8]

Internet-Draft               RTP for WebRTC                    July 2013

4.4.  Use of RTP Sessions

   An association amongst a set of participants communicating using RTP
   is known as an RTP session.  A participant 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
   different transport addresses 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
   addresses (e.g., different UDP ports) for compatibility with legacy

   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 media 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 12.9).  Implementations are REQUIRED to support transport
   of all RTP media streams, independent of media type, in a single RTP
   session according to [I-D.ietf-avtcore-multi-media-rtp-session].  If
   such RTP session set-up is to be used, this MUST be negotiated during
   the signalling phase [I-D.ietf-mmusic-sdp-bundle-negotiation].

   It is also possible to use a shim-based approach to run multiple RTP
   sessions on a single transport-layer flow.  This gives advantages in
   some gateway scenarios, and makes it easy to distinguish groups of
   RTP media streams that might need distinct processing.  One way of
   doing this is described in
   [I-D.westerlund-avtcore-transport-multiplexing].  At the time of this
   writing, there is no consensus to use a shim-based approach in WebRTC

   Further discussion about when different RTP session structures and
   multiplexing methods are suitable can be found in

4.5.  RTP and RTCP Multiplexing

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

Perkins, et al.         Expires January 16, 2014                [Page 9]

Internet-Draft               RTP for WebRTC                    July 2013

   a single port for each RTP session is REQUIRED, as specified in
   [RFC5761].  For backwards compatibility, implementations are also
   REQUIRED to support RTP and RTCP sent on separate transport-layer

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

4.6.  Reduced Size RTCP

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

   To avoid this problem, [RFC5506] specifies how to reduce the mean
   RTCP message size and allow for more frequent feedback.  Frequent
   feedback, in turn, is essential to make real-time applications
   quickly aware of changing network conditions, and to allow them to
   adapt their transmission and encoding behaviour.  Support for 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 endpoint does not
   agree to the use of non-compound RTCP in the signalling exchange.

4.7.  Symmetric RTP/RTCP

   To ease traversal of NAT and firewall devices, implementations are
   REQUIRED to implement and use Symmetric RTP [RFC4961].  The reasons
   for using symmetric RTP is primarily to avoid issues with NAT and
   Firewalls by ensuring that the flow is actually bi-directional and
   thus kept alive and registered as flow the intended recipient
   actually wants.  In addition, it saves resources, specifically ports
   at the end-points, but also in the network as NAT mappings or
   firewall state is not unnecessary bloated.  Also the amount of QoS
   state is reduced.

4.8.  Choice of RTP Synchronisation Source (SSRC)

   Implementations are REQUIRED to support signalled RTP synchronisation
   source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined

Perkins, et al.         Expires January 16, 2014               [Page 10]

Internet-Draft               RTP for WebRTC                    July 2013

   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

   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 media stream with a non-RTP
   context (e.g., to associate an RTP media 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
   media stream are often sufficient to associate that media 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 media 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="

4.9.  Generation of the RTCP Canonical Name (CNAME)

   The RTCP Canonical Name (CNAME) provides a persistent transport-level
   identifier for an RTP endpoint.  While the Synchronisation Source
   (SSRC) identifier for an RTP endpoint can change if a collision is
   detected, or when the RTP application is restarted, its RTCP CNAME is
   meant to stay unchanged, so that RTP endpoints can be uniquely
   identified and associated with their RTP media streams within a set
   of related RTP sessions.  For proper functionality, each RTP endpoint
   needs to have a unique RTCP CNAME value.

   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME, but these are not sufficient in the presence of NAT
   devices.  In addition, long-term persistent identifiers can be
   problematic from a privacy viewpoint.  Accordingly, support for
   generating a short-term persistent RTCP CNAMEs following
   [I-D.ietf-avtcore-6222bis] is RECOMMENDED.

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

Perkins, et al.         Expires January 16, 2014               [Page 11]

Internet-Draft               RTP for WebRTC                    July 2013

5.  WebRTC Use of RTP: Extensions

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

5.1.  Conferencing Extensions

   RTP is inherently a group communication protocol.  Groups can be
   implemented using a centralised server, multi-unicast, or using IP
   multicast.  While IP multicast is popular in IPTV systems, overlay-
   based topologies dominate in interactive conferencing environments.
   Such overlay-based topologies typically use one or more central
   servers to connect end-points in a star or flat tree topology.  These
   central servers can be implemented in a number of ways as discussed
   in Appendix A, and in the memo on RTP Topologies

   Not all of the possible the overlay-based topologies are suitable for
   use in the WebRTC environment.  Specifically:

   o  The use of video switching MCUs makes the use of RTCP for
      congestion control and quality of service reports problematic (see
      Section 3.7 of [I-D.westerlund-avtcore-rtp-topologies-update]).

   o  The use of content modifying MCUs with RTCP termination breaks RTP
      loop detection, and prevents receivers from identifying active
      senders (see section 3.8 of

   o  RTP Transport Translators (Topo-Translator) are not of immediate
      interest to WebRTC, although the main difference compared to point
      to point is the possibility of seeing multiple different transport
      paths in any RTCP feedback.

   Accordingly, only Point to Point (Topo-Point-to-Point), Multiple
   concurrent Point to Point (Mesh) and RTP Mixers (Topo-Mixer)
   topologies are needed to achieve the use-cases to be supported in
   WebRTC initially.  These RECOMMENDED topologies are expected to be
   supported by all WebRTC end-points (these topologies require no
   special RTP-layer support in the end-point if the RTP features
   mandated in this memo are implemented).

   The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
   designed to be used with centralised conferencing, where an RTP

Perkins, et al.         Expires January 16, 2014               [Page 12]

Internet-Draft               RTP for WebRTC                    July 2013

   middlebox (e.g., a conference bridge) receives a participant's RTP
   media streams and distributes them to the other participants.  These
   extensions are not necessary for interoperability; an RTP endpoint
   that does not implement these extensions will work correctly, but
   might offer poor performance.  Support for the listed extensions will
   greatly improve the quality of experience and, to provide a
   reasonable baseline quality, some these extensions are mandatory to
   be supported by WebRTC end-points.

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

5.1.1.  Full Intra Request (FIR)

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

5.1.2.  Picture Loss Indication (PLI)

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

5.1.3.  Slice Loss Indication (SLI)

   The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF
   profile [RFC4585].  It is used by a receiver to tell the encoder that
   it has detected the loss or corruption of one or more consecutive
   macro blocks, and would like to have these repaired somehow.  Support
   for this feedback message is OPTIONAL as a loss tolerance mechanism.

5.1.4.  Reference Picture Selection Indication (RPSI)

Perkins, et al.         Expires January 16, 2014               [Page 13]

Internet-Draft               RTP for WebRTC                    July 2013

   Reference Picture Selection Indication (RPSI) is defined in
   Section 6.3.3 of the RTP/AVPF profile [RFC4585].  Some video coding
   standards allow the use of older reference pictures than the most
   recent one for predictive coding.  If such a codec is in used, and if
   the encoder has learned about a loss of encoder-decoder
   synchronisation, a known-as-correct reference picture can be used for
   future coding.  The RPSI message allows this to be signalled.
   Support for RPSI messages is OPTIONAL.

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

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

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

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

5.2.  Header Extensions

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

   As noted in [RFC5285], the requirement from the RTP specification
   that header extensions are "designed so that the header extension may
   be ignored" [RFC3550] stands.  To be specific, header extensions MUST
   only be used for data that can safely be ignored by the recipient
   without affecting interoperability, and MUST NOT be used when the
   presence of the extension has changed the form or nature of the rest

Perkins, et al.         Expires January 16, 2014               [Page 14]

Internet-Draft               RTP for WebRTC                    July 2013

   of the packet in a way that is not compatible with the way the stream
   is signalled (e.g., as defined by the payload type).  Valid examples
   might include metadata that is additional to the usual RTP

5.2.1.  Rapid Synchronisation

   Many RTP sessions require synchronisation between audio, video, and
   other content.  This synchronisation is performed by receivers, using
   information contained in RTCP SR packets, as described in the RTP
   specification [RFC3550].  This basic mechanism can be slow, however,
   so it is RECOMMENDED that the rapid RTP synchronisation extensions
   described in [RFC6051] be implemented 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 a client to inform a mixer about the level of audio
   activity in the packet to which the header is attached.  This enables
   a central node to make mixing or selection decisions without decoding
   or detailed inspection of the payload, reducing the complexity in
   some types of central RTP nodes.  It can also save decoding resources
   in receivers, which can choose to decode only the most relevant RTP
   media streams based on audio activity levels.

   The Client-to-Mixer Audio Level [RFC6464] extension is RECOMMENDED to
   be implemented.  If it is implemented, it is REQUIRED that the header
   extensions are encrypted according to
   [I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
   contained in these header extensions can be considered sensitive.

5.2.3.  Mixer-to-Client Audio Level

   The Mixer to Client Audio Level header extension [RFC6465] provides
   the client with the audio level of the different sources mixed into a
   common mix by a RTP mixer.  This enables a user interface to indicate
   the relative activity level of each session participant, rather than
   just being included or not based on the CSRC field.  This is a pure
   optimisations of non critical functions, and is hence OPTIONAL to
   implement.  If it is implemented, it is REQUIRED that the header
   extensions are encrypted according to
   [I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information
   contained in these header extensions can be considered sensitive.

Perkins, et al.         Expires January 16, 2014               [Page 15]

Internet-Draft               RTP for WebRTC                    July 2013

6.  WebRTC Use of RTP: Improving Transport Robustness

   There are tools that can make RTP media streams robust against packet
   loss and reduce the impact of loss on media quality.  However, they
   all add extra bits compared to a non-robust stream.  The overhead of
   these extra bits 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
   will support negative acknowledgements (NACKs) for RTP data packets
   [RFC4585].  This feedback can be used to inform a sender of the loss
   of particular RTP packets, subject to the capacity limitations of the
   RTCP feedback channel.  A sender can use this information to optimise
   the user experience by adapting the media encoding to compensate for
   known lost packets, for example.

   Senders are REQUIRED to understand the Generic NACK message defined
   in Section 6.2.1 of [RFC4585], but MAY choose to ignore this feedback
   (following Section 4.2 of [RFC4585]).  Receivers MAY send NACKs for
   missing RTP packets; [RFC4585] provides some guidelines on when to
   send NACKs.  It is not expected that a receiver will send a NACK for
   every lost RTP packet, rather it needs to consider the cost of
   sending NACK feedback, and the importance of the lost packet, to make
   an informed decision on whether it is worth telling the sender about
   a packet loss event.

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

Perkins, et al.         Expires January 16, 2014               [Page 16]

Internet-Draft               RTP for WebRTC                    July 2013

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

6.2.  Forward Error Correction (FEC)

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

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

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

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

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

Perkins, et al.         Expires January 16, 2014               [Page 17]

Internet-Draft               RTP for WebRTC                    July 2013

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

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

7.1.  Boundary Conditions and Circuit Breakers

   In the absence of a concrete congestion control algorithm, all WebRTC
   implementations MUST implement the RTP circuit breaker algorithm that
   is in described [I-D.ietf-avtcore-rtp-circuit-breakers].  The circuit
   breaker defines a conservative boundary condition for safe operation,
   chosen such that applications that trigger the circuit breaker will
   almost certainly be causing severe network congestion.  Any future
   RTP congestion control algorithms are expected to operate within the
   envelope allowed by the circuit breaker.

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

Perkins, et al.         Expires January 16, 2014               [Page 18]

Internet-Draft               RTP for WebRTC                    July 2013

7.2.  RTCP Limitations for Congestion Control

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

   The total RTCP bandwidth is limited in its transmission rate to a
   fraction of the RTP traffic (by default 5%).  RTCP packets are larger
   than, e.g., TCP ACKs (even when non-compound RTCP packets are used).
   The RTP media stream bit rate thus limits the maximum feedback rate
   as a function of the mean RTCP packet size.

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

   In group communication, the share of RTCP bandwidth needs to be
   shared by all group members, reducing the capacity and thus the
   reporting frequency per node.

   Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP
   bandwidth, split across two entities in a point-to-point session.  An
   endpoint could thus send a report of 100 bytes about every 70ms or
   for every other frame in a 30 fps video.

7.3.  Congestion Control Interoperability and Legacy Systems

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

Perkins, et al.         Expires January 16, 2014               [Page 19]

Internet-Draft               RTP for WebRTC                    July 2013

   When interworking with legacy implementations that support RTCP using
   the RTP/AVP profile [RFC3551], congestion feedback is provided in
   RTCP RR packets every few seconds.  Implementations that have to
   interwork with such end-points MUST ensure that they keep within the
   RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers]
   constraints to limit the congestion they can cause.

   If a legacy end-point supports RTP/AVPF, this enables negotiation of
   important parameters for frequent reporting, such as the "trr-int"
   parameter, and the possibility that the end-point supports some
   useful feedback format for congestion control purpose such as TMMBR
   [RFC5104].  Implementations that have to interwork with such end-
   points MUST ensure that they stay within the RTP circuit breaker
   [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the
   congestion they can cause, but might find that they can achieve
   better congestion response depending on the amount of feedback that
   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 media streams they send and receive.  These RTCP reports can
   be used for performance monitoring purposes, since they include basic
   packet loss and jitter statistics.

Perkins, et al.         Expires January 16, 2014               [Page 20]

Internet-Draft               RTP for WebRTC                    July 2013

   A large number of additional performance metrics are supported by the
   RTCP Extended Reports (XR) framework [RFC3611].  It is not yet clear
   what extended metrics are appropriate for use in the WebRTC context,
   so implementations are not expected to generate any 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.

   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] 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

   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

   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

Perkins, et al.         Expires January 16, 2014               [Page 21]

Internet-Draft               RTP for WebRTC                    July 2013

      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:" attribute to indicate a trr-
      int value of 4 seconds.

   Transport Information:  Source and destination IP address(s) and
      ports for RTP and RTCP MUST be signalled for each RTP session.  In
      WebRTC these transport addresses will be provided by ICE that
      signals candidates and arrives at nominated candidate address
      pairs.  If RTP and RTCP multiplexing [RFC5761] is to be used, such
      that a single port is used for RTP and RTCP flows, this MUST be
      signalled (see Section 4.5).  If several RTP sessions are to be
      multiplexed onto a single transport layer flow, this MUST also be
      signalled (see Section 4.4).

   RTP Payload Types, media formats, and 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

   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.

Perkins, et al.         Expires January 16, 2014               [Page 22]

Internet-Draft               RTP for WebRTC                    July 2013

11.  WebRTC API Considerations

   The WebRTC API and its media function have the concept of a WebRTC
   MediaStream that consists of zero or more tracks.  A track is an
   individual stream of media from any type of media source like a
   microphone or a camera, but also conceptual sources, like a audio mix
   or a video composition, are possible.  The tracks within a WebRTC
   MediaStream are expected to be synchronized.

   A track correspond to the media received with one particular SSRC.
   There might be additional SSRCs associated with that SSRC, like for
   RTP retransmission or Forward Error Correction.  However, one SSRC
   will identify an RTP media stream and its timing.

   As a result, a WebRTC MediaStream is a collection of SSRCs carrying
   the different media included in the synchronised aggregate.
   Therefore, also the synchronization state associated with the
   included SSRCs are part of concept.  It is important to consider that
   there can be multiple different WebRTC MediaStreams containing a
   given Track (SSRC).  To avoid unnecessary duplication of media at the
   transport level in such cases, a need arises for a binding defining
   which WebRTC MediaStreams a given SSRC is associated with at the
   signalling level.

   The API also needs to be capable of handling when new SSRCs are
   received but not previously signalled by signalling in some fashion.
   Note, that not all SSRCs carries media directly associated with a
   media source, instead they can be repair or redundancy information
   for one or a set of SSRCs.

   A proposal for how the binding between WebRTC MediaStreams and SSRC
   can be done is specified in "Cross Session Stream Identification in
   the Session Description Protocol" [I-D.alvestrand-rtcweb-msid].

   (tbd: This text needs to be improved and achieved consensus on.
   Interim meeting in June 2012 shows large differences in opinions.)

   (tbd: It is an open question whether these considerations are best
   discussed in this draft, in the W3C WebRTC API spec, or elsewhere.

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.

Perkins, et al.         Expires January 16, 2014               [Page 23]

Internet-Draft               RTP for WebRTC                    July 2013

12.1.  RTP Sessions and PeerConnections

   An RTP session is an association among RTP nodes, which have a single
   shared SSRC space.  An RTP session can include a large number of end-
   points and nodes, each sourcing, sinking, manipulating, or reporting
   on the RTP media streams being sent within the RTP session.

   A PeerConnection is a point-to-point association between an end-point
   and some other peer node.  That peer node can be either an end-point
   or a centralized processing node of some type.  Hence, an RTP session
   can terminate immediately at the far end of a PeerConnection, or it
   might continue as further discussed below for multiparty sessions
   (Section 12.3) and sessions with multiple end points (Section 12.7).

   A PeerConnection can contain one or more RTP sessions, depending on
   how it is set up, and how many UDP flows it uses.  A common usage has
   been to have one RTP session per media type, e.g.  one for audio and
   one for video, each sent over a different UDP flow.  However, the
   default usage in WebRTC will be to use one RTP session for all media
   types, with RTP and RTCP multiplexing (Section 4.5) also mandated.
   This RTP session then uses only one UDP flow.  However, for legacy
   interworking and flow-based network prioritization (Section 12.9), a
   WebRTC end-point needs to support a mode of operation where one RTP
   session per media type is used.  Currently, each RTP session has to
   use its own UDP flow in this case, however it might be possible to
   multiplex several RTP sessions over a single UDP flow, see
   Section 4.4.

   The multi-unicast- or mesh-based multi-party topology (Figure 1) is a
   good example for this section as it concerns the relation between RTP
   sessions and PeerConnections.  In this topology, each participant
   sends individual unicast RTP/UDP/IP flows to each of the other
   participants using independent PeerConnections in a full mesh.  This
   topology has the benefit of not requiring central nodes.  The
   downside is that it increases the used bandwidth at each sender by
   requiring one copy of the RTP media streams for each participant that
   are part of the same session beyond the sender itself.  Hence, this
   topology is limited to scenarios with few participants unless the
   media is very low bandwidth.

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

Perkins, et al.         Expires January 16, 2014               [Page 24]

Internet-Draft               RTP for WebRTC                    July 2013

                                  | C |

                          Figure 1: Multi-unicast

   The multi-unicast topology could be implemented as a single RTP
   session, spanning multiple peer-to-peer transport layer connections,
   or as several pairwise RTP sessions, one between each pair of peers.
   To maintain a coherent mapping between the relation between RTP
   sessions and PeerConnections we recommend that one implements this as
   individual RTP sessions.  The only downside is that end-point A will
   not learn of the quality of any transmission happening between B and
   C based on RTCP.  This has not been seen as a significant downside as
   no one has yet seen a clear need for why A would need to know about
   the B's and C's communication.  An advantage of using separate RTP
   sessions is that it enables using different media bit-rates to the
   different peers, thus not forcing B to endure the same quality
   reductions if there are limitations in the transport from A to C as C

12.2.  Multiple Sources

   A WebRTC end-point might have multiple cameras, microphones or audio
   inputs and thus a single end-point can source multiple RTP media
   streams of the same media type concurrently.  Even if an end-point
   does not have multiple media sources of the same media type it has to
   support transmission using multiple SSRCs concurrently in the same
   RTP session.  This is due to the requirement on an WebRTC end-point
   to support multiple media types in one RTP session.  For example, one
   audio and one video source can result in the end-point sending with
   two different SSRCs in the same RTP session.  As multi-party
   conferences are supported, as discussed below in Section 12.3, a
   WebRTC end-point will need to be capable of receiving, decoding and
   play out multiple RTP media streams of the same type concurrently.

   tbd: there needs to be a way of indicating how RTP stream relate when
   there are multiple sources, possibly with simulcast or layered
   coding, and different types of mixer or other middlebox.  It is
   possible that the various BUNDLE/Plan-X proposals will solve this,
   but it might also need RTP-level stream identification.  To be
   resolved once the outcome of the BUNDLE and plan-X discussions is

   tbd: Are any mechanism needed to signal limitations in the number of
   active SSRC that an end-point can handle?

12.3.  Multiparty

Perkins, et al.         Expires January 16, 2014               [Page 25]

Internet-Draft               RTP for WebRTC                    July 2013

   There are numerous situations and clear use cases for WebRTC
   supporting RTP sessions supporting multi-party.  This can be realized
   in a number of ways using a number of different implementation
   strategies.  In the following, the focus is on the different set of
   WebRTC end-point requirements that arise from different sets of
   multi-party topologies.

   The multi-unicast mesh (Figure 1)-based multi-party topology
   discussed above provides a non-centralized solution but can incur a
   heavy tax on the end-points' outgoing paths.  It can also consume
   large amount of encoding resources if each outgoing stream is
   specifically encoded.  If an encoding is transmitted to multiple
   parties, as in some implementations of the mesh case, a requirement
   on the end-point becomes to be able to create RTP media streams
   suitable for multiple destinations requirements.  These requirements
   can both be dependent on transport path and the different end-points
   preferences related to play out of the media.

                   +---+      +------------+      +---+
                   | A |<---->|            |<---->| B |
                   +---+      |            |      +---+
                              |   Mixer    |
                   +---+      |            |      +---+
                   | C |<---->|            |<---->| D |
                   +---+      +------------+      +---+

                Figure 2: RTP Mixer with Only Unicast Paths

   A Mixer (Figure 2) is an RTP end-point that optimizes the
   transmission of RTP media streams from certain perspectives, either
   by only sending some of the received RTP media stream to any given
   receiver or by providing a combined RTP media stream out of a set of
   contributing streams.  There are various methods of implementation as
   discussed in Appendix A.3.  A common aspect is that these central
   nodes can use a number of tools to control the media encoding
   provided by a WebRTC end-point.  This includes functions like
   requesting breaking the encoding chain and have the encoder produce a
   so called Intra frame.  Another is limiting the bit-rate of a given
   stream to better suit the mixer view of the multiple down-streams.
   Others are controlling the most suitable frame-rate, picture
   resolution, the trade-off between frame-rate and spatial quality.

   A mixer gets a significant responsibility to correctly perform
   congestion control, source identification, manage synchronization
   while providing the application with suitable media optimizations.

   Mixers also need to be trusted nodes when it comes to security as it
   manipulates either RTP or the media itself before sending it on

Perkins, et al.         Expires January 16, 2014               [Page 26]

Internet-Draft               RTP for WebRTC                    July 2013

   towards the end-point(s), thus they need to be able to decrypt and
   then encrypt it before sending it out.

12.4.  SSRC Collision Detection

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

   In a point-to-point session where each SSRC is associated with either
   of the two end-points and where the main media carrying SSRC
   identifier will be announced in the signalling channel, a collision
   is less likely to occur due to the information about used SSRCs
   provided by Source-Specific SDP Attributes [RFC5576].  Still if both
   end-points start uses an new SSRC identifier prior to having
   signalled it to the peer and received acknowledgement on the
   signalling message, there can be collisions.  The Source-Specific SDP
   Attributes [RFC5576] contains no mechanism to resolve SSRC collisions
   or reject a end-points usage of an SSRC.

   There could also appear SSRC values that are not signalled.  This is
   more likely than it appears as certain RTP functions need extra SSRCs
   to provide functionality related to another (the "main") SSRC, for
   example, SSRC multiplexed RTP retransmission [RFC4588].  In those
   cases, an end-point can create a new SSRC that strictly doesn't need
   to be announced over the signalling channel to function correctly on
   both RTP and PeerConnection level.

   The more likely case for SSRC collision is that multiple end-points
   in a multiparty conference create new sources and signals those
   towards the central server.  In cases where the SSRC/CSRC are
   propagated between the different end-points from the central node
   collisions can occur.

   Another scenario is when the central node manages to connect an end-
   point's PeerConnection to another PeerConnection the end-point
   already has, thus forming a loop where the end-point will receive its
   own traffic.  While is is clearly considered a bug, it is important
   that the end-point is able to recognise and handle the case when it
   occurs.  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 PeerConnections
   by a RTP middlebox.  To resolve the more generic case where multiple

Perkins, et al.         Expires January 16, 2014               [Page 27]

Internet-Draft               RTP for WebRTC                    July 2013

   PeerConnections are interconnected, then identification of the media
   source(s) part of a MediaStreamTrack being propagated across multiple
   interconnected PeerConnection needs to be preserved across these

12.5.  Contributing Sources and the CSRC List

   RTP allows a mixer, or other RTP-layer middlebox, to combine media
   flows from multiple sources to form a new media flow.  The RTP data
   packets in that new flow will include a Contributing Source (CSRC)
   list, indicating which original SSRCs contributed to the combined
   packet.  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 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

   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.

   This memo does not require implementations to be able to add a CSRC
   list to outgoing RTP packets.  It is expected that the any CSRC list
   will be added by a mixer or other middlebox that performs in-network
   processing of RTP streams.  If there is a desire to allow end-system
   mixing, the requirement in Section 4.1 will need to be updated to
   support setting the CSRC list in outgoing RTP data packets.

12.6.  Media Synchronization

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

Perkins, et al.         Expires January 16, 2014               [Page 28]

Internet-Draft               RTP for WebRTC                    July 2013

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

12.7.  Multiple RTP End-points

   Some usages of RTP beyond the recommend topologies result in that an
   WebRTC end-point sending media in an RTP session out over a single
   PeerConnection will receive receiver reports from multiple RTP
   receivers.  Note that receiving multiple receiver reports is expected
   because any RTP node that has multiple SSRCs has to report to the
   media sender.  The difference here is that they are multiple nodes,
   and thus will likely have different path characteristics.

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

   The topologies in which an end-point receives receiver reports from
   multiple other end-points are the centralized relay, multicast and an
   end-point forwarding an RTP media stream.  Having multiple RTP nodes
   receive an RTP flow and send reports and feedback about it has
   several impacts.  As previously discussed (Section 12.3) any codec
   control and rate control needs to be capable of merging the
   requirements and preferences to provide a single best encoding
   according to the situation RTP media stream.  Specifically, when it
   comes to congestion control it needs to be capable of identifying the
   different end-points to form independent congestion state information
   for each different path.

   Providing source authentication in multi-party scenarios is a
   challenge.  In the mixer-based topologies, end-points source

Perkins, et al.         Expires January 16, 2014               [Page 29]

Internet-Draft               RTP for WebRTC                    July 2013

   authentication is based on, firstly, verifying that media comes from
   the mixer by cryptographic verification and, secondly, trust in the
   mixer to correctly identify any source towards the end-point.  In RTP
   sessions where multiple end-points are directly visible to an end-
   point, all end-points will have knowledge about each others' master
   keys, and can thus inject packets claimed to come from another end-
   point in the session.  Any node performing relay can perform non-
   cryptographic mitigation by preventing forwarding of packets that
   have SSRC fields that came from other end-points before.  For
   cryptographic verification of the source SRTP would require
   additional security mechanisms, like TESLA for SRTP [RFC4383].

12.8.  Simulcast

   This section discusses simulcast in the meaning of providing a node,
   for example a Mixer, with multiple different encoded versions of the
   same media source.  In the WebRTC context, this could be accomplished
   in two ways.  One is to establish multiple PeerConnection all being
   feed the same set of WebRTC MediaStreams.  Another method is to use
   multiple WebRTC MediaStreams that are differently configured when it
   comes to the media parameters.  This would result in that multiple
   different RTP Media Streams (SSRCs) being in used with different
   encoding based on the same media source (camera, microphone).

   When intending to use simulcast it is important that this is made
   explicit so that the end-points don't automatically try to optimize
   away the different encodings and provide a single common version.
   Thus, some explicit indications that the intent really is to have
   different media encodings is likely needed.  It is to be noted that
   it might be a central node, rather than an WebRTC end-point that
   would benefit from receiving simulcast media sources.

   tbd: How to perform simulcast needs to be determined and the
   appropriate API or signalling for its usage needs to be defined.

12.9.  Differentiated Treatment of Flows

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

   It is expected that the WebRTC API will allow the application to
   indicate relative priorities for different MediaStreamTracks.  These
   priorities can then be used to influence the local RTP processing,
   especially when it comes to congestion control response in how to

Perkins, et al.         Expires January 16, 2014               [Page 30]

Internet-Draft               RTP for WebRTC                    July 2013

   divide the available bandwidth between the RTP flows.  Any changes in
   relative priority will also need to be considered for RTP flows that
   are associated with the main RTP flows, such as RTP retransmission
   streams and FEC.  The importance of such associated RTP traffic flows
   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 associated RTP flows as for the
   primary RTP flow.

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

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

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

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

   Flow-based differentiation will provide the same treatment to all
   packets within a flow, i.e., relative prioritization is not possible.
   Moreover, if the resources are limited it might not be possible to
   provide differential treatment compared to best-effort for all the
   flows in a WebRTC application.  When flow-based differentiation is
   available the WebRTC application needs to know about it so that it
   can provide the separation of the RTP media streams onto different
   UDP flows to enable a more granular usage of flow based
   differentiation.  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 media streams.  2) The information needs
   to be propagated to the operating system when transmitting the
   packet.  These issues are discussed in DSCP and other packet markings
   for RTCWeb QoS [I-D.ietf-rtcweb-qos].

Perkins, et al.         Expires January 16, 2014               [Page 31]

Internet-Draft               RTP for WebRTC                    July 2013

   For packet based marking schemes it would be possible in the context
   to mark individual RTP packets differently based on the relative
   priority of the RTP payload.  For example video codecs that has I,P
   and B pictures could prioritise any payloads carrying only B frames
   less, as these are less damaging to loose.  But as default policy all
   RTP packets related to a media stream ought to be provided with the
   same prioritization.

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

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 apply to this memo also.

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

   The Extended Secure RTP Profile for Real-time Transport Control
   Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
   handling of fundamental issues by offering confidentiality, integrity
   and partial source authentication.  A mandatory to implement media
   security solution is (tbd).

   RTCP packets convey a Canonical Name (CNAME) identifier that is used
   to associate media flows 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

Perkins, et al.         Expires January 16, 2014               [Page 32]

Internet-Draft               RTP for WebRTC                    July 2013

   audio codecs).  These 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).

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.  Open Issues

   This section contains a summary of the open issues or to be done
   things noted in the document:

   1.  tbd: The API mapping to RTP level concepts has to be agreed and
       documented in Section 11.

   2.  tbd: An open question if any requirements are needed to agree and
       limit the number of simultaneously used media sources (SSRCs)
       within an RTP session.  See Section 12.2 and Section 4.1.

   3.  tbd: The method for achieving simulcast of a media source has to
       be decided as discussed in Section 12.8.

   4.  tbd: Possible documentation of what support for differentiated
       treatment that are needed on RTP level as the API and the network
       level specification matures as discussed in Section 12.9.

   5.  tbd: Editing of Appendix A to remove redundancy between this and
       the update of RTP Topologies

16.  Acknowledgements

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

17.  References

Perkins, et al.         Expires January 16, 2014               [Page 33]

Internet-Draft               RTP for WebRTC                    July 2013

17.1.  Normative References

              Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-06
              (work in progress), July 2013.

              Terriberry, T., "Update to Remove DVI4 from the
              Recommended Codecs for the RTP Profile for Audio and Video
              Conferences with Minimal Control (RTP/AVP)", draft-ietf-
              avtcore-avp-codecs-03 (work in progress), July 2013.

              Westerlund, M., Perkins, C., and J. Lennox, "Sending
              Multiple Types of Media in a Single RTP Session", draft-
              ietf-avtcore-multi-media-rtp-session-03 (work in
              progress), July 2013.

              Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", draft-ietf-
              avtcore-rtp-circuit-breakers-02 (work in progress),
              February 2013.

              Lennox, J., Westerlund, M., Wu, Q., 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-00 (work
              in progress), July 2013.

              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-01 (work in progress),
              July 2013.

              Lennox, J., "Encryption of Header Extensions in the Secure
              Real-Time Transport Protocol (SRTP)", draft-ietf-avtcore-
              srtp-encrypted-header-ext-05 (work in progress), February


Perkins, et al.         Expires January 16, 2014               [Page 34]

Internet-Draft               RTP for WebRTC                    July 2013

              Petit-Huguenin, M. and G. Zorn, "Support for Multiple
              Clock Rates in an RTP Session", draft-ietf-avtext-
              multiple-clock-rates-09 (work in progress), April 2013.

              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Multiplexing Negotiation Using Session Description
              Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
              bundle-negotiation-04 (work in progress), June 2013.

              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-06 (work
              in progress), February 2013.

              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-07 (work in progress), July 2013.

              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-05 (work in progress), July 2013.

              Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
              Single Lower-Layer Transport", draft-westerlund-avtcore-
              transport-multiplexing-05 (work in progress), February

   [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

   [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.

Perkins, et al.         Expires January 16, 2014               [Page 35]

Internet-Draft               RTP for WebRTC                    July 2013

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

   [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

   [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.

   [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.

Perkins, et al.         Expires January 16, 2014               [Page 36]

Internet-Draft               RTP for WebRTC                    July 2013

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

17.2.  Informative References

              Alvestrand, H., "Cross Session Stream Identification in
              the Session Description Protocol", draft-alvestrand-
              rtcweb-msid-02 (work in progress), May 2012.

              Wing, D., McGrew, D., and K. Fischer, "Encrypted Key
              Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03
              (work in progress), October 2011.

              Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
              other packet markings for RTCWeb QoS", draft-ietf-rtcweb-
              qos-00 (work in progress), October 2012.

              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements", draft-
              ietf-rtcweb-use-cases-and-requirements-11 (work in
              progress), June 2013.

              Jesup, R. and H. Alvestrand, "Congestion Control
              Requirements For Real Time Media", draft-jesup-rtp-
              congestion-reqs-00 (work in progress), March 2012.

              Westerlund, M., Perkins, C., and H. Alvestrand,
              "Guidelines for using the Multiplexing Features of RTP",
              draft-westerlund-avtcore-multiplex-architecture-03 (work
              in progress), February 2013.

              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              westerlund-avtcore-rtp-topologies-update-02 (work in
              progress), February 2013.

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

Perkins, et al.         Expires January 16, 2014               [Page 37]

Internet-Draft               RTP for WebRTC                    July 2013

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

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

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

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

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

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

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

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

Appendix A.  Supported RTP Topologies

   RTP supports both unicast and group communication, with participants
   being connected using wide range of transport-layer topologies.  Some
   of these topologies involve only the end-points, while others use RTP
   translators and mixers to provide in-network processing.  Properties
   of some RTP topologies are discussed in
   [I-D.westerlund-avtcore-rtp-topologies-update], and we further
   describe those expected to be useful for WebRTC in the following.  We
   also goes into important RTP session aspects that the topology or
   implementation variant can place on a WebRTC end-point.

Perkins, et al.         Expires January 16, 2014               [Page 38]

Internet-Draft               RTP for WebRTC                    July 2013

   This section includes RTP topologies beyond the RECOMMENDED ones.
   This in an attempt to highlight the differences and the in many case
   small differences in implementation to support a larger set of
   possible topologies.

   (tbd: This section needs reworking and clearer relation to

A.1.  Point to Point

   The point-to-point RTP topology (Figure 3) is the simplest scenario
   for WebRTC applications.  This is going to be very common for user to
   user calls.

                            +---+         +---+
                            | A |<------->| B |
                            +---+         +---+

                         Figure 3: Point to Point

   This being the basic one lets use the topology to high-light a couple
   of details that are common for all RTP usage in the WebRTC context.
   First is the intention to multiplex RTP and RTCP over the same UDP-
   flow.  Secondly is the question of using only a single RTP session or
   one per media type for legacy interoperability.  Thirdly is the
   question of using multiple sender sources (SSRCs) per end-point.

   Historically, RTP and RTCP have been run on separate UDP ports.  With
   the increased use of Network Address/Port Translation (NAPT) this has
   become problematic, since maintaining multiple NAT bindings can be
   costly.  It also complicates firewall administration, since multiple
   ports need to be opened to allow RTP traffic.  To reduce these costs
   and session set-up times, support for multiplexing RTP data packets
   and RTCP control packets on a single port [RFC5761] will be

   In cases where there is only one type of media (e.g., a voice-only
   call) this topology will be implemented as a single RTP session, with
   bidirectional flows of RTP and RTCP packets, all then multiplexed
   onto a single 5-tuple.  If multiple types of media are to be used
   (e.g., audio and video), then each type media can be sent as a
   separate RTP session using a different 5-tuple, allowing for separate
   transport level treatment of each type of media.  Alternatively, all
   types of media can be multiplexed onto a single 5-tuple as a single
   RTP session, or as several RTP sessions if using a demultiplexing
   shim.  Multiplexing different types of media onto a single 5-tuple
   places some limitations on how RTP is used, as described in "RTP
   Multiplexing Architecture"

Perkins, et al.         Expires January 16, 2014               [Page 39]

Internet-Draft               RTP for WebRTC                    July 2013

   [I-D.westerlund-avtcore-multiplex-architecture].  It is not expected
   that these limitations will significantly affect the scenarios
   targeted by WebRTC, but they can impact interoperability with legacy

   An RTP session have good support for simultaneously transport
   multiple media sources.  Each media source uses an unique SSRC
   identifier and each SSRC has independent RTP sequence number and
   timestamp spaces.  This is being utilized in WebRTC for several
   cases.  One is to enable multiple media sources of the same type, an
   end-point that has two video cameras can potentially transmit video
   from both to its peer(s).  Another usage is when a single RTP session
   is being used for both multiple media types, thus an end-point can
   transmit both audio and video to the peer(s).  Thirdly to support
   multi-party cases as will be discussed below support for multiple
   SSRC of the same media type is needed.

   Thus we can introduce a couple of different notations in the below
   two alternate figures of a single peer connection in a point to point
   set-up.  The first depicting a setup where the peer connection
   established has two different RTP sessions, one for audio and one for
   video.  The second one using a single RTP session.  In both cases A
   has two video streams to send and one audio stream.  B has only one
   audio and video stream.  These are used to illustrate the relation
   between a peerConnection, the UDP flow(s), the RTP session(s) and the
   SSRCs that will be used in the later cases also.  In the below
   figures RTCP flows are not included.  They will flow bi-directionally
   between any RTP session instances in the different nodes.

            +-A-------------+                 +-B-------------+
            | +-PeerC1------|                 |-PeerC1------+ |
            | | +-UDP1------|                 |-UDP1------+ | |
            | | | +-RTP1----|                 |-RTP1----+ | | |
            | | | | +-Audio-|                 |-Audio-+ | | | |
            | | | | |    AA1|---------------->|       | | | | |
            | | | | |       |<----------------|BA1    | | | | |
            | | | | +-------|                 |-------+ | | | |
            | | | +---------|                 |---------+ | | |
            | | +-----------|                 |-----------+ | |
            | |             |                 |             | |
            | | +-UDP2------|                 |-UDP2------+ | |
            | | | +-RTP2----|                 |-RTP1----+ | | |
            | | | | +-Video-|                 |-Video-+ | | | |
            | | | | |    AV1|---------------->|       | | | | |
            | | | | |    AV2|---------------->|       | | | | |
            | | | | |       |<----------------|BV1    | | | | |
            | | | | +-------|                 |-------+ | | | |
            | | | +---------|                 |---------+ | | |

Perkins, et al.         Expires January 16, 2014               [Page 40]

Internet-Draft               RTP for WebRTC                    July 2013

            | | +-----------|                 |-----------+ | |
            | +-------------|                 |-------------+ |
            +---------------+                 +---------------+

              Figure 4: Point to Point: Multiple RTP sessions

   As can be seen above in the Point to Point: Multiple RTP sessions
   (Figure 4) the single Peer Connection contains two RTP sessions over
   different UDP flows UDP 1 and UDP 2, i.e.  their 5-tuples will be
   different, normally on source and destination ports.  The first RTP
   session (RTP1) carries audio, one stream in each direction AA1 and
   BA1.  The second RTP session contains two video streams from A (AV1
   and AV2) and one from B to A (BV1).

            +-A-------------+                 +-B-------------+
            | +-PeerC1------|                 |-PeerC1------+ |
            | | +-UDP1------|                 |-UDP1------+ | |
            | | | +-RTP1----|                 |-RTP1----+ | | |
            | | | | +-Audio-|                 |-Audio-+ | | | |
            | | | | |    AA1|---------------->|       | | | | |
            | | | | |       |<----------------|BA1    | | | | |
            | | | | +-------|                 |-------+ | | | |
            | | | |         |                 |         | | | |
            | | | | +-Video-|                 |-Video-+ | | | |
            | | | | |    AV1|---------------->|       | | | | |
            | | | | |    AV2|---------------->|       | | | | |
            | | | | |       |<----------------|BV1    | | | | |
            | | | | +-------|                 |-------+ | | | |
            | | | +---------|                 |---------+ | | |
            | | +-----------|                 |-----------+ | |
            | +-------------|                 |-------------+ |
            +---------------+                 +---------------+

               Figure 5: Point to Point: Single RTP session.

   In (Figure 5) there is only a single UDP flow and RTP session (RTP1).
   This RTP session carries a total of five (5) RTP media streams
   (SSRCs).  From A to B there is Audio (AA1) and two video (AV1 and
   AV2).  From B to A there is Audio (BA1) and Video (BV1).

A.2.  Multi-Unicast (Mesh)

   For small multiparty calls, it is practical to set up a multi-unicast
   topology (Figure 6).  In this topology, each participant sends
   individual unicast RTP/UDP/IP flows to each of the other participants
   using independent PeerConnections in a full mesh.

Perkins, et al.         Expires January 16, 2014               [Page 41]

Internet-Draft               RTP for WebRTC                    July 2013

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

                          Figure 6: Multi-unicast

   This topology has the benefit of not requiring central nodes.  The
   downside is that it increases the used bandwidth at each sender by
   requiring one copy of the RTP media streams for each participant that
   are part of the same session beyond the sender itself.  Hence, this
   topology is limited to scenarios with few participants unless the
   media is very low bandwidth.  The multi-unicast topology could be
   implemented as a single RTP session, spanning multiple peer-to-peer
   transport layer connections, or as several pairwise RTP sessions, one
   between each pair of peers.  To maintain a coherent mapping between
   the relation between RTP sessions and PeerConnections we recommend
   that one implements this as individual RTP sessions.  The only
   downside is that end-point A will not learn of the quality of any
   transmission happening between B and C based on RTCP.  This has not
   been seen as a significant downside as now one has yet seen a need
   for why A would need to know about the B's and C's communication.  An
   advantage of using separate RTP sessions is that it enables using
   different media bit-rates to the different peers, thus not forcing B
   to endure the same quality reductions if there are limitations in the
   transport from A to C as C will.

        +-A------------------------+              +-B-------------+
        |+---+       +-PeerC1------|              |-PeerC1------+ |
        ||MIC|       | +-UDP1------|              |-UDP1------+ | |
        |+---+       | | +-RTP1----|              |-RTP1----+ | | |
        | |  +----+  | | | +-Audio-|              |-Audio-+ | | | |
        | +->|ENC1|--+-+-+-+--->AA1|------------->|       | | | | |
        | |  +----+  | | | |       |<-------------|BA1    | | | | |
        | |          | | | +-------|              |-------+ | | | |
        | |          | | +---------|              |---------+ | | |
        | |          | +-----------|              |-----------+ | |
        | |          +-------------|              |-------------+ |
        | |                        |              |---------------+
        | |                        |
        | |                        |              +-C-------------+
        | |          +-PeerC2------|              |-PeerC2------+ |

Perkins, et al.         Expires January 16, 2014               [Page 42]

Internet-Draft               RTP for WebRTC                    July 2013

        | |          | +-UDP2------|              |-UDP2------+ | |
        | |          | | +-RTP2----|              |-RTP2----+ | | |
        | |  +----+  | | | +-Audio-|              |-Audio-+ | | | |
        | +->|ENC2|--+-+-+-+--->AA2|------------->|       | | | | |
        |    +----+  | | | |       |<-------------|CA1    | | | | |
        |            | | | +-------|              |-------+ | | | |
        |            | | +---------|              |---------+ | | |
        |            | +-----------|              |-----------+ | |
        |            +-------------|              |-------------+ |
        +--------------------------+              +---------------+

            Figure 7: Session structure for Multi-Unicast Setup

   Lets review how the RTP sessions looks from A's perspective by
   considering both how the media is a handled and what PeerConnections
   and RTP sessions that are set-up in Figure 7.  A's microphone is
   captured and the digital audio can then be feed into two different
   encoder instances each beeing associated with two different
   PeerConnections (PeerC1 and PeerC2) each containing independent RTP
   sessions (RTP1 and RTP2).  The SSRCs in each RTP session will be
   completely independent and the media bit-rate produced by the encoder
   can also be tuned to address any congestion control requirements
   between A and B differently then for the path A to C.

   For media encodings which are more resource consuming, like video,
   one could expect that it will be common that end-points that are
   resource constrained will use a different implementation strategy
   where the encoder is shared between the different PeerConnections as
   shown below Figure 8.

       +-A----------------------+                 +-B-------------+
       |+---+                   |                 |               |
       ||CAM|     +-PeerC1------|                 |-PeerC1------+ |
       |+---+     | +-UDP1------|                 |-UDP1------+ | |
       |  |       | | +-RTP1----|                 |-RTP1----+ | | |
       |  V       | | | +-Video-|                 |-Video-+ | | | |
       |+----+    | | | |       |<----------------|BV1    | | | | |
       ||ENC |----+-+-+-+--->AV1|---------------->|       | | | | |
       |+----+    | | | +-------|                 |-------+ | | | |
       |  |       | | +---------|                 |---------+ | | |
       |  |       | +-----------|                 |-----------+ | |
       |  |       +-------------|                 |-------------+ |
       |  |                     |                 |---------------+
       |  |                     |
       |  |                     |                 +-C-------------+
       |  |       +-PeerC2------|                 |-PeerC2------+ |
       |  |       | +-UDP2------|                 |-UDP2------+ | |
       |  |       | | +-RTP2----|                 |-RTP2----+ | | |

Perkins, et al.         Expires January 16, 2014               [Page 43]

Internet-Draft               RTP for WebRTC                    July 2013

       |  |       | | | +-Video-|                 |-Video-+ | | | |
       |  +-------+-+-+-+--->AV2|---------------->|       | | | | |
       |          | | | |       |<----------------|CV1    | | | | |
       |          | | | +-------|                 |-------+ | | | |
       |          | | +---------|                 |---------+ | | |
       |          | +-----------|                 |-----------+ | |
       |          +-------------|                 |-------------+ |
       +------------------------+                 +---------------+

               Figure 8: Single Encoder Multi-Unicast Setup

   This will clearly save resources consumed by encoding but does
   introduce the need for the end-point A to make decisions on how it
   encodes the media so it suites delivery to both B and C.  This is not
   limited to congestion control, also preferred resolution to receive
   based on dispaly area available is another aspect requiring
   consideration.  The need for this type of decision logic does arise
   in several different topologies and implementation.

A.3.  Mixer Based

   An mixer (Figure 9) is a centralised point that selects or mixes
   content in a conference to optimise the RTP session so that each end-
   point only needs connect to one entity, the mixer.  The mixer can
   also reduce the bit-rate needed from the mixer down to a conference
   participants as the media sent from the mixer to the end-point can be
   optimised in different ways.  These optimisations include methods
   like only choosing media from the currently most active speaker or
   mixing together audio so that only one audio stream is needed instead
   of 3 in the depicted scenario (Figure 9).

                   +---+      +------------+      +---+
                   | A |<---->|            |<---->| B |
                   +---+      |            |      +---+
                              |   Mixer    |
                   +---+      |            |      +---+
                   | C |<---->|            |<---->| D |
                   +---+      +------------+      +---+

                Figure 9: RTP Mixer with Only Unicast Paths

   Mixers have two downsides, the first is that the mixer has to be a
   trusted node as they either performs media operations or at least re-
   packetize the media.  Both type of operations requires when using
   SRTP that the mixer verifies integrity, decrypts the content, perform
   its operation and form new RTP packets, encrypts and integrity
   protect them.  This applies to all types of mixers described below.

Perkins, et al.         Expires January 16, 2014               [Page 44]

Internet-Draft               RTP for WebRTC                    July 2013

   The second downside is that all these operations and optimization of
   the session requires processing.  How much depends on the
   implementation as will become evident below.

   The implementation of an mixer can take several different forms and
   we will discuss the main themes available that doesn't break RTP.

   Please note that a Mixer could also contain translator
   functionalities, like a media transcoder to adjust the media bit-rate
   or codec used on a particular RTP media stream.

A.3.1.  Media Mixing

   This type of mixer is one which clearly can be called RTP mixer is
   likely the one that most thinks of when they hear the term mixer.
   Its basic patter of operation is that it will receive the different
   participants RTP media stream.  Select which that are to be included
   in a media domain mix of the incoming RTP media streams.  Then create
   a single outgoing stream from this mix.

   Audio mixing is straight forward and commonly possible to do for a
   number of participants.  Lets assume that you want to mix N number of
   streams from different participants.  Then the mixer need to perform
   decoding N times.  Then it needs to produce N or N+1 mixes, the
   reasons that different mixes are needed are so that each contributing
   source get a mix which don't contain themselves, as this would result
   in an echo.  When N is lower than the number of all participants one
   can produce a Mix of all N streams for the group that are curently
   not included in the mix, thus N+1 mixes.  These audio streams are
   then encoded again, RTP packetized and sent out.

   Video can't really be "mixed" and produce something particular useful
   for the users, however creating an composition out of the contributed
   video streams can be done.  In fact it can be done in a number of
   ways, tiling the different streams creating a chessboard, selecting
   someone as more important and showing them large and a number of
   other sources as smaller is another.  Also here one commonly need to
   produce a number of different compositions so that the contributing
   part doesn't need to see themselves.  Then the mixer re-encodes the
   created video stream, RTP packetize it and send it out

   The problem with media mixing is that it both consume large amount of
   media processing and encoding resources.  The second is the quality
   degradation created by decoding and re-encoding the RTP media stream.
   Its advantage is that it is quite simplistic for the clients to
   handle as they don't need to handle local mixing and composition.

Perkins, et al.         Expires January 16, 2014               [Page 45]

Internet-Draft               RTP for WebRTC                    July 2013

     +-A-------------+             +-MIXER--------------------------+
     | +-PeerC1------|             |-PeerC1--------+                |
     | | +-UDP1------|             |-UDP1--------+ |                |
     | | | +-RTP1----|             |-RTP1------+ | |        +-----+ |
     | | | | +-Audio-|             |-Audio---+ | | | +---+  |     | |
     | | | | |    AA1|------------>|---------+-+-+-+-|DEC|->|     | |
     | | | | |       |<------------|MA1 <----+ | | | +---+  |     | |
     | | | | |       |             |(BA1+CA1)|\| | | +---+  |     | |
     | | | | +-------|             |---------+ +-+-+-|ENC|<-| B+C | |
     | | | +---------|             |-----------+ | | +---+  |     | |
     | | +-----------|             |-------------+ |        |  M  | |
     | +-------------|             |---------------+        |  E  | |
     +---------------+             |                        |  D  | |
                                   |                        |  I  | |
     +-B-------------+             |                        |  A  | |
     | +-PeerC2------|             |-PeerC2--------+        |     | |
     | | +-UDP2------|             |-UDP2--------+ |        |  M  | |
     | | | +-RTP2----|             |-RTP2------+ | |        |  I  | |
     | | | | +-Audio-|             |-Audio---+ | | | +---+  |  X  | |
     | | | | |    BA1|------------>|---------+-+-+-+-|DEC|->|  E  | |
     | | | | |       |<------------|MA2 <----+ | | | +---+  |  R  | |
     | | | | +-------|             |(BA1+CA1)|\| | | +---+  |     | |
     | | | +---------|             |---------+ +-+-+-|ENC|<-| A+C | |
     | | +-----------|             |-----------+ | | +---+  |     | |
     | +-------------|             |-------------+ |        |     | |
     +---------------+             |---------------+        |     | |
                                   |                        |     | |
     +-C-------------+             |                        |     | |
     | +-PeerC3------|             |-PeerC3--------+        |     | |
     | | +-UDP3------|             |-UDP3--------+ |        |     | |
     | | | +-RTP3----|             |-RTP3------+ | |        |     | |
     | | | | +-Audio-|             |-Audio---+ | | | +---+  |     | |
     | | | | |    CA1|------------>|---------+-+-+-+-|DEC|->|     | |
     | | | | |       |<------------|MA3 <----+ | | | +---+  |     | |
     | | | | +-------|             |(BA1+CA1)|\| | | +---+  |     | |
     | | | +---------|             |---------+ +-+-+-|ENC|<-| A+B | |
     | | +-----------|             |-----------+ | | +---+  |     | |
     | +-------------|             |-------------+ |        +-----+ |
     +---------------+             |---------------+                |

            Figure 10: Session and SSRC details for Media Mixer

   From an RTP perspective media mixing can be very straight forward as
   can be seen in Figure 10.  The mixer present one SSRC towards the
   peer client, e.g.  MA1 to Peer A, which is the media mix of the other
   participants.  As each peer receives a different version produced by
   the mixer there are no actual relation between the different RTP

Perkins, et al.         Expires January 16, 2014               [Page 46]

Internet-Draft               RTP for WebRTC                    July 2013

   sessions in the actual media or the transport level information.
   There is however one connection between RTP1-RTP3 in this figure.  It
   has to do with the SSRC space and the identity information.  When A
   receives the MA1 stream which is a combination of BA1 and CA1 streams
   in the other PeerConnections RTP could enable the mixer to include
   CSRC information in the MA1 stream to identify the contributing
   source BA1 and CA1.

   The CSRC has in its turn utility in RTP extensions, like the in
   Section 5.2.3 discussed Mixer to Client audio levels RTP header
   extension [RFC6465].  If the SSRC from one PeerConnection are used as
   CSRC in another PeerConnection then RTP1, RTP2 and RTP3 becomes one
   joint session as they have a common SSRC space.  At this stage one
   also need to consider which RTCP information one need to expose in
   the different legs.  For the above situation commonly nothing more
   than the Source Description (SDES) information and RTCP BYE for CSRC
   need to be exposed.  The main goal would be to enable the correct
   binding against the application logic and other information sources.
   This also enables loop detection in the RTP session.

A.3.1.1.  RTP Session Termination

   There exist an possible implementation choice to have the RTP
   sessions being separated between the different legs in the multi-
   party communication session and only generate RTP media streams in
   each without carrying on RTP/RTCP level any identity information
   about the contributing sources.  This removes both the functionality
   that CSRC can provide and the possibility to use any extensions that
   build on CSRC and the loop detection.  It might appear a
   simplification if SSRC collision would occur between two different
   end-points as they can be avoided to be resolved and instead remapped
   between the independent sessions if at all exposed.  However, SSRC/
   CSRC remapping requires that SSRC/CSRC are never exposed to the
   WebRTC JavaScript client to use as reference.  This as they only have
   local importance if they are used on a multi-party session scope the
   result would be mis-referencing.  Also SSRC collision handling will
   still be needed as it can occur between the mixer and the end-point.

   Session termination might appear to resolve some issues, it however
   creates other issues that needs resolving, like loop detection,
   identification of contributing sources and the need to handle mapped
   identities and ensure that the right one is used towards the right
   identities and never used directly between multiple end-points.

A.3.2.  Media Switching

   An RTP Mixer based on media switching avoids the media decoding and
   encoding cycle in the mixer, but not the decryption and re-encryption

Perkins, et al.         Expires January 16, 2014               [Page 47]

Internet-Draft               RTP for WebRTC                    July 2013

   cycle as one rewrites RTP headers.  This both reduces the amount of
   computational resources needed in the mixer and increases the media
   quality per transmitted bit.  This is achieve by letting the mixer
   have a number of SSRCs that represents conceptual or functional
   streams the mixer produces.  These streams are created by selecting
   media from one of the by the mixer received RTP media streams and
   forward the media using the mixers own SSRCs.  The mixer can then
   switch between available sources if that is needed by the concept for
   the source, like currently active speaker.

   To achieve a coherent RTP media stream from the mixer's SSRC the
   mixer is forced to rewrite the incoming RTP packet's header.  First
   the SSRC field has to be set to the value of the Mixer's SSRC.
   Secondly, the sequence number is set to the next in the sequence of
   outgoing packets it sent.  Thirdly the RTP timestamp value needs to
   be adjusted using an offset that changes each time one switch media
   source.  Finally depending on the negotiation the RTP payload type
   value representing this particular RTP payload configuration might
   have to be changed if the different PeerConnections have not arrived
   on the same numbering for a given configuration.  This also requires
   that the different end-points do support a common set of codecs,
   otherwise media transcoding for codec compatibility is still needed.

   Lets consider the operation of media switching mixer that supports a
   video conference with six participants (A-F) where the two latest
   speakers in the conference are shown to each participants.  Thus the
   mixer has two SSRCs sending video to each peer.

     +-A-------------+             +-MIXER--------------------------+
     | +-PeerC1------|             |-PeerC1--------+                |
     | | +-UDP1------|             |-UDP1--------+ |                |
     | | | +-RTP1----|             |-RTP1------+ | |        +-----+ |
     | | | | +-Video-|             |-Video---+ | | |        |     | |
     | | | | |    AV1|------------>|---------+-+-+-+------->|     | |
     | | | | |       |<------------|MV1 <----+-+-+-+-BV1----|     | |
     | | | | |       |<------------|MV2 <----+-+-+-+-EV1----|     | |
     | | | | +-------|             |---------+ | | |        |     | |
     | | | +---------|             |-----------+ | |        |     | |
     | | +-----------|             |-------------+ |        |  S  | |
     | +-------------|             |---------------+        |  W  | |
     +---------------+             |                        |  I  | |
                                   |                        |  T  | |
     +-B-------------+             |                        |  C  | |
     | +-PeerC2------|             |-PeerC2--------+        |  H  | |
     | | +-UDP2------|             |-UDP2--------+ |        |     | |
     | | | +-RTP2----|             |-RTP2------+ | |        |  M  | |
     | | | | +-Video-|             |-Video---+ | | |        |  A  | |
     | | | | |    BV1|------------>|---------+-+-+-+------->|  T  | |

Perkins, et al.         Expires January 16, 2014               [Page 48]

Internet-Draft               RTP for WebRTC                    July 2013

     | | | | |       |<------------|MV3 <----+-+-+-+-AV1----|  R  | |
     | | | | |       |<------------|MV4 <----+-+-+-+-EV1----|  I  | |
     | | | | +-------|             |---------+ | | |        |  X  | |
     | | | +---------|             |-----------+ | |        |     | |
     | | +-----------|             |-------------+ |        |     | |
     | +-------------|             |---------------+        |     | |
     +---------------+             |                        |     | |
                                   :                        :     : :
                                   :                        :     : :
     +-F-------------+             |                        |     | |
     | +-PeerC6------|             |-PeerC6--------+        |     | |
     | | +-UDP6------|             |-UDP6--------+ |        |     | |
     | | | +-RTP6----|             |-RTP6------+ | |        |     | |
     | | | | +-Video-|             |-Video---+ | | |        |     | |
     | | | | |    CV1|------------>|---------+-+-+-+------->|     | |
     | | | | |       |<------------|MV11 <---+-+-+-+-AV1----|     | |
     | | | | |       |<------------|MV12 <---+-+-+-+-EV1----|     | |
     | | | | +-------|             |---------+ | | |        |     | |
     | | | +---------|             |-----------+ | |        |     | |
     | | +-----------|             |-------------+ |        +-----+ |
     | +-------------|             |---------------+                |
     +---------------+             +--------------------------------+

                   Figure 11: Media Switching RTP Mixer

   The Media Switching RTP mixer can similar to the Media Mixing one
   reduce the bit-rate needed towards the different peers by selecting
   and switching in a sub-set of RTP media streams out of the ones it
   receives from the conference participations.

   To ensure that a media receiver can correctly decode the RTP media
   stream after a switch, it becomes necessary to ensure for state
   saving codecs that they start from default state at the point of
   switching.  Thus one common tool for video is to request that the
   encoding creates an intra picture, something that isn't dependent on
   earlier state.  This can be done using Full Intra Request RTCP codec
   control message as discussed in Section 5.1.1.

   Also in this type of mixer one could consider to terminate the RTP
   sessions fully between the different PeerConnection.  The same
   arguments and considerations as discussed in Appendix A.3.1.1 applies

Perkins, et al.         Expires January 16, 2014               [Page 49]

Internet-Draft               RTP for WebRTC                    July 2013

A.3.3.  Media Projecting

   Another method for handling media in the RTP mixer is to project all
   potential sources (SSRCs) into a per end-point independent RTP
   session.  The mixer can then select which of the potential sources
   that are currently actively transmitting media, despite that the
   mixer in another RTP session receives media from that end-point.
   This is similar to the media switching Mixer but have some important
   differences in RTP details.

     +-A-------------+             +-MIXER--------------------------+
     | +-PeerC1------|             |-PeerC1--------+                |
     | | +-UDP1------|             |-UDP1--------+ |                |
     | | | +-RTP1----|             |-RTP1------+ | |        +-----+ |
     | | | | +-Video-|             |-Video---+ | | |        |     | |
     | | | | |    AV1|------------>|---------+-+-+-+------->|     | |
     | | | | |       |<------------|BV1 <----+-+-+-+--------|     | |
     | | | | |       |<------------|CV1 <----+-+-+-+--------|     | |
     | | | | |       |<------------|DV1 <----+-+-+-+--------|     | |
     | | | | |       |<------------|EV1 <----+-+-+-+--------|     | |
     | | | | |       |<------------|FV1 <----+-+-+-+--------|     | |
     | | | | +-------|             |---------+ | | |        |     | |
     | | | +---------|             |-----------+ | |        |     | |
     | | +-----------|             |-------------+ |        |  S  | |
     | +-------------|             |---------------+        |  W  | |
     +---------------+             |                        |  I  | |
                                   |                        |  T  | |
     +-B-------------+             |                        |  C  | |
     | +-PeerC2------|             |-PeerC2--------+        |  H  | |
     | | +-UDP2------|             |-UDP2--------+ |        |     | |
     | | | +-RTP2----|             |-RTP2------+ | |        |  M  | |
     | | | | +-Video-|             |-Video---+ | | |        |  A  | |
     | | | | |    BV1|------------>|---------+-+-+-+------->|  T  | |
     | | | | |       |<------------|AV1 <----+-+-+-+--------|  R  | |
     | | | | |       |<------------|CV1 <----+-+-+-+--------|  I  | |
     | | | | |       | :    :    : |: :  : : : : : :  :  : :|  X  | |
     | | | | |       |<------------|FV1 <----+-+-+-+--------|     | |
     | | | | +-------|             |---------+ | | |        |     | |
     | | | +---------|             |-----------+ | |        |     | |
     | | +-----------|             |-------------+ |        |     | |
     | +-------------|             |---------------+        |     | |
     +---------------+             |                        |     | |
                                   :                        :     : :
                                   :                        :     : :
     +-F-------------+             |                        |     | |
     | +-PeerC6------|             |-PeerC6--------+        |     | |
     | | +-UDP6------|             |-UDP6--------+ |        |     | |
     | | | +-RTP6----|             |-RTP6------+ | |        |     | |

Perkins, et al.         Expires January 16, 2014               [Page 50]

Internet-Draft               RTP for WebRTC                    July 2013

     | | | | +-Video-|             |-Video---+ | | |        |     | |
     | | | | |    CV1|------------>|---------+-+-+-+------->|     | |
     | | | | |       |<------------|AV1 <----+-+-+-+--------|     | |
     | | | | |       | :    :    : |: :  : : : : : :  :  : :|     | |
     | | | | |       |<------------|EV1 <----+-+-+-+--------|     | |
     | | | | +-------|             |---------+ | | |        |     | |
     | | | +---------|             |-----------+ | |        |     | |
     | | +-----------|             |-------------+ |        +-----+ |
     | +-------------|             |---------------+                |
     +---------------+             +--------------------------------+

                     Figure 12: Media Projecting Mixer

   So in this six participant conference depicted above in (Figure 12)
   one can see that end-point A will in this case be aware of 5 incoming
   SSRCs, BV1-FV1.  If this mixer intend to have the same behavior as in
   Appendix A.3.2 where the mixer provides the end-points with the two
   latest speaking end-points, then only two out of these five SSRCs
   will concurrently transmit media to A.  As the mixer selects which
   source in the different RTP sessions that transmit media to the end-
   points each RTP media stream will require some rewriting when being
   projected from one session into another.  The main thing is that the
   sequence number will need to be consecutively incremented based on
   the packet actually being transmitted in each RTP session.  Thus the
   RTP sequence number offset will change each time a source is turned
   on in RTP session.

   As the RTP sessions are independent the SSRC numbers used can be
   handled independently also thus working around any SSRC collisions by
   having remapping tables between the RTP sessions.  However the
   related WebRTC MediaStream signalling need to be correspondingly
   changed to ensure consistent WebRTC MediaStream to SSRC mappings
   between the different PeerConnections and the same comment that
   higher functions MUST NOT use SSRC as references to RTP media streams
   applies also here.

   The mixer will also be responsible to act on any RTCP codec control
   requests coming from an end-point and decide if it can act on it
   locally or needs to translate the request into the RTP session that
   contains the media source.  Both end-points and the mixer will need
   to implement conference related codec control functionalities to
   provide a good experience.  Full Intra Request to request from the
   media source to provide switching points between the sources,
   Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
   to aggregate congestion control response towards the media source and
   have it adjust its bit-rate in case the limitation is not in the
   source to mixer link.

Perkins, et al.         Expires January 16, 2014               [Page 51]

Internet-Draft               RTP for WebRTC                    July 2013

   This version of the mixer also puts different requirements on the
   end-point when it comes to decoder instances and handling of the RTP
   media streams providing media.  As each projected SSRC can at any
   time provide media the end-point either needs to handle having thus
   many allocated decoder instances or have efficient switching of
   decoder contexts in a more limited set of actual decoder instances to
   cope with the switches.  The WebRTC application also gets more
   responsibility to update how the media provides is to be presented to
   the user.

A.4.  Translator Based

   There is also a variety of translators.  The core commonality is that
   they do not need to make themselves visible in the RTP level by
   having an SSRC themselves.  Instead they sit between one or more end-
   point and perform translation at some level.  It can be media
   transcoding, protocol translation or covering missing functionality
   for a legacy end-point or simply relay packets between transport
   domains or to realize multi-party.  We will go in details below.

A.4.1.  Transcoder

   A transcoder operates on media level and really used for two
   purposes, the first is to allow two end-points that doesn't have a
   common set of media codecs to communicate by translating from one
   codec to another.  The second is to change the bit-rate to a lower
   one.  For WebRTC end-points communicating with each other only the
   first one is relevant.  In certain legacy deployment media transcoder
   will be necessary to ensure both codecs and bit-rate falls within the
   envelope the legacy end-point supports.

   As transcoding requires access to the media, the transcoder has to be
   within the security context and access any media encryption and
   integrity keys.  On the RTP plane a media transcoder will in practice
   fork the RTP session into two different domains that are highly
   decoupled when it comes to media parameters and reporting, but not
   identities.  To maintain signalling bindings to SSRCs a transcoder is
   likely needing to use the SSRC of one end-point to represent the
   transcoded RTP media stream to the other end-point(s).  The
   congestion control loop can be terminated in the transcoder as the
   media bit-rate being sent by the transcoder can be adjusted
   independently of the incoming bit-rate.  However, for optimizing
   performance and resource consumption the translator needs to consider
   what signals or bit-rate reductions it needs to send towards the
   source end-point.  For example receiving a 2.5 Mbps video stream and
   then send out a 250 kbps video stream after transcoding is a waste of
   resources.  In most cases a 500 kbps video stream from the source in
   the right resolution is likely to provide equal quality after

Perkins, et al.         Expires January 16, 2014               [Page 52]

Internet-Draft               RTP for WebRTC                    July 2013

   transcoding as the 2.5 Mbps source stream.  At the same time
   increasing media bit-rate further than what is needed to represent
   the incoming quality accurate is also wasted resources.

       +-A-------------+             +-Translator------------------+
       | +-PeerC1------|             |-PeerC1--------+             |
       | | +-UDP1------|             |-UDP1--------+ |             |
       | | | +-RTP1----|             |-RTP1------+ | |             |
       | | | | +-Audio-|             |-Audio---+ | | | +---+       |
       | | | | |    AA1|------------>|---------+-+-+-+-|DEC|----+  |
       | | | | |       |<------------|BA1 <----+ | | | +---+    |  |
       | | | | |       |             |         |\| | | +---+    |  |
       | | | | +-------|             |---------+ +-+-+-|ENC|<-+ |  |
       | | | +---------|             |-----------+ | | +---+  | |  |
       | | +-----------|             |-------------+ |        | |  |
       | +-------------|             |---------------+        | |  |
       +---------------+             |                        | |  |
                                     |                        | |  |
       +-B-------------+             |                        | |  |
       | +-PeerC2------|             |-PeerC2--------+        | |  |
       | | +-UDP2------|             |-UDP2--------+ |        | |  |
       | | | +-RTP1----|             |-RTP1------+ | |        | |  |
       | | | | +-Audio-|             |-Audio---+ | | | +---+  | |  |
       | | | | |    BA1|------------>|---------+-+-+-+-|DEC|--+ |  |
       | | | | |       |<------------|AA1 <----+ | | | +---+    |  |
       | | | | |       |             |         |\| | | +---+    |  |
       | | | | +-------|             |---------+ +-+-+-|ENC|<---+  |
       | | | +---------|             |-----------+ | | +---+       |
       | | +-----------|             |-------------+ |             |
       | +-------------|             |---------------+             |
       +---------------+             +-----------------------------+

                        Figure 13: Media Transcoder

   Figure 13 exposes some important details.  First of all you can see
   the SSRC identifiers used by the translator are the corresponding
   end-points.  Secondly, there is a relation between the RTP sessions
   in the two different PeerConnections that are represented by having
   both parts be identified by the same level and they need to share
   certain contexts.  Also certain type of RTCP messages will need to be
   bridged between the two parts.  Certain RTCP feedback messages are
   likely needed to be sourced by the translator in response to actions
   by the translator and its media encoder.

A.4.2.  Gateway / Protocol Translator

   Gateways are used when some protocol feature that are needed are not
   supported by an end-point wants to participate in session.  This RTP

Perkins, et al.         Expires January 16, 2014               [Page 53]

Internet-Draft               RTP for WebRTC                    July 2013

   translator in Figure 14 takes on the role of ensuring that from the
   perspective of participant A, participant B appears as a fully
   compliant WebRTC end-point (that is, it is the combination of the
   Translator and participant B that looks like a WebRTC end point).

                              |            |
                   +---+      | Translator |      +---+
                   | A |<---->| to legacy  |<---->| B |
                   +---+      | end-point  |      +---+
                   WebRTC     |            |     Legacy

       Figure 14: Gateway (RTP translator) towards legacy end-point

   For WebRTC there are a number of requirements that could force the
   need for a gateway if a WebRTC end-point is to communicate with a
   legacy end-point, such as support of ICE and DTLS-SRTP for key
   management.  On RTP level the main functions that might be missing in
   a legacy implementation that otherwise support RTP are RTCP in
   general, SRTP implementation, congestion control and feedback
   messages needed to make it work.

       +-A-------------+             +-Translator------------------+
       | +-PeerC1------|             |-PeerC1------+               |
       | | +-UDP1------|             |-UDP1------+ |               |
       | | | +-RTP1----|             |-RTP1-----------------------+|
       | | | | +-Audio-|             |-Audio---+                  ||
       | | | | |    AA1|------------>|---------+----------------+ ||
       | | | | |       |<------------|BA1 <----+--------------+ | ||
       | | | | |       |<---RTCP---->|<--------+----------+   | | ||
       | | | | +-------|             |---------+      +---+-+ | | ||
       | | | +---------|             |---------------+| T   | | | ||
       | | +-----------|             |-----------+ | || R   | | | ||
       | +-------------|             |-------------+ || A   | | | ||
       +---------------+             |               || N   | | | ||
                                     |               || S   | | | ||
       +-B-(Legacy)----+             |               || L   | | | ||
       |               |             |               || A   | | | ||
       |   +-UDP2------|             |-UDP2------+   || T   | | | ||
       |   | +-RTP1----|             |-RTP1----------+| E   | | | ||
       |   | | +-Audio-|             |-Audio---+      +---+-+ | | ||
       |   | | |       |<---RTCP---->|<--------+----------+   | | ||
       |   | | |    BA1|------------>|---------+--------------+ | ||
       |   | | |       |<------------|AA1 <----+----------------+ ||
       |   | | +-------|             |---------+                  ||
       |   | +---------|             |----------------------------+|
       |   +-----------|             |-----------+                 |

Perkins, et al.         Expires January 16, 2014               [Page 54]

Internet-Draft               RTP for WebRTC                    July 2013

       |               |             |                             |
       +---------------+             +-----------------------------+

                  Figure 15: RTP/RTCP Protocol Translator

   The legacy gateway can be implemented in several ways and what it
   need to change is highly dependent on what functions it need to proxy
   for the legacy end-point.  One possibility is depicted in Figure 15
   where the RTP media streams are compatible and forward without
   changes.  However, their RTP header values are captured to enable the
   RTCP translator to create RTCP reception information related to the
   leg between the end-point and the translator.  This can then be
   combined with the more basic RTCP reports that the legacy endpoint
   (B) provides to give compatible and expected RTCP reporting to A.
   Thus enabling at least full congestion control on the path between A
   and the translator.  If B has limited possibilities for congestion
   response for the media then the translator might need the capability
   to perform media transcoding to address cases where it otherwise
   would need to terminate media transmission.

   As the translator are generating RTP/RTCP traffic on behalf of B to A
   it will need to be able to correctly protect these packets that it
   translates or generates.  Thus security context information are
   needed in this type of translator if it operates on the RTP/RTCP
   packet content or media.  In fact one of the more likely scenario is
   that the translator (gateway) will need to have two different
   security contexts one towards A and one towards B and for each RTP/
   RTCP packet do a authenticity verification, decryption followed by a
   encryption and integrity protection operation to resolve mismatch in
   security systems.

A.4.3.  Relay

   There exist a class of translators that operates on transport level
   below RTP and thus do not effect RTP/RTCP packets directly.  They
   come in two distinct flavours, the one used to bridge between two
   different transport or address domains to more function as a gateway
   and the second one which is to to provide a group communication
   feature as depicted below in Figure 16.

                   +---+      +------------+      +---+
                   | A |<---->|            |<---->| B |
                   +---+      |            |      +---+
                              | Translator |
                   +---+      |            |      +---+
                   | C |<---->|            |<---->| D |
                   +---+      +------------+      +---+

Perkins, et al.         Expires January 16, 2014               [Page 55]

Internet-Draft               RTP for WebRTC                    July 2013

         Figure 16: RTP Translator (Relay) with Only Unicast Paths

   The first kind is straight forward and is likely to exist in WebRTC
   context when an legacy end-point is compatible with the exception for
   ICE, and thus needs a gateway that terminates the ICE and then
   forwards all the RTP/RTCP traffic and key management to the end-point
   only rewriting the IP/UDP to forward the packet to the legacy node.

   The second type is useful if one wants a less complex central node or
   a central node that is outside of the security context and thus do
   not have access to the media.  This relay takes on the role of
   forwarding the media (RTP and RTCP) packets to the other end-points
   but doesn't perform any RTP or media processing.  Such a device
   simply forwards the media from each sender to all of the other
   participants, and is sometimes called a transport-layer translator.
   In Figure 16, participant A will only need to send a media once to
   the relay, which will redistribute it by sending a copy of the stream
   to participants B, C, and D.  Participant A will still receive three
   RTP streams with the media from B, C and D if they transmit
   simultaneously.  This is from an RTP perspective resulting in an RTP
   session that behaves equivalent to one transporter over an IP Any
   Source Multicast (ASM).

   This results in one common RTP session between all participants
   despite that there will be independent PeerConnections created to the
   translator as depicted below Figure 17.

     +-A-------------+             +-RELAY--------------------------+
     | +-PeerC1------|             |-PeerC1--------+                |
     | | +-UDP1------|             |-UDP1--------+ |                |
     | | | +-RTP1----|             |-RTP1-------------------------+ |
     | | | | +-Video-|             |-Video---+                    | |
     | | | | |    AV1|------------>|---------------------------+  | |
     | | | | |       |<------------|BV1 <--------------------+ |  | |
     | | | | |       |<------------|CV1 <------------------+ | |  | |
     | | | | +-------|             |---------+             | | |  | |
     | | | +---------|             |-------------------+   ^ ^ V  | |
     | | +-----------|             |-------------+ |   |   | | |  | |
     | +-------------|             |---------------+   |   | | |  | |
     +---------------+             |                   |   | | |  | |
                                   |                   |   | | |  | |
     +-B-------------+             |                   |   | | |  | |
     | +-PeerC2------|             |-PeerC2--------+   |   | | |  | |
     | | +-UDP2------|             |-UDP2--------+ |   |   | | |  | |
     | | | +-RTP2----|             |-RTP1--------------+   | | |  | |
     | | | | +-Video-|             |-Video---+             | | |  | |
     | | | | |    BV1|------------>|-----------------------+ | |  | |
     | | | | |       |<------------|AV1 <----------------------+  | |

Perkins, et al.         Expires January 16, 2014               [Page 56]

Internet-Draft               RTP for WebRTC                    July 2013

     | | | | |       |<------------|CV1 <--------------------+ |  | |
     | | | | +-------|             |---------+             | | |  | |
     | | | +---------|             |-------------------+   | | |  | |
     | | +-----------|             |-------------+ |   |   V ^ V  | |
     | +-------------|             |---------------+   |   | | |  | |
     +---------------+             |                   |   | | |  | |
                                   :                   |   | | |  | |
                                   :                   |   | | |  | |
     +-C-------------+             |                   |   | | |  | |
     | +-PeerC3------|             |-PeerC3--------+   |   | | |  | |
     | | +-UDP3------|             |-UDP3--------+ |   |   | | |  | |
     | | | +-RTP3----|             |-RTP1--------------+   | | |  | |
     | | | | +-Video-|             |-Video---+             | | |  | |
     | | | | |    CV1|------------>|-------------------------+ |  | |
     | | | | |       |<------------|AV1 <----------------------+  | |
     | | | | |       |<------------|BV1 <------------------+      | |
     | | | | +-------|             |---------+                    | |
     | | | +---------|             |------------------------------+ |
     | | +-----------|             |-------------+ |                |
     | +-------------|             |---------------+                |
     +---------------+             +--------------------------------+

                  Figure 17: Transport Multi-party Relay

   As the Relay RTP and RTCP packets between the UDP flows as indicated
   by the arrows for the media flow a given WebRTC end-point, like A
   will see the remote sources BV1 and CV1.  There will be also two
   different network paths between A, and B or C.  This results in that
   the client A has to be capable of handling that when determining
   congestion state that there might exist multiple destinations on the
   far side of a PeerConnection and that these paths have to be treated
   differently.  It also results in a requirement to combine the
   different congestion states into a decision to transmit a particular
   RTP media stream suitable to all participants.

   It is also important to note that the relay can not perform selective
   relaying of some sources and not others.  The reason is that the RTCP
   reporting in that case becomes inconsistent and without explicit
   information about it being blocked has to be interpreted as severe

   In this usage it is also necessary that the session management has
   configured a common set of RTP configuration including RTP payload
   formats as when A sends a packet with pt=97 it will arrive at both B
   and C carrying pt=97 and having the same packetization and encoding,
   no entity will have manipulated the packet.

Perkins, et al.         Expires January 16, 2014               [Page 57]

Internet-Draft               RTP for WebRTC                    July 2013

   When it comes to security there exist some additional requirements to
   ensure that the property that the relay can't read the media traffic
   is enforced.  First of all the key to be used has to be agreed such
   so that the relay doesn't get it, e.g.  no DTLS-SRTP handshake with
   the relay, instead some other method needs to be used.  Secondly, the
   keying structure has to be capable of handling multiple end-points in
   the same RTP session.

   The second problem can basically be solved in two ways.  Either a
   common master key from which all derive their per source key for
   SRTP.  The second alternative which might be more practical is that
   each end-point has its own key used to protects all RTP/RTCP packets
   it sends.  Each participants key are then distributed to the other
   participants.  This second method could be implemented using DTLS-
   SRTP to a special key server and then use Encrypted Key Transport
   [I-D.ietf-avt-srtp-ekt] to distribute the actual used key to the
   other participants in the RTP session Figure 18.  The first one could
   be achieved using MIKEY messages in SDP.

                 +---+                               +---+
                 |   |         +-----------+         |   |
                 | A |<------->| DTLS-SRTP |<------->| C |
                 |   |<--   -->|   HOST    |<--   -->|   |
                 +---+   \ /   +-----------+   \ /   +---+
                          X                     X
                 +---+   / \   +-----------+   / \   +---+
                 |   |<--   -->|    RTP    |<--   -->|   |
                 | B |<------->|   RELAY   |<------->| D |
                 |   |         +-----------+         |   |
                 +---+                               +---+

             Figure 18: DTLS-SRTP host and RTP Relay Separated

   The relay can still verify that a given SSRC isn't used or spoofed by
   another participant within the multi-party session by binding SSRCs
   on their first usage to a given source address and port pair.
   Packets carrying that source SSRC from other addresses can be
   suppressed to prevent spoofing.  This is possible as long as SRTP is
   used which leaves the SSRC of the packet originator in RTP and RTCP
   packets in the clear.  If such packet level method for enforcing
   source authentication within the group, then there exist
   cryptographic methods such as TESLA [RFC4383] that could be used for
   true source authentication.

A.5.  End-point Forwarding

   An WebRTC end-point (B in Figure 19) will receive a WebRTC
   MediaStream (set of SSRCs) over a PeerConnection (from A).  For the

Perkins, et al.         Expires January 16, 2014               [Page 58]

Internet-Draft               RTP for WebRTC                    July 2013

   moment is not decided if the end-point is allowed or not to in its
   turn send that WebRTC MediaStream over another PeerConnection to C.
   This section discusses the RTP and end-point implications of allowing
   such functionality, which on the API level is extremely simplistic to

                          +---+    +---+    +---+
                          | A |--->| B |--->| C |
                          +---+    +---+    +---+

                     Figure 19: MediaStream Forwarding

   There exist two main approaches to how B forwards the media from A to
   C.  The first one is to simply relay the RTP media stream.  The
   second one is for B to act as a transcoder.  Lets consider both

   A relay approach will result in that the WebRTC end-points will have
   to have the same capabilities as being discussed in Relay
   (Appendix A.4.3).  Thus A will see an RTP session that is extended
   beyond the PeerConnection and see two different receiving end-points
   with different path characteristics (B and C).  Thus A's congestion
   control needs to be capable of handling this.  The security solution
   can either support mechanism that allows A to inform C about the key
   A is using despite B and C having agreed on another set of keys.
   Alternatively B will decrypt and then re-encrypt using a new key.
   The relay based approach has the advantage that B does not need to
   transcode the media thus both maintaining the quality of the encoding
   and reducing B's complexity requirements.  If the right security
   solutions are supported then also C will be able to verify the
   authenticity of the media coming from A.  As downside A are forced to
   take both B and C into consideration when delivering content.

   The media transcoder approach is similar to having B act as Mixer
   terminating the RTP session combined with the transcoder as discussed
   in Appendix A.4.1.  A will only see B as receiver of its media.  B
   will responsible to produce a RTP media stream suitable for the B to
   C PeerConnection.  This might require media transcoding for
   congestion control purpose to produce a suitable bit-rate.  Thus
   loosing media quality in the transcoding and forcing B to spend the
   resource on the transcoding.  The media transcoding does result in a
   separation of the two different legs removing almost all
   dependencies.  B could choice to implement logic to optimize its
   media transcoding operation, by for example requesting media
   properties that are suitable for C also, thus trying to avoid it
   having to transcode the content and only forward the media payloads
   between the two sides.  For that optimization to be practical WebRTC
   end-points have to support sufficiently good tools for codec control.

Perkins, et al.         Expires January 16, 2014               [Page 59]

Internet-Draft               RTP for WebRTC                    July 2013

A.6.  Simulcast

   This section discusses simulcast in the meaning of providing a node,
   for example a stream switching Mixer, with multiple different encoded
   version of the same media source.  In the WebRTC context that appears
   to be most easily accomplished by establishing multiple
   PeerConnection all being feed the same set of WebRTC MediaStreams.
   Each PeerConnection is then configured to deliver a particular media
   quality and thus media bit-rate.  This will work well as long as the
   end-point implements media encoding according to Figure 7.  Then each
   PeerConnection will receive an independently encoded version and the
   codec parameters can be agreed specifically in the context of this

   For simulcast to work one needs to prevent that the end-point deliver
   content encoded as depicted in Figure 8.  If a single encoder
   instance is feed to multiple PeerConnections the intention of
   performing simulcast will fail.

   Thus it needs to be considered to explicitly signal which of the two
   implementation strategies that are desired and which will be done.
   At least making the application and possible the central node
   interested in receiving simulcast of an end-points RTP media streams
   to be aware if it will function or not.

Authors' Addresses

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

   Email: csp@csperkins.org

   Magnus Westerlund
   Farogatan 6
   SE-164 80 Kista

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

Perkins, et al.         Expires January 16, 2014               [Page 60]

Internet-Draft               RTP for WebRTC                    July 2013

   Joerg Ott
   Aalto University
   School of Electrical Engineering
   Espoo  02150

   Email: jorg.ott@aalto.fi

Perkins, et al.         Expires January 16, 2014               [Page 61]

Html markup produced by rfcmarkup 1.129b, available from https://tools.ietf.org/tools/rfcmarkup/