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Versions: (draft-perkins-rtcweb-rtp-usage) 00 01 02 03 04 05 06 07 08 09 10 11 12 15 16 17

Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Standards Track                           M. Westerlund
Expires: April 25, 2013                                         Ericsson
                                                                  J. Ott
                                                        Aalto University
                                                        October 22, 2012


  Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
                     draft-ietf-rtcweb-rtp-usage-05

Abstract

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

Status of this Memo

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

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

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

   This Internet-Draft will expire on April 25, 2013.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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



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


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Rationale  . . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  5
   4.  WebRTC Use of RTP: Core Protocols  . . . . . . . . . . . . . .  6
     4.1.  RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . .  6
     4.2.  Choice of the RTP Profile  . . . . . . . . . . . . . . . .  7
     4.3.  Choice of RTP Payload Formats  . . . . . . . . . . . . . .  8
     4.4.  RTP Session Multiplexing . . . . . . . . . . . . . . . . .  8
     4.5.  RTP and RTCP Multiplexing  . . . . . . . . . . . . . . . .  9
     4.6.  Reduced Size RTCP  . . . . . . . . . . . . . . . . . . . . 10
     4.7.  Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 10
     4.8.  Choice of RTP Synchronisation Source (SSRC)  . . . . . . . 10
     4.9.  Generation of the RTCP Canonical Name (CNAME)  . . . . . . 11
   5.  WebRTC Use of RTP: Extensions  . . . . . . . . . . . . . . . . 11
     5.1.  Conferencing Extensions  . . . . . . . . . . . . . . . . . 11
       5.1.1.  Full Intra Request (FIR) . . . . . . . . . . . . . . . 12
       5.1.2.  Picture Loss Indication (PLI)  . . . . . . . . . . . . 12
       5.1.3.  Slice Loss Indication (SLI)  . . . . . . . . . . . . . 13
       5.1.4.  Reference Picture Selection Indication (RPSI)  . . . . 13
       5.1.5.  Temporal-Spatial Trade-off Request (TSTR)  . . . . . . 13
       5.1.6.  Temporary Maximum Media Stream Bit Rate Request
               (TMMBR)  . . . . . . . . . . . . . . . . . . . . . . . 13
     5.2.  Header Extensions  . . . . . . . . . . . . . . . . . . . . 14
       5.2.1.  Rapid Synchronisation  . . . . . . . . . . . . . . . . 14
       5.2.2.  Client-to-Mixer Audio Level  . . . . . . . . . . . . . 14
       5.2.3.  Mixer-to-Client Audio Level  . . . . . . . . . . . . . 15
   6.  WebRTC Use of RTP: Improving Transport Robustness  . . . . . . 15
     6.1.  Negative Acknowledgements and RTP Retransmission . . . . . 15
     6.2.  Forward Error Correction (FEC) . . . . . . . . . . . . . . 16
   7.  WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 16
     7.1.  Boundary Conditions and Circuit Breakers . . . . . . . . . 17
     7.2.  RTCP Limitations for Congestion Control  . . . . . . . . . 18
     7.3.  Congestion Control Interoperability With Legacy Systems  . 19
   8.  WebRTC Use of RTP: Performance Monitoring  . . . . . . . . . . 19
   9.  WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 20
   10. Signalling Considerations  . . . . . . . . . . . . . . . . . . 20
   11. WebRTC API Considerations  . . . . . . . . . . . . . . . . . . 21
     11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 21
   12. RTP Implementation Considerations  . . . . . . . . . . . . . . 22



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     12.1. RTP Sessions and PeerConnection  . . . . . . . . . . . . . 22
     12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 24
     12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 24
     12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 25
     12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 26
     12.6. Media Synchronization  . . . . . . . . . . . . . . . . . . 27
     12.7. Multiple RTP End-points  . . . . . . . . . . . . . . . . . 27
     12.8. Simulcast  . . . . . . . . . . . . . . . . . . . . . . . . 28
     12.9. Differentiated Treatment of Flows  . . . . . . . . . . . . 29
   13. Open Issues  . . . . . . . . . . . . . . . . . . . . . . . . . 30
   14. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 31
   15. Security Considerations  . . . . . . . . . . . . . . . . . . . 31
   16. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 32
   17. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32
     17.1. Normative References . . . . . . . . . . . . . . . . . . . 32
     17.2. Informative References . . . . . . . . . . . . . . . . . . 35
   Appendix A.  Supported RTP Topologies  . . . . . . . . . . . . . . 36
     A.1.  Point to Point . . . . . . . . . . . . . . . . . . . . . . 37
     A.2.  Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 40
     A.3.  Mixer Based  . . . . . . . . . . . . . . . . . . . . . . . 43
       A.3.1.  Media Mixing . . . . . . . . . . . . . . . . . . . . . 43
       A.3.2.  Media Switching  . . . . . . . . . . . . . . . . . . . 46
       A.3.3.  Media Projecting . . . . . . . . . . . . . . . . . . . 49
     A.4.  Translator Based . . . . . . . . . . . . . . . . . . . . . 52
       A.4.1.  Transcoder . . . . . . . . . . . . . . . . . . . . . . 52
       A.4.2.  Gateway / Protocol Translator  . . . . . . . . . . . . 53
       A.4.3.  Relay  . . . . . . . . . . . . . . . . . . . . . . . . 55
     A.5.  End-point Forwarding . . . . . . . . . . . . . . . . . . . 59
     A.6.  Simulcast  . . . . . . . . . . . . . . . . . . . . . . . . 60
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61





















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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.

   The structure of this memo is as follows.  Section 2 outlines our
   rationale in preparing this memo and choosing these RTP features.
   Section 3 defines requirement terminology.  Requirements for core RTP
   protocols are described in Section 4 and suggested RTP extensions are
   described in Section 5.  Section 6 outlines mechanisms that can
   increase robustness to network problems, while Section 7 describes
   congestion control and rate adaptation mechanisms.  The discussion of
   mandated RTP mechanisms concludes in Section 8 with a review of
   performance monitoring and network management tools that can be used
   in the WebRTC context.  Section 9 gives some guidelines for future
   incorporation of other RTP and RTP Control Protocol (RTCP) extensions
   into this framework.  Section 10 describes requirements placed on the
   signalling channel.  Section 11 discusses the relationship between
   features of the RTP framework and the WebRTC application programming
   interface (API), and Section 12 discusses RTP implementation
   considerations.  This memo concludes with an appendix discussing
   several different RTP Topologies, and how they affect the RTP
   session(s) and various implementation details of possible realization
   of central nodes.


2.  Rationale

   The RTP framework comprises the RTP data transfer protocol, the RTP
   control protocol, and numerous RTP payload formats, profiles, and
   extensions.  This range of add-ons has allowed RTP to meet various
   needs that were not envisaged by the original protocol designers, and
   to support many new media encodings, but raises the question of what



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   extensions are to be supported by new implementations.  The
   development of the WebRTC framework provides an opportunity for us to
   review the available RTP features and extensions, and to define a
   common baseline feature set for all WebRTC implementations of RTP.
   This builds on the past 15 years development of RTP to mandate the
   use of extensions that have shown widespread utility, while still
   remaining compatible with the wide installed base of RTP
   implementations where possible.

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

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


3.  Terminology

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

   We define the following 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



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      endpoints and any interconnecting middle nodes.

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

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


4.  WebRTC Use of RTP: Core Protocols

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

4.1.  RTP and RTCP

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

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

   o  Support for use of multiple simultaneous SSRC values in a single
      RTP session, including support for RTP end-points that send many
      SSRC values simultaneously.

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

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

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

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



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

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

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

   Other implementation considerations are discussed in Section 12.

4.2.  Choice of the RTP Profile

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



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   form of source authentication.  WebRTC implementations MUST NOT send
   packets using the basic RTP/AVP profile or the RTP/AVPF profile; they
   MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP
   packets that are generated.  The default and mandatory to implement
   transforms listed in Section 5 of [RFC3711] SHALL apply.

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

4.3.  Choice of RTP Payload Formats

   Implementations MUST follow the WebRTC Audio Codec and Processing
   Requirements [I-D.ietf-rtcweb-audio] and SHOULD follow the updated
   recommendations for audio codecs in the RTP/AVP Profile
   [I-D.terriberry-avp-codecs].  Support for other audio codecs is
   OPTIONAL.

   (tbd: the mandatory to implement video codec is not yet decided)

   Endpoints MAY signal support for multiple RTP payload formats, or
   multiple configurations of a single RTP payload format, provided each
   payload format uses a different RTP payload type number.  An endpoint
   that has signalled support for multiple RTP payload formats SHOULD
   accept data in any of those payload formats at any time, unless it
   has previously signalled limitations on its decoding capability.
   This requirement is constrained if several media types are sent in
   the same RTP session.  In such a case, a source (SSRC) is restricted
   to switching only between the RTP payload formats signalled for the
   media type that is being sent by that source; see Section 4.4.  To
   support rapid rate adaptation by changing codec, RTP does not require
   advance signalling for changes between RTP payload formats that were
   signalled during session set-up.

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

4.4.  RTP Session Multiplexing

   An association amongst a set of participants communicating with RTP
   is known as an RTP session.  A participant can be involved in
   multiple RTP sessions at the same time.  In a multimedia session,
   each medium has typically been carried in a separate RTP session with



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   its own RTCP packets (i.e., one RTP session for the audio, with a
   separate RTP session using a different transport address for the
   video; if SDP is used, this corresponds to one RTP session for each
   "m=" line in the SDP).  WebRTC implementations of RTP are REQUIRED to
   implement support for multimedia sessions in this way, for
   compatibility with legacy systems.

   In today's networks, however, with the widespread use of Network
   Address/Port Translators (NAT/NAPT) and Firewalls (FW), it is
   desirable to reduce the number of transport addresses used by real-
   time media applications using RTP by combining multimedia traffic in
   a single RTP session.  (Details of how this is to be done are tbd,
   but see [I-D.lennox-rtcweb-rtp-media-type-mux],
   [I-D.holmberg-mmusic-sdp-bundle-negotiation] and
   [I-D.westerlund-avtcore-multiplex-architecture].)  Using a single RTP
   session also effects the possibility for differentiated treatment of
   media flows.  This is further discussed in Section 12.9.

   WebRTC implementations of RTP are REQUIRED to support multiplexing of
   a multimedia session onto a single RTP session according to (tbd).
   If such RTP session multiplexing is to be used, this MUST be
   negotiated during the signalling phase.  Support for multiple RTP
   sessions over a single UDP flow as defined by
   [I-D.westerlund-avtcore-transport-multiplexing] is RECOMMENDED/
   OPTIONAL.

   (tbd: No consensus on the level of including support of Multiple RTP
   sessions over a single UDP flow.)

4.5.  RTP and RTCP Multiplexing

   Historically, RTP and RTCP have been run on separate transport layer
   addresses (e.g., two UDP ports for each RTP session, one port for RTP
   and one port for RTCP).  With the increased use of Network Address/
   Port Translation (NAPT) this has become problematic, since
   maintaining multiple NAT bindings can be costly.  It also complicates
   firewall administration, since multiple ports need to be opened to
   allow RTP traffic.  To reduce these costs and session set-up times,
   support for multiplexing RTP data packets and RTCP control packets on
   a single port for each RTP session is REQUIRED, as specified in
   [RFC5761].  For backwards compatibility, implementations are also
   REQUIRED to support sending of RTP and RTCP to separate destination
   ports.

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



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   keep-alives of RTP sessions [RFC6263].

4.6.  Reduced Size RTCP

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

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

4.7.  Symmetric RTP/RTCP

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

4.8.  Choice of RTP Synchronisation Source (SSRC)

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

   Use of the "a=ssrc:" attribute is OPTIONAL.  Implementations MUST
   support random SSRC assignment, and MUST support SSRC collision
   detection and resolution, both according to [RFC3550].




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4.9.  Generation of the RTCP Canonical Name (CNAME)

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

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

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


5.  WebRTC Use of RTP: Extensions

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

5.1.  Conferencing Extensions

   RTP is inherently a group communication protocol.  Groups can be
   implemented using a centralised server, multi-unicast, or using IP
   multicast.  While IP multicast was popular in early deployments, in
   today's practice, overlay-based conferencing dominates, typically
   using one or more central servers to connect endpoints in a star or
   flat tree topology.  These central servers can be implemented in a
   number of ways as discussed in Appendix A, and in the memo on RTP
   Topologies [I-D.westerlund-avtcore-rtp-topologies-update].

   As discussed in Section 3.7 of
   [I-D.westerlund-avtcore-rtp-topologies-update], the use of a video
   switching MCU makes the use of RTCP for congestion control, or any
   type of quality reports, very problematic.  Also, as discussed in



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   section 3.8 of [I-D.westerlund-avtcore-rtp-topologies-update], the
   use of a content modifying MCU with RTCP termination breaks RTP loop
   detection and removes the ability for receivers to identify active
   senders.  RTP Transport Translators (Topo-Translator) are not of
   immediate interest to WebRTC, although the main difference compared
   to point to point is the possibility of seeing multiple different
   transport paths in any RTCP feedback.  Accordingly, only Point to
   Point (Topo-Point-to-Point), Multiple concurrent Point to Point
   (Mesh) and RTP Mixers (Topo-Mixer) topologies are needed to achieve
   the use-cases to be supported in WebRTC initially.  These RECOMMENDED
   topologies are expected to be supported by all WebRTC end-points
   (these topologies require no special RTP-layer support in the end-
   point if the RTP features mandated in this memo are implemented).

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

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

5.1.1.  Full Intra Request (FIR)

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

5.1.2.  Picture Loss Indication (PLI)

   The Picture Loss Indication is defined in Section 6.3.1 of the RTP/
   AVPF profile [RFC4585].  It is used by a receiver to tell the sending
   encoder that it lost the decoder context and would like to have it
   repaired somehow.  This is semantically different from the Full Intra



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   Request above as there there could be multiple ways to fulfil the
   request.  It is REQUIRED that WebRTC senders understand and react to
   this feedback message as a loss tolerance mechanism; receivers MAY
   send PLI messages.

5.1.3.  Slice Loss Indication (SLI)

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

5.1.4.  Reference Picture Selection Indication (RPSI)

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

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

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

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

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




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5.2.  Header Extensions

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

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

5.2.1.  Rapid Synchronisation

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

5.2.2.  Client-to-Mixer Audio Level

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

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



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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.


6.  WebRTC Use of RTP: Improving Transport Robustness

   There are some tools that can make RTP flows robust against Packet
   loss and reduce the impact on media quality.  However, they all add
   extra bits compared to a non-robust stream.  These extra bits need to
   be considered, and the aggregate bit-rate MUST be rate-controlled.
   Thus, improving robustness might require a lower base encoding
   quality, but has the potential to deliver that quality with fewer
   errors.  The mechanisms described in the following sub-sections can
   be used to improve tolerance to packet loss.

6.1.  Negative Acknowledgements and RTP Retransmission

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

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

   The RTP Retransmission Payload Format [RFC4588] offers the ability to
   retransmit lost packets based on NACK feedback.  Retransmission needs
   to be used with care in interactive real-time applications to ensure



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   that the retransmitted packet arrives in time to be useful, but can
   be effective in environments with relatively low network RTT (an RTP
   sender can estimate the RTT to the receivers using the information in
   RTCP SR and RR packets).  The use of retransmissions can also
   increase the forward RTP bandwidth, and can potentially worsen the
   problem if the packet loss was caused by network congestion.  We
   note, however, that retransmission of an important lost packet to
   repair decoder state can have lower cost than sending a full intra
   frame.  It is not appropriate to blindly retransmit RTP packets in
   response to a NACK.  The importance of lost packets and the
   likelihood of them arriving in time to be useful needs to be
   considered before RTP retransmission is used.

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

6.2.  Forward Error Correction (FEC)

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

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

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


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

   WebRTC will be used in heterogeneous network environments using a



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

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

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

7.1.  Boundary Conditions and Circuit Breakers

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

   The session establishment signalling will also necessarily establish
   boundaries to which the media bit-rate will conform.  The choice of
   media codecs provides upper- and lower-bounds on the supported bit-
   rates that the application can utilise to provide useful quality, and



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   the packetization choices that exist.  In addition, the signalling
   channel can establish maximum media bit-rate boundaries using the SDP
   "b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media
   Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo).
   The combination of media codec choice and signalled bandwidth limits
   SHOULD be used to limit traffic based on known bandwidth limitations,
   for example the capacity of the edge links, to the extent possible.

7.2.  RTCP Limitations for Congestion Control

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

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

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

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

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







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7.3.  Congestion Control Interoperability With Legacy Systems

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

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

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


8.  WebRTC Use of RTP: Performance Monitoring

   RTCP does contains a basic set of RTP flow monitoring metrics like
   packet loss and jitter.  There are a number of extensions that could
   be included in the set to be supported.  However, in most cases which
   RTP monitoring that is needed depends on the application, which makes
   it difficult to select which to include when the set of applications
   is very large.

   Exposing some metrics in the WebRTC API needs to be considered
   allowing the application to gather the measurements of interest.
   However, security implications for the different data sets exposed
   will need to be considered in this.

   (tbd: If any RTCP XR metrics need to be added is still an open
   question, but possible to extend at a later stage)






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9.  WebRTC Use of RTP: Future Extensions

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

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


10.  Signalling Considerations

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

   RTP Profile:  The name of the RTP profile to be used in session.  The
      RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
      on basic level, as can their secure variants RTP/SAVP [RFC3711]
      and RTP/SAVPF [RFC5124].  The secure variants of the profiles do
      not directly interoperate with the non-secure variants, due to the
      presence of additional header fields in addition to any
      cryptographic transformation of the packet content.  As WebRTC
      requires the usage of the RTP/SAVPF profile this can be inferred
      as there is only a single profile, but in SDP this is still
      information that has to be signalled.  Interworking functions
      might transform this into RTP/SAVP for a legacy use case by
      indicating to the WebRTC end-point a RTP/SAVPF end-point and
      limiting the usage of the a=rtcp attribute to indicate a trr-int
      value of 4 seconds.

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



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      signalled (see Section 4.5).  If several RTP sessions are to be
      multiplexed onto a single transport layer flow, this MUST also be
      signalled (see Section 4.4).

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

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

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

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


11.  WebRTC API Considerations

   The following sections describe how the WebRTC API features map onto
   the RTP mechanisms described in this memo.

11.1.  API MediaStream to RTP Mapping

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



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   MediaStream are expected to be synchronized.

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

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

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

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


12.  RTP Implementation Considerations

   The following provide some guidance on the implementation of the RTP
   features described in this memo.

   This section discusses RTP functionality that is part of the RTP
   standard, needed by decisions made, or to enable use cases raised and
   their motivations.  This discussion is from an WebRTC end-point
   perspective.  It will occasionally talk about central nodes, but as
   this specification is for an end-point, this is where the focus lies.
   For more discussion on the central nodes and details about RTP
   topologies please see Appendix A.

   The section will touch on the relation with certain RTP/RTCP
   extensions, but will focus on the RTP core functionality.  The
   definition of what functionalities and the level of requirement on
   implementing it is defined in Section 2.

12.1.  RTP Sessions and PeerConnection

   An RTP session is an association among RTP nodes, which have one
   common SSRC space.  An RTP session can include any number of end-
   points and nodes sourcing, sinking, manipulating or reporting on the



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   RTP media streams being sent within the RTP session.  A
   PeerConnection being a point-to-point association between an end-
   point and another node.  That peer node can be both an end-point or
   centralized processing node of some type; thus, the RTP session can
   terminate immediately on the far end of the PeerConnection, but it
   might also continue as further discussed below in Multiparty
   (Section 12.3) and Multiple RTP End-points (Section 12.7).

   A PeerConnection can contain one or more RTP session depending on how
   it is setup and how many UDP flows it uses.  A common usage has been
   to have one RTP session per media type, e.g. one for audio and one
   for video, each sent over different UDP flows.  However, the default
   usage in WebRTC will be to use one RTP session for all media types.
   This usage then uses only one UDP flow, as also RTP and RTCP
   multiplexing is mandated (Section 4.5).  However, for legacy
   interworking and network prioritization (Section 12.9) based on
   flows, a WebRTC end-point needs to support a mode of operation where
   one RTP session per media type is used.  Currently, each RTP session
   has to use its own UDP flow.  Discussions are ongoing if a solution
   enabling multiple RTP sessions over a single UDP flow, see
   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
                                   +---+
                                   | C |
                                   +---+

                          Figure 1: Multi-unicast

   The multi-unicast topology could be implemented as a single RTP



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

12.2.  Multiple Sources

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

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

12.3.  Multiparty

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

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



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   suitable for multiple destinations requirements.  These requirements
   can both be dependent on transport path and the different end-points
   preferences related to play out of the media.

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

                Figure 2: RTP Mixer with Only Unicast Paths

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

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

   Mixers also need to be trusted nodes when it comes to security as it
   manipulates either RTP or the media itself before sending it on
   towards the end-point(s), thus they need to be able to decrypt and
   then encrypt it before sending it out.

12.4.  SSRC Collision Detection

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

   In a point-to-point session where each SSRC is associated with either
   of the two end-points and where the main media carrying SSRC
   identifier will be announced in the signalling channel, a collision



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   is less likely to occur due to the information about used SSRCs
   provided by Source-Specific SDP Attributes [RFC5576].  Still if both
   end-points start uses an new SSRC identifier prior to having
   signalled it to the peer and received acknowledgement on the
   signalling message, there can be collisions.  The Source-Specific SDP
   Attributes [RFC5576] contains no mechanism to resolve SSRC collisions
   or reject a end-points usage of an SSRC.

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

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

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

12.5.  Contributing Sources

   Contributing Sources (CSRC) is a functionality in the RTP header that
   allows an RTP node to combine media packets from multiple sources
   into one and to identify which sources yielded the result.  For
   WebRTC end-points, supporting contributing sources is trivial.  The
   set of CSRCs is provided in a given RTP packet.  This information can
   then be exposed to the applications using some form of API, possibly
   a mapping back into WebRTC MediaStream identities to avoid having to
   expose two name spaces and the handling of SSRC collision handling to
   the JavaScript.

   (tbd: does the API need to provide the ability to add a CSRC list to
   an outgoing packet? this is only useful if the sender is mixing
   content)

   There are also at least one extension that depends on the CSRC list
   being used: the Mixer-to-client audio level [RFC6465], which enhances
   the information provided by the CSRC to actual energy levels for



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   audio for each contributing source.

12.6.  Media Synchronization

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

   The next provision is that the internal clocks of all media sources,
   i.e., what drives the RTP timestamp, can be correlated to a system
   clock that is provided in RTCP Sender Reports encoded in an NTP
   format.  By correlating all RTP timestamps to a common system clock
   for all sources, the timing relation of the different RTP 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



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   receive an RTP flow and send reports and feedback about it has
   several impacts.  As previously discussed (Section 12.3) any codec
   control and rate control needs to be capable of merging the
   requirements and preferences to provide a single best encoding
   according to the situation RTP media stream.  Specifically, when it
   comes to congestion control it needs to be capable of identifying the
   different end-points to form independent congestion state information
   for each different path.

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

12.8.  Simulcast

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

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

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






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12.9.  Differentiated Treatment of Flows

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

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

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

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

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

   With the exception of DiffServ both flow based and DPI have issues
   with running multiple media types and flows on a single UDP flow,
   especially when combined with data transport (SCTP/DTLS).  DPI has
   issues because multiple types of flows are aggregated and thus it
   becomes more difficult to analyse them.  The flow-based
   differentiation will provide the same treatment to all packets within
   the flow, i.e., relative prioritization is not possible.  Moreover,
   if the resources are limited it might not be possible to provide
   differential treatment compared to best-effort for all the flows in a
   WebRTC application.

   When flow-based differentiation is available the WebRTC application
   needs to know about it so that it can provide the separation of the
   RTP media streams onto different UDP flows to enable a more granular
   usage of flow based differentiation.

   DiffServ assumes that either the end-point or a classifier can mark
   the packets with an appropriate DSCP so that the packets are treated
   according to that marking.  If the end-point is to mark the traffic
   two requirements arise in the WebRTC context: 1) The WebRTC
   application or browser has to know which DSCP to use and that it can
   use them on some set of RTP media streams. 2) The information needs



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   to be propagated to the operating system when transmitting the
   packet.  These issues are discussed in DSCP and other packet markings
   for RTCWeb QoS [I-D.ietf-rtcweb-qos].

   tbd: The model for providing differentiated treatment needs to be
   evolved.  Most of this is not the responsibility of this memo.
   However, this memo could include:

   1.  How can the application can prioritize MediaStreamTracks
       differently in the API?

   2.  How MediaStreamTrack prioritization maps to the RTP level, and
       what type of marking behaviour can occur on the RTP media stream
       and its datagram?


13.  Open Issues

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

   1.   Need to add references to the RTP payload format for the Video
        Codec chosen in Section 4.3.

   2.   The methods and solutions for RTP multiplexing over a single
        transport is not yet finalized in Section 4.4.

   3.   RTP congestion control algorithms will probably require some
        feedback information to be conveyed in RTCP.  Are the tools that
        are mandated by this memo sufficient, or do we need additional
        information?

   4.   RTP congestion control could be implementing using either a
        sender-based algorithm or a receiver-based algorithm.  To ensure
        interoperability, does this memo need to mandate which end is in
        charge of congestion control for a path?

   5.   Still open if any RTCP XR performance metrics are needed, as
        discussed in Section 8.

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

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





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   8.   Is an API needed for expressing any application level media
        mixing of an RTP media stream so that the correct CSRC list can
        be set as discussed in Section 12.5?

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

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

   11.  Editing of Appendix A to remove redundancy between this and the
        update of RTP Topologies
        [I-D.westerlund-avtcore-rtp-topologies-update].


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.  Security Considerations

   The security considerations for the WebRTC framework are described in
   [I-D.ietf-rtcweb-security].  The overall security architecture for
   WebRTC is described in [I-D.ietf-rtcweb-security-arch].

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

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

   tbd: Privacy concerns, and the generation of untraceable CNAMEs, are
   under discussion.

   The guidelines in [RFC6562] apply when using variable bit rate (VBR)
   audio codecs, e.g., Opus or the Mixer audio level header extensions.



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16.  Acknowledgements

   The authors would like to thank Harald Alvestrand, Cary Bran, Charles
   Eckel and Cullen Jennings for valuable feedback.


17.  References

17.1.  Normative References

   [I-D.holmberg-mmusic-sdp-bundle-negotiation]
              Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
              Using Session Description Protocol (SDP) Port Numbers",
              draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
              progress), October 2011.

   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
              Breakers for Unicast Sessions",
              draft-ietf-avtcore-rtp-circuit-breakers-00 (work in
              progress), October 2012.

   [I-D.ietf-avtcore-srtp-encrypted-header-ext]
              Lennox, J., "Encryption of Header Extensions in the Secure
              Real-Time Transport Protocol (SRTP)",
              draft-ietf-avtcore-srtp-encrypted-header-ext-02 (work in
              progress), July 2012.

   [I-D.ietf-avtext-multiple-clock-rates]
              Petit-Huguenin, M. and G. Zorn, "Support for Multiple
              Clock Rates in an RTP Session",
              draft-ietf-avtext-multiple-clock-rates-06 (work in
              progress), October 2012.

   [I-D.ietf-rtcweb-audio]
              Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", draft-ietf-rtcweb-audio-00 (work in
              progress), September 2012.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-04 (work
              in progress), June 2012.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for RTC-Web",
              draft-ietf-rtcweb-security-03 (work in progress),
              June 2012.



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   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "RTCWEB Security Architecture",
              draft-ietf-rtcweb-security-arch-05 (work in progress),
              October 2012.

   [I-D.lennox-rtcweb-rtp-media-type-mux]
              Rosenberg, J. and J. Lennox, "Multiplexing Multiple Media
              Types In a Single Real-Time Transport Protocol (RTP)
              Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work
              in progress), October 2011.

   [I-D.rescorla-avtcore-6222bis]
              Rescorla, E. and A. Begen, "Guidelines for Choosing RTP
              Control Protocol (RTCP) Canonical Names (CNAMEs)",
              draft-rescorla-avtcore-6222bis-00 (work in progress),
              October 2012.

   [I-D.terriberry-avp-codecs]
              Terriberry, T., "Update to Recommended Codecs for the AVP
              RTP Profile", draft-terriberry-avp-codecs-00 (work in
              progress), August 2012.

   [I-D.westerlund-avtcore-transport-multiplexing]
              Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
              Single Lower-Layer Transport",
              draft-westerlund-avtcore-transport-multiplexing-04 (work
              in progress), October 2012.

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

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

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

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

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

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.



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              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 2006.

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

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

   [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.




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

17.2.  Informative References

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

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

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

   [I-D.ietf-rtcweb-use-cases-and-requirements]
              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements",
              draft-ietf-rtcweb-use-cases-and-requirements-09 (work in
              progress), June 2012.

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

   [I-D.westerlund-avtcore-multiplex-architecture]
              Westerlund, M., Burman, B., Perkins, C., and H.
              Alvestrand, "Guidelines for using the Multiplexing
              Features of RTP",
              draft-westerlund-avtcore-multiplex-architecture-02 (work
              in progress), July 2012.

   [I-D.westerlund-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies",
              draft-westerlund-avtcore-rtp-topologies-update-01 (work in
              progress), October 2012.

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



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              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 2006.

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

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

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

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

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

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


Appendix A.  Supported RTP Topologies

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

   This section includes RTP topologies beyond the RECOMMENDED ones.



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   This in an attempt to highlight the differences and the in many case
   small differences in implementation to support a larger set of
   possible topologies.

   (tbd: This section needs reworking and clearer relation to
   [I-D.westerlund-avtcore-rtp-topologies-update].)

A.1.  Point to Point

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

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

                         Figure 3: Point to Point

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

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

   In cases where there is only one type of media (e.g., a voice-only
   call) this topology will be implemented as a single RTP session, with
   bidirectional flows of RTP and RTCP packets, all then multiplexed
   onto a single 5-tuple.  If multiple types of media are to be used
   (e.g., audio and video), then each type media can be sent as a
   separate RTP session using a different 5-tuple, allowing for separate
   transport level treatment of each type of media.  Alternatively, all
   types of media can be multiplexed onto a single 5-tuple as a single
   RTP session, or as several RTP sessions if using a demultiplexing
   shim.  Multiplexing different types of media onto a single 5-tuple
   places some limitations on how RTP is used, as described in "RTP
   Multiplexing Architecture"
   [I-D.westerlund-avtcore-multiplex-architecture].  It is not expected



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   that these limitations will significantly affect the scenarios
   targeted by WebRTC, but they can impact interoperability with legacy
   systems.

   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.
























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

              Figure 4: Point to Point: Multiple RTP sessions

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



















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            +-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.

                              +---+      +---+
                              | 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



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



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



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   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.

   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



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   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.





















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      +-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



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   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



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   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.
























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      +-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  | |
      | | | | |       |<------------|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



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   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
   here.

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.



























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



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                     Figure 12: Media Projecting Mixer

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

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

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

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





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A.4.  Translator Based

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

A.4.1.  Transcoder

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

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










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

                        Figure 13: Media Transcoder

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

A.4.2.  Gateway / Protocol Translator

   Gateways are used when some protocol feature that are needed are not
   supported by an end-point wants to participate in session.  This RTP
   translator in Figure 14 takes on the role of ensuring that from the
   perspective of participant A, participant B appears as a fully
   compliant WebRTC end-point (that is, it is the combination of the
   Translator and participant B that looks like a WebRTC end point).



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                               +------------+
                               |            |
                    +---+      | 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 <----+----------------+ ||
       |   | | +-------|             |---------+                  ||
       |   | +---------|             |----------------------------+|
       |   +-----------|             |-----------+                 |
       |               |             |                             |
       +---------------+             +-----------------------------+


                  Figure 15: RTP/RTCP Protocol Translator



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

         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



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   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.






























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      +-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 <----------------------+  | |
      | | | | |       |<------------|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



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   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
   congestion.

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

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

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













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

             Figure 18: DTLS-SRTP host and RTP Relay Separated

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

A.5.  End-point Forwarding

   An WebRTC end-point (B in Figure 19) will receive a WebRTC
   MediaStream (set of SSRCs) over a PeerConnection (from A).  For the
   moment is not decided if the end-point is allowed or not to in its
   turn send that WebRTC MediaStream over another PeerConnection to C.
   This section discusses the RTP and end-point implications of allowing
   such functionality, which on the API level is extremely simplistic to
   perform.

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

                     Figure 19: MediaStream Forwarding

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

   A relay approach will result in that the WebRTC end-points will have
   to have the same capabilities as being discussed in Relay
   (Appendix A.4.3).  Thus A will see an RTP session that is extended



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

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

A.6.  Simulcast

   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
   PeerConnection.

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




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   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
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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


   Joerg Ott
   Aalto University
   School of Electrical Engineering
   Espoo  02150
   Finland

   Email: jorg.ott@aalto.fi
















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