Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Standards Track                           M. Westerlund
Expires: December 6, 2012 January 17, 2013                                       Ericsson
                                                                  J. Ott
                                                        Aalto University
                                                            June 4,
                                                           July 16, 2012

  Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
                     draft-ietf-rtcweb-rtp-usage-03
                     draft-ietf-rtcweb-rtp-usage-04

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 December 6, 2012. January 17, 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
   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  . . . . . . . . . . . . . .  7  8
     4.4.  RTP Session Multiplexing . . . . . . . . . . . . . . . . .  8  9
     4.5.  RTP and RTCP Multiplexing  . . . . . . . . . . . . . . . .  8 10
     4.6.  Reduced Size RTCP  . . . . . . . . . . . . . . . . . . . .  9 10
     4.7.  Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . .  9 11
     4.8.  Choice of RTP Synchronisation Source (SSRC)  . . . . . . . 11
     4.9.  Generation of the RTCP Canonical Name (CNAME)  . . . . . . 10 11
   5.  WebRTC Use of RTP: Extensions  . . . . . . . . . . . . . . . . 10 12
     5.1.  Conferencing Extensions  . . . . . . . . . . . . . . . . . 10 12
       5.1.1.  Full Intra Request (FIR) . . . . . . . . . . . . . . . . . . 11 13
       5.1.2.  Picture Loss Indication (PLI)  . . . . . . . . . . . . . . . 11 13
       5.1.3.  Slice Loss Indication (SLI)  . . . . . . . . . . . . . . . . 11 13
       5.1.4.  Reference Picture Selection Indication (RPSI)  . . . . 14
       5.1.5.  Temporal-Spatial Trade-off Request (TSTR)  . . . . . . 12
       5.1.5. 14
       5.1.6.  Temporary Maximum Media Stream Bit Rate Request  . . . 12 14
     5.2.  Header Extensions  . . . . . . . . . . . . . . . . . . . . 12 14
       5.2.1.  Rapid Synchronisation  . . . . . . . . . . . . . . . . 12 15
       5.2.2.  Client to Mixer  Client-to-Mixer Audio Level  . . . . . . . . . . . . . 13 15
       5.2.3.  Mixer to Client  Mixer-to-Client Audio Level  . . . . . . . . . . . . . 13 15
   6.  WebRTC Use of RTP: Improving Transport Robustness  . . . . . . 13 16
     6.1.  Negative Acknowledgements and RTP Retransmission . . . . . . . . . . . . . . . . . . . . . . 14 16
     6.2.  Forward Error Correction (FEC) . . . . . . . . . . . . . . 15
       6.2.1.  Basic Redundancy . . . . . . . . . . . . . . . . . . . 15
       6.2.2.  Block Based FEC  . . . . . . . . . . . . . . . . . . . 16
       6.2.3.  Recommendations for FEC  . . . . . . . . . . . . . . . 17
   7.  WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 17
     7.1.  Congestion Control Requirements  . . . . . . . . . . . . . 19 18
     7.2.  Rate Control Boundary Conditions . . . . . . . . . . . . . 19
     7.3.  RTCP Limiations  . . . . . . . . . . . . Limitations for Congestion Control  . . . . . . . . . 19
     7.4.  Congestion Control Interoperability With Legacy Interop Limitations . . . . . . . . . . . . . . . Systems  . 20
   8.  WebRTC Use of RTP: Performance Monitoring  . . . . . . . . . . 21 20
   9.  WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 21
   10. Signalling Considerations  . . . . . . . . . . . . . . . . . . 21
   11. WebRTC API Considerations  . . . . . . . . . . . . . . . . . . 23 22
     11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 23 22
   12. RTP Implementation Considerations  . . . . . . . . . . . . . . 23
     12.1. RTP Sessions and PeerConnection  . . . . . . . . . . . . . 24 23
     12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 25
     12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 25
     12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 27 26
     12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 28 27
     12.6. Media Synchronization  . . . . . . . . . . . . . . . . . . 29 28
     12.7. Multiple RTP End-points  . . . . . . . . . . . . . . . . . 29 28
     12.8. Simulcast  . . . . . . . . . . . . . . . . . . . . . . . . 30 29
     12.9. Differentiated Treatment of Flows  . . . . . . . . . . . . 30 29
   13. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 31
   14. Security Considerations  . . . . . . . . . . . . . . . . . . . 32 31
   15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 32 31
   16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32
     16.1. Normative References . . . . . . . . . . . . . . . . . . . 32
     16.2. Informative References . . . . . . . . . . . . . . . . . . 35 34
   Appendix A.  Supported RTP Topologies  . . . . . . . . . . . . . . 37 36
     A.1.  Point to Point . . . . . . . . . . . . . . . . . . . . . . 37 36
     A.2.  Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 40 39
     A.3.  Mixer Based  . . . . . . . . . . . . . . . . . . . . . . . 43 42
       A.3.1.  Media Mixing . . . . . . . . . . . . . . . . . . . . . 43 42
       A.3.2.  Media Switching  . . . . . . . . . . . . . . . . . . . 46 45
       A.3.3.  Media Projecting . . . . . . . . . . . . . . . . . . . 49 48
     A.4.  Translator Based . . . . . . . . . . . . . . . . . . . . . 52 51
       A.4.1.  Transcoder . . . . . . . . . . . . . . . . . . . . . . 52 51
       A.4.2.  Gateway / Protocol Translator  . . . . . . . . . . . . 53 52
       A.4.3.  Relay  . . . . . . . . . . . . . . . . . . . . . . . . 55 54
     A.5.  End-point Forwarding . . . . . . . . . . . . . . . . . . . 59 58
     A.6.  Simulcast  . . . . . . . . . . . . . . . . . . . . . . . . 60 59
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61 60

1.  Introduction

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

   The Web Real-Time communication (WebRTC) framework is a new protocol
   framework that provides the
   protocol building blocks to support for 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 must are to be implemented by all WebRTC-aware browsers, 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 recommended RTP extensions
   are described in Section 5.  Section 6 outlines mechanisms that can
   increase robustness to network problems, while Section 7 describes
   the required congestion control and rate adaptation mechanisms.  The
   discussion of required 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
   features should
   extensions are to be supported by new implementations? 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.

   While the baseline set of

   RTP and RTCP extensions not discussed in this document can still be
   implemented by a WebRTC end-point, but they are considered optional,
   are not required for interoperability, and do not provide features
   needed to address the WebRTC use cases and requirements
   [I-D.ietf-rtcweb-use-cases-and-requirements].

   While the baseline set of RTP features and extensions defined in this
   memo is targetted 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 apppropriate appropriate for other desktop or mobile video
   conferencing systems, or for room-based high-quality telepresence
   applications.

3.  Terminology

   This memo specifies various requirements levels for implementation or
   use of RTP features and extensions.  When we describe the importance
   of RTP extensions, or the need for implementation support, we use the
   following requirement levels to specify the importance of the feature
   in the WebRTC framework:

   MUST:  This word, or the terms "REQUIRED" or "SHALL", mean that the
      definition is an absolute requirement of the specification.

   SHOULD:  This word, or the adjective "RECOMMENDED", mean that there
      may exist valid reasons in particular circumstances to ignore a
      particular item, but the full implications must be understood and
      carefully weighed before choosing a different course.

   MAY:  This word, or the adjective "OPTIONAL", mean that an item is
      truly optional.  One vendor may choose to include the item because
      a particular marketplace requires it or because the vendor feels
      that it enhances the product while another vendor may omit the
      same item.  An implementation which does not include a particular
      option MUST be prepared to interoperate with another
      implementation which does include the option, though perhaps with
      reduced functionality.  In the same vein an implementation which
      does include a particular option MUST be prepared to interoperate
      with another implementation which does not include the option
      (except, of course, for the feature the option provides.)

   These key words are used in a manner consistent with their definition
   in [RFC2119].  The above interpretation of these key words applies
   only when written in ALL CAPS.  Lower- or mixed-case uses of these
   key words are not to be interpreted as carrying special significance
   in this memo.

   We define the following terms:

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

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

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

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

4.  WebRTC Use of RTP: Core Protocols

   The following sections describe the core features of RTP and RTCP
   that MUST 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 MUST 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. values (but see also Section 4.8).

   o  Support for reception of RTP data packets containing CSRC lists,
      as generated by RTP mixers. 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 standard sending and receiving RTCP packet types, include SR, RR, SDES, and BYE packets.

   o
      packet types, with OPTIONAL support for other RTCP packet types;
      implementations MUST ignore unknown RTCP packet types.

   o  Support for multiple end-points in a single RTP session, and for
      scaling the RTCP transmission interval according to the number of
      participants in the session; support for randomised RTCP
      transmission intervals to avoid synchronisation of RTCP reports. 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 targetted for purely VoIP targeted at VoIP-only systems, do not support all of
   the above features. 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] is REQUIRED to 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 RTP/AVPF part of RTP/SAVPF is required to get RTCP-based feedback extensions 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.  The RTP/
   AVPF profile  These
   extensions also saves save RTCP bandwidth, and will commonly only use the
   full RTCP bandwidth allocation when if there are many events that require
   feedback.  The RTP/AVPF functionality is  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 RTP/SAVP part of the RTP/SAVPF secure RTP profile is for support for Secure
   RTP (SRTP) [RFC3711].  This provides needed to provide SRTP media encryption,
   integrity protection, replay protection and a limited form of source
   authentication.

   WebRTC implementation implementations MUST NOT send packets using the basic RTP/AVP
   profile or the RTP/AVPF profile; they MUST use employ the full RTP/SAVPF profile.  WebRTC
   implementations
   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.

   (tbd: There is ongoing discussion on what additional keying mechanism
   is to be required, what are the mandated cryptographic transforms.
   This section needs to
   Other key management schemes MAY be updated based on the results of that
   discussion.) supported.

4.3.  Choice of RTP Payload Formats

   (tbd: say something about the choice

   The requirement from Section 6 of RTP Payload Format for
   WebRTC.  If there is [RFC3551] that "Audio applications
   operating under this profile SHOULD, at a mandatory minimum, be able to implement set send
   and/or receive payload types 0 (PCMU) and 5 (DVI4)" applies, since
   Section 4.2 of codecs, this
   should reference them.  In any case, it should reference a discussion
   of signalling for memo mandates the choice use of codec, once that discussion reaches
   closure.)
   Endpoints may signal support for multiple media formats, or the RTP/SAVPF profile,
   which inherits this restriction from the RTP/AVP profile.

   (tbd: there is ongoing discussion on whether support for other audio
   and video codecs is to be mandated)

   Endpoints MAY signal support for multiple media formats, or multiple
   configurations of a single format, provided each uses a different RTP
   payload type number.  An endpoint that has signalled it's its support for
   multiple formats is REQUIRED to accept data in any of those formats
   at any time, unless it has previously signalled limitations on it's its
   decoding capability.

   This requirement is modified constrained if several media types are sent in
   the same RTP session, in that case session.  In such a case, a source (SSRC) is restricted
   to switch switching only between any the RTP payload format established formats signalled for the
   media type that is being sent by that source; see Section 4.4.  To
   support rapid rate adaptation, RTP does not require signalling in
   advance for changes between payload formats that were signalled
   during session setup.

   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 may can be involved in
   multiple RTP sessions at the same time.  In a multimedia session,
   each medium has typically been carried in a separate RTP session with
   its own RTCP packets (i.e., one RTP session for the audio, with a
   separate RTP session running on using a different transport connection 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 layer ports 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 treament 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. 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
   ports 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 Address/
   Port Translation (NAPT) this has become problematic, since
   maintaining multiple NAT bindings can be costly.  It also complicates
   firewall administration, since multiple ports must need to be opened to
   allow RTP traffic.  To reduce these costs and session setup times,
   support for multiplexing RTP data packets and RTCP control packets on
   a single port [RFC5761] for each RTP session is REQUIRED.

   (tbd: Are WebRTC REQUIRED, as specified in
   [RFC5761].  For backwards compatibility, implementations required are also
   REQUIRED to support the case where
   the sending of RTP and RTCP are run on to separate UDP ports, for interoperability
   with legacy systems?) 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 may can be useful to keep-
   alive keep NAT bindings,
   bindings alive, and is the recommend method for application level
   keep-alives of RTP sessions [RFC6263].

4.6.  Reduced Size RTCP

   RTCP packets are usually sent as compound RTCP packets, and [RFC3550]
   requires that those compound packets start with an Sender Report (SR)
   or Receiver Report (RR) packet.  When using frequent RTCP feedback
   messages, these general 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 application applications
   quickly aware of changing network conditions conditions, and to allow them to
   adapt their transmission and encoding behaviour.  Support for RFC5506 sending
   RTCP feedback packets as [RFC5506] non-compound packets is
   REQUIRED. REQUIRED
   when signalled.  For backwards compatibility, implementations are
   also REQUIRED to support the use of compound RTCP feedback packets.

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 addition, it saves
   resources in the form of
   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.  Generation  Choice 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 may change

   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].

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, it's its RTCP CNAME is
   meant to stay unchanged, so that RTP endpoints can be uniquely
   identified and associated with their RTP media streams. 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, some may find long-term persistent identifiers can be
   problematic from a privacy viewpoint.  Accordingly, support for
   generating a short-term persistent RTCP CNAMEs following method (b)
   specified in Section 4.2 of "Guidelines for Choosing RTP Control
   Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is REQUIRED,
   since RECOMMENDED.
   Note, however, that this addresses both concerns. does not resolve the privacy concern as
   there is not sufficient randomness to avoid tracking of an end-point.

   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 according to the recommended
   form above.

   (tbd: there seems to be a growing consensus that the working group
   wants randomly-chosen CNAME values; need to reference a draft that
   describes how this is to be done)

5.  WebRTC Use of RTP: Extensions

   There are a number of RTP extensions that are either required 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 strongly recommended suggested for use within WebRTC. 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 [RFC5117].

   As discussed in Section 3.5 of [RFC5117], the use of a video
   switching MCU makes the use of RTCP for congestion control, or any
   type of quality reports, very problematic.  Also, as discussed in
   section 3.6 of [RFC5117], the use of a content modifying MCU with
   RTCP termination breaks RTP loop detection and removes the ability
   for receivers to identify active senders.  Accordingly, only  RTP Transport Translators (relays), RTP Mixers,
   (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 end-point based
   forwarding RTP Mixers (Topo-Mixer)
   topologies are needed to achieve the use-cases to be supported in WebRTC.
   WebRTC initially.  These RECOMMENDED topologies are expected to be
   supported by all WebRTC end-points (these three topologies require no
   special RTP-layer support in the end-point, end-point if the RTP features
   mandated in this memo are implemented).

   The RTP protocol extensions described below to be used with conferencing, described
   below, are not required for correctness; an RTP endpoint that 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 may offer poor performance.  Support for the listed
   extensions will greatly improve the quality of experience, however, in the context of centralised
   conferencing, where one RTP Mixer (Conference Focus) receives experience and, to
   provide a
   participants media streams and distribute them reasonable baseline quality, some these extensions are
   mandatory to the other
   participants.  These messages be supported by WebRTC end-points.

   The RTCP packets assisting in such operation are defined in the
   Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based (RTCP)-
   Based Feedback (RTP/
   AVPF) (RTP/AVPF) [RFC4585] and the "Codec Control Messages
   in the RTP Audio-
   Visual Audio-Visual Profile with Feedback (AVPF)" (CCM) [RFC5104]
   and are fully usable by the Secure variant of this profile (RTP/SAVPF) (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 have 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 predicted predictive media encoding
   with long prediction chains.  It is REQUIRED that this feedback
   message is supported by RTP senders in WebRTC, since it greatly
   improves the user experience when using centralised mixers-based
   conferencing.

5.1.2.  Picture Loss Indication (PLI)

   The Picture Loss Indication is defined in Section 6.3.1 of the RTP/
   AVPF profile [RFC4585].  It is used by a receiver to tell the sending
   encoder that it lost the decoder context and would like to have it
   repaired somehow.  This is semantically different from the Full Intra
   Request above as there can exist there may be multiple methods to fulfil fulfill the
   request.  It is RECOMMENDED REQUIRED that senders understand and react to this
   feedback message is supported as a loss tolerance mechanism. 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
   macroblocks, 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 synchronicity, 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 use temporal-spatial trade-off request and notification are defined
   in Sections 3.5.2 and 4.3.2 of this RTCP
   feedback message [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 as a loss tolerance mechanism.

5.1.5. OPTIONAL.

5.1.6.  Temporary Maximum Media Stream Bit Rate Request

   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 is are used by a media receiver, receiver to inform the sending party that
   there is a current limitation on the amount of bandwidth available to
   this receiver.  This can be for may have various reasons, and can reasons; for example
   be used by example, an RTP
   mixer may 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.  A RTP media
   stream sender receiving a TMMBR for its SSRC MUST follow the
   limitations set by the message; the sending of TMMBR requests is
   OPTIONAL.

5.2.  Header Extensions

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

   As noted in [RFC5285], the requirement from the RTP specification
   that header extensions are "designed so that the header extension may
   be ignored" [RFC3550] stands.  To be specific, header extensions MUST
   only be used for data that can safely be ignored by the recipient
   without affecting interoperability, and MUST NOT be used when the
   presence of the extension has changed the form or nature of the rest
   of the packet in a way that is not compatible with the way the stream
   is signalled (e.g., as defined by the payload type).  Valid examples
   might include metadata that is additional to the usual RTP
   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  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 to. attached.  This enables
   a central node to make mixing or selection decisions without decoding
   or detailed inspection of the payload.  Thus payload, reducing the needed complexity in
   some types of central RTP nodes.  It can also be used to save decoding resources
   in a WebRTC receiver in a mesh topology, receivers, which if
   it has limited decoding resources, may select can choose to decode only the most relevant RTP
   media streams based on audio activity levels.

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

5.2.3.  Mixer to Client  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  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 rate-controlled.
   Thus, improving robustness might require a lower base encoding
   quality, but has the potential to give 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.  Retransmission

   Support for  Negative Acknowledgements and RTP retransmission as defined by "RTP Retransmission
   Payload Format" [RFC4588] is RECOMMENDED.

   The retransmission scheme in RTP allows flexible application

   As a consequence of
   retransmissions.  Only selected missing supporting the RTP/SAVPF profile, implementations
   will support negative acknowlegdements (NACKs) for RTP data packets
   [RFC4585].  This feedback can be requested by used to inform a sender of the receiver.  It also allows for loss
   of particular RTP packets, subject to the capacity limitations of the
   RTCP feedback channel.  A sender can use this information to prioritise between
   missing packets based on senders knowledge about their content.
   Compared to TCP, RTP retransmission also allows one optimise
   the user experience by adapting the media encoding to give up on a
   packet that despite retransmission(s) still has not been received
   within a time window.

   "WebRTC Media Transport Requirements" [I-D.cbran-rtcweb-data] raises
   two issues that they think makes RTP Retransmission unsuitable compensate for
   WebRTC.  We here consider these issues and explain why they
   known lost packets, for example.

   Senders are in
   fact not a reason REQUIRED to exclude RTP retransmission from understand the tool box
   available Generic NACK message defined
   in Section 6.2.1 of [RFC4585], but MAY choose to WebRTC media sessions.

   The additional latency added by [RFC4588] will exceed the latency
   threshold ignore this feedback
   (following Section 4.2 of [RFC4585]).  Receivers MAY send NACKs for interactive voice and video:
   missing RTP Retransmission packets; [RFC4585] provides some guidelines on when to
   send NACKs.  It is not expected that a receiver will
      require at least one round trip time for send a retransmission request
      and repair packet to arrive.  Thus NACK for
   every lost RTP packet, rather it should consider the general suitability cost of
      using retransmissions will depend on the actual network path
      latency between sending
   NACK feedback, and the end-points.  In many importance of the actual usages lost packet, to make an
   informed decision on whether it is worth telling the
      latency between two end-points will be low enough for sender about a
   packet loss event.

   The RTP
      retransmission to be effective.  Interactive communication Retransmission Payload Format [RFC4588] offers the ability to
   retransmit lost packets based on NACK feedback.  Retransmission needs
   to be used with
      end-to-end delays of 400 ms still provide a fair quality.  Even
      removing half of that care in end-point delays allows functional
      retransmission between end-points on the same continent.  In
      addition, some interactive real-time applications may accept temporary delay spikes to
      allow for retransmission of crucial codec information such an
      parameter sets, intra picture etc, rather than getting no media at
      all.

   The undesirable increase in ensure
   that the retransmitted packet transmission at arrives in time to be useful, but can
   be effective in environments with relatively low network RTT (an RTP
   sender can estimate the point when
   congestion occurs:  Congestion loss will impact RTT to the rate controls
      view of available bit-rate for transmission.  When receivers using
      retransmission one will have to prioritise between performing
      retransmissions and the quality one can achieve with ones
      adaptable codecs.  In many information in
   RTCP SR and RR packets).  The use cases one prefer error free or low
      rates of error with reduced base quality over high degrees of
      error at a higher base quality.

   The WebRTC end-point implementations will need to both select when to
   enable RTP retransmissions based on API settings can also
   increase the forward RTP bandwidth, and measurements of can potentially worsen the actual round trip time.  In addition for each NACK request
   problem if the packet loss was caused by network congestion.  We
   note, however, that retransmission of an important lost packet to
   repair decoder state may be lower cost than sending a
   media sender receives it will need full intra
   frame.  It is not appropriate to blindly retransmit RTP packets in
   response to make a prioritisation based on
   the NACK.  The importance of lost packets and the requested media, the probability that the
   packet will reach the receiver
   likelihood of them arriving in time for being usable, the
   consumption of available bit-rate 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 impact sender believes it is useful
   to send a retransmission of the media quality
   for new encodings.

   To conclude, packet(s) referenced in the issues raised are implementation concerns that an
   implementation needs to take into consideration, they are NACK.  An
   RTP sender is not
   arguments against including a highly versatile and efficient packet
   loss repair mechanism. expected to retransmit every NACKed packet.

6.2.  Forward Error Correction (FEC)

   Support

   The use of Forward Error Correction (FEC) can provide an effective
   protection against some type degree of FEC to combat packet loss, at the effects cost of packet loss is
   beneficial, but is heavily application dependent.  However, some FEC
   mechanisms steady
   bandwidth overhead.  There are encumbered.

   The main benefit from several FEC is the relatively low additional delay
   needed to protect against packet losses.  The transmission schemes that are defined
   for use with RTP.  Some of any
   repair these schemes are specific to a particular
   RTP payload format, others operate across RTP packets should preferably and can be done used
   with a time delay that is
   just larger than any loss events normally encountered.  That way payload format.  It should be noted that using redundancy
   encoding or FEC will lead to increased playout delay, which should be
   considered when choosing the
   repair packet isn't also lost redundancy or FEC formats and their
   respective parameters.

   If an RTP payload format negotiated for use in the same event a WebRTC session
   supports redundant transmission or FEC as the source data.

   The amount a standard feature of repair packets needed varies depending on that
   payload format, then that support MAY be used in the amount
   and pattern of packet loss to be recovered, and on the mechanism used
   to derive repair data.  The later choice also effects the the
   additional delay required to both encode the repair packets and in
   the receiver to be able to recover the lost packet(s).

6.2.1.  Basic Redundancy

   The method for providing basic redundancy is to simply retransmit a
   some time earlier sent packet.  This is relatively simple in theory,
   i.e. one saves any outgoing source (original) packet in a buffer
   marked with a timestamp of actual transmission, some X ms later one
   transmit this packet again.  Where X is selected to be longer than
   the common loss events.  Thus any loss events shorter than X can be
   recovered assuming that one doesn't get an another loss event before
   all the packets lost in the first event has been received.

   The downside of basic redundancy is the overhead.  To provide each
   packet with once chance of recovery, then the transmission rate
   increases with 100% as one needs to send each packet twice.  It is
   possible to only redundantly send really important packets thus
   reducing the overhead below 100% for some other trade-off is
   overhead.

   In addition the basic retransmission of the same packet using the
   same SSRC in the same RTP session is not possible in RTP context.
   The reason is that one would then destroy the RTCP reporting if one
   sends the same packet twice with the same sequence number.  Thus one
   needs more elaborate mechanisms.

   RTP Payload Format Support:  Some RTP payload format do support basic
      redundancy within the RTP paylaod format itself.  Examples are
      AMR-WB [RFC4867] and G.719 [RFC5404].

   RTP Payload for Redundant Audio Data:  This audio and text redundancy
      format defined in [RFC2198] allows for multiple levels of
      redundancy with different delay in their transmissions, as long as
      the source plus payload parts to be redundantly transmitted
      together fits into one MTU.  This should work fine for most
      interactive audio and text use cases as both the codec bit-rates
      and the framing intervals normally allow for this requirement to
      hold.  This payload format also don't increase the packet rate, as
      original data and redundant data are sent together.  This format
      does not allow perfect recovery, only recovery of information
      deemed necessary for audio, for example the sequence number of the
      original data is lost.

   RTP Retransmission Format:  The RTP Retransmission Payload format
      [RFC4588] can be used to pro-actively send redundant packets using
      either SSRC or session multiplexing.  By using different SSRCs or
      a different session for the redundant packets the RTCP receiver
      reports will be correct.  The retransmission payload format is
      used to recover the packets original data thus enabling a perfect
      recovery.

   Duplication Grouping Semantics in the Session Description Protocol:
      This [I-D.begen-mmusic-redundancy-grouping] is proposal for new
      SDP signalling to indicate media stream duplication using
      different RTP sessions, or different SSRCs to separate the source
      and the redundant copy of the stream.

6.2.2.  Block Based FEC

   Block based redundancy collects a number of source packets into a
   data block for processing.  The processing results in some number of
   repair packets that is then transmitted to the other end allowing the
   receiver to attempt to recover some number of lost packets in the
   block.  The benefit of block based approaches is the overhead which
   can be lower than 100% and still recover one or more lost source
   packet from the block.  The optimal block codes allows for each
   received repair packet to repair a single loss within the block.
   Thus 3 repair packets that are received should allow for any set of 3
   packets within the block to be recovered.  In reality one commonly
   don't reach this level of performance for any block sizes and number
   of repair packets, and taking the computational complexity into
   account there are even more trade-offs to make among the codes.

   One result of the block based approach is the extra delay, as one
   needs to collect enough data together before being able to calculate
   the repair packets.  In addition sufficient amount of the block needs
   to be received prior to recovery.  Thus additional delay are added on
   both sending and receiving side to ensure possibility to recover any
   packet within the block.

   The redundancy overhead and the transmission pattern of source and
   repair data can be altered from block to block, thus allowing a
   adaptive process adjusting to meet the actual amount of loss seen on
   the network path and reported in RTCP.

   The alternatives that exist for block based FEC with RTP are the
   following:

   RTP Payload Format for Generic Forward Error Correction:  This RTP
      payload format [RFC5109] defines an XOR based recovery packet.
      This is the simplest processing wise that an block based FEC
      scheme can be.  It also results in some limited properties, as
      each repair packet can only repair a single loss.  To handle
      multiple close losses a scheme of hierarchical encodings are need.
      Thus increasing the overhead significantly.

   Forward Error Correction (FEC) Framework:  This framework
      [I-D.ietf-fecframe-framework] defines how not only RTP packets but
      how arbitrary packet flows can be protected.  Some solutions
      produced or under development in FECFRAME WG are RTP specific.
      There exist alternatives supporting block codes such as Reed-
      Salomon and Raptor.

6.2.3.  Recommendations for FEC

   Open Issue: Decision of need for FEC and if to be included in
   recommendation which FEC scheme to be supported needs to be
   documented.

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

   WebRTC will be used in very varied network environment with a
   hetrogenous set of link technologies, including wired and wireless,
   interconnecting peers at different topological locations resulting in
   network paths with widely varying one way delays, bit-rate capacity,
   load levels and traffic mixes.  In addition individual end-points
   will open one or more WebRTC sessions between one or more peers.
   Each of these session may contain different mixes of media and data
   flows.  Assymetric usage of media bit-rates and number of media
   streams is also to be expected.  A single end-point may receive zero
   to many simultanous media streams while itself transmitting one or
   more streams.

   The WebRTC application is very dependent from a quality perspective
   on the media adapation working well so that an end-point doesn't
   transmit significantly more than the path is capable of handling.  If
   it would, the result would be high levels of packet loss or delay
   spikes causing media degradations.

   WebRTC applications using more than a single media stream of any
   media type or data flows has an additional concern.  In this case the
   different flows should try to avoid affecting each other negatively.
   In addition in case there is a resource limiation, the available
   resources needs to be shared.  How to share them is something the
   application should prioritize so that the limiation in quality or
   capabilities are the ones that provide the least affect on the
   application.

   This hetrogenous situation results in a requirement to have
   functionality that adapts to the available capacity and that competes
   fairly with other network flows.  If it would not compete fairly
   enough WebRTC could be used as an attack method for starving out
   other traffic on specific links as long as the attacker is able to
   create traffic across a specific link.  This is not far-fetched for a
   web-service capable of attracting large number of end-points and use
   the service, combined with BGP routing state a server could pick
   client pairs to drive traffic WebRTC session,
   subject to specific paths.

   The above estalish a clear need based on any appropriate signalling.

   There are several reasons why there
   need to be a well working media adaptation mechanism.  This mechanism
   also have a number of requirements on what services it should provide
   and what performance it needs to provide.

   The biggest issue is block-based FEC schemes that there are no standardised and ready to use
   mechanism that can simply be included in WebRTC.  Thus there will be
   need designed for use
   with RTP independent of the IETF to produce such a specification.  Therefore chosen RTP payload format.  At the
   suggested way forward time
   of this writing there is to specify requirements no consensus on any solution for
   the media adaptation.  These requirements which, if any, of these FEC
   schemes is appropriate for now proposed to be
   documented use in the WebRTC context.  Accordingly,
   this specification.  In addition a proposed detailed
   solution will be developed, but is expected to take longer time to
   finalize than this document.

7.1.  Congestion Control Requirements

   Requirements for congestion control memo makes no recommendation on the choice of block-based FEC
   for WebRTC sessions are discussed
   in [I-D.jesup-rtp-congestion-reqs].

   Implementations are REQUIRED to implement the RTP circuit breakers
   described in [I-D.perkins-avtcore-rtp-circuit-breakers].

7.2.  Rate Control Boundary Conditions

   The session establishment signalling will establish certain boundary
   that the media bit-rate adaptation can act within.  First use.

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

   WebRTC will be used in very varied network environment with a
   heterogeneous set of media codecs provide practical limitations link technologies, including wired and wireless,
   interconnecting peers at different topological locations resulting in the supported
   bit-rate span where it can provide useful quality, which
   packetization choices that exist.  Next the signalling can establish
   maximum media
   network paths with widely varying one way delays, bit-rate boundaries using SDP b=AS capacity,
   load levels and traffic mixes.  In addition, individual end-points
   will open one or b=CT.

7.3.  RTCP Limiations

   Experience with the congestion control algorithms more WebRTC sessions between one or more peers.
   Each of TCP [RFC5681],
   TFRC [RFC5348], these session may contain different mixes of media 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 capabilities data
   flows.  Asymmetric usage of real-time media
   traffic to adapt to changing path conditions may be less rapid than
   for the elastic applications TCP was designed for, but frequent
   feedback bit-rates and number of RTP media
   streams is still required also to allow the congestion control algorithm be expected.  A single end-point may receive zero
   to track the path dynamics. many simultaneous RTP media streams while itself transmitting one
   or more streams.

   The total RTCP bandwidth WebRTC application is limited in its transmission rate to very dependent from a
   fraction of quality perspective
   on the RTP traffic (by default 5%).  RTCP packets are larger
   than, e.g., TCP ACKs (even when non-compound RTCP packets are used).
   The media stream bit rate thus limits adaptation working well so that an end-point doesn't
   transmit significantly more than the maximum feedback rate as a
   function path is capable of handling.  If
   it would, the mean RTCP packet size.

   Interactive communication may not result would be able to afford waiting for high levels of packet losses to occur to indicate congestion, because an increase in
   playout loss or delay due
   spikes causing media quality degradation.

   WebRTC applications using more than a single RTP media stream of any
   media type or data flows have an additional concern.  In this case,
   the different flows should try to queuing (most prominent avoid affecting each other
   negatively.  In addition, in wireless networks)
   may easily lead to packets being dropped due to late arrival at case there is a resource limitation, the
   receiver.  Therefore, more sophisticated cues may
   available resources need to be reported
   -- shared.  How to be defined in a suitable congestion control framework as noted
   above -- which, in turn, increase share them is
   something the report size again.  For
   example, different RTCP XR report blocks (jointly) provide application should prioritize so that the
   necessary details to implement a variety of congestion control
   algorithms, but limitations
   in quality or capabilities are those that have the (compound) report size grows quickly.

   In group communication, least impact on
   the share application.

   Overall, the diversity of RTCP bandwidth needs operating environments lead to the need for
   functionality that adapts to be
   shared by all group members, reducing the available 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 that competes
   fairly with other network flows.  If it would not compete fairly
   enough WebRTC could thus send a report of 100 bytes about every 70ms or be used as an attack method for every starving out
   other frame traffic on specific links as long as the attacker is able to
   create traffic across the links in question.  A possible attack
   scenario is to use a 30 fps video.

7.4.  Legacy Interop Limitations

   Congestion control interoperability with most type web-service capable of legacy devices,
   even using an translator could be difficult.  There are numerous
   reasons for this:

   No RTCP Support:  There exist legacy implementations that does not
      even implement RTCP at all.  Thus no feedback at all is provided.

   RTP/AVP Minimal RTCP Interval attracting large numbers
   of 5s:  RTP [RFC3550] under end-points, combined with BGP routing state to have the RTP/AVP
      profile specifies server
   pick client pairs to drive traffic to specific paths.

   The above clearly motivates the need for a recommended minimal fixed interval well working media
   adaptation mechanism.  This mechanism also have a number of 5
      seconds.  Sending RTCP report blocks as seldom as 5 seconds makes
   requirements on what services it very difficult for a sender should provide and what performance
   it needs to use these reports provide.

   The biggest issue is that there are no standardised and react ready to
      any congestion event.

   RTP/AVP Scaled Minimal Interval:  If use
   mechanism that can simply be included in WebRTC.  Thus, there will be
   a legacy device uses need for the scaled
      minimal RTCP compound interval, IETF to produce such a specification.  Therefore, the "RECOMMENDED value
   suggested way forward is to specify requirements on any solution for
   the media adaptation.  For now, we propose that these requirements be
   documented in this specification.  In addition, a proposed detailed
   solution will be developed, but is expected to take longer time to
   finalize than this document.

7.1.  Congestion Control Requirements

   Requirements for the
      reduced minimum congestion control of WebRTC sessions are discussed
   in seconds is 360 divided by [I-D.jesup-rtp-congestion-reqs].

   Implementations are REQUIRED to implement the session bandwidth RTP circuit breakers
   described in kilobits/second" ([RFC3550], section 6.2). [I-D.perkins-avtcore-rtp-circuit-breakers].

   (tbd: Should add the RTP/RTCP Mechanisms that an WebRTC
   implementation is required to support.  Potential candidates include
   Transmission Timestamps (RFC 5450).)

7.2.  Rate Control Boundary Conditions

   The minimal
      interval drops below a second, still several times session establishment signalling will establish certain boundary
   that the RTT in
      almost media bit-rate adaptation can act within.  First of all paths in the Internet, when the session bandwidht
      becomes 360 kbps.  A session bandwidth of 1 Mbps still has a
      minimal interval
   set of 360 ms.  Thus, with media codecs provide practical limitations in the exception for rather
      high bandwidth sessions, getting frequent enough RTCP Report
      Blocks to report supported
   bit-rate span where it can provide useful quality, which
   packetization choices that exist.  Next the signalling can establish
   maximum media bit-rate boundaries using SDP b=AS or b=CT.

   (tbd: This section needs expanding on how to use these limits)

7.3.  RTCP Limitations for Congestion Control

   Experience with the order 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 RTT is very difficult real-time media traffic may not
   have to adapt to changing path conditions as long rapidly as the legacy device uses the RTP/AVP profile.

   RTP/AVPF Supporting Legacy Device:  If a legacy device supports RTP/
      AVPF, then that enables negotation of important parameters needed for
      frequent reporting, such as the "trr-int" parameter, and
   the
      possibility that elastic applications TCP was designed for, but frequent feedback
   is still required to allow the end-point supports some useful feedback
      format for congestion control purpose such as TMMBR [RFC5104].

   It has been suggested on algorithm to track
   the WebRTC mailing list that if
   interoperating with really path dynamics.

   The total RTCP bandwidth is limited legacy devices an WebRTC end-point
   may not send more than 64 kbps of media streams, to avoid it causing
   massive congestion on most paths in its transmission rate to a
   fraction of the Internet RTP traffic (by default 5%).  RTCP packets are larger
   than, e.g., TCP ACKs (even when communicating
   with a legacy node not providing sufficient non-compound RTCP packets are used).
   The RTP media stream bit rate thus limits the maximum feedback for effective
   congestion control.  This warrants further discussion rate
   as there is
   clearly a number of link layers that don't even provide that amount
   of bit-rate consistently, and that assumes no competing traffic.

8.  WebRTC Use function of RTP: Performance Monitoring the mean RTCP does contains a basic set of RTP flow monitoring points like packet loss and jitter.  There exist a number of extensions that
   could size.

   Interactive communication may not be included able to afford waiting for
   packet losses to occur to indicate congestion, because an increase in the set
   playout delay due to be supported.  However, queuing (most prominent in most cases
   which RTP monitoring that is needed depends on the application, which
   makes it difficult wireless networks)
   may easily lead to select which packets being dropped due to include when late arrival at the set of
   applications is very large.

   Exposing some metrics
   receiver.  Therefore, more sophisticated cues may need to be reported
   -- to be defined in a suitable congestion control framework as noted
   above -- which, in turn, increase the WebRTC API should be considered allowing report size again.  For
   example, different RTCP XR report blocks (jointly) provide the application
   necessary details to gather the measurements implement a variety of interest.  However,
   security implications for congestion control
   algorithms, but the different data sets exposed will need
   to be considered in this.

9.  WebRTC Use of RTP: Future Extensions

   It is possible that (compound) report size grows quickly.

   In group communication, the core set share of RTP protocols and RTP extensions
   specified in this memo will prove insufficient for the future RTCP bandwidth 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
   shared by all group members, reducing the RTP
   Control Protocol [RFC5968], and SHOULD take into account any future
   guidelines for extending RTP capacity and related protocols that have been
   developed.

   Authors of future extensions are urged to consider thus the wide range
   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
   environments 100 bytes about every 70ms or
   for every other frame in which RTP is used when recommending extensions, since
   extensions that a 30 fps video.

7.4.  Congestion Control Interoperability With Legacy Systems

   There are applicable in some scenarios can be problematic
   in others.  Where possible, the WebRTC framework should adopt RTP
   extensions legacy implementations that are of general utility, do not implement RTCP, and
   hence do not provide any congestion feedback.  Congestion control
   cannot be performed with these end-points.  WebRTC implementations
   that must interwork with such end-points MUST limit their
   transmission to enable easy gatewaying a low rate, equivalent to
   other applications a VoIP call using RTP, rather than adopt mechanisms a low
   bandwidth codec, that are
   narrowly targetted at specific WebRTC use cases.

10.  Signalling Considerations

   RTP is built unlikely to cause any significant
   congestion.

   When interworking with legacy implementations that support RTCP using
   the assumption of an external signalling channel RTP/AVP profile [RFC3551], congestion feedback is provided in
   RTCP RR packets every few seconds.  Implementations that can be used are required
   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 interwork with such end-points MUST ensure that they keep within
   the RTP profile circuit breaker [I-D.perkins-avtcore-rtp-circuit-breakers]
   constraints to be used in session.  The
      RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles limit the congestion they can interoperate
      on basic level, cause.

   If a legacy end-point supports RTP/AVPF, this enables negotiation of
   important parameters for frequent reporting, such as can their secure variants RTP/SAVP [RFC3711] the "trr-int"
   parameter, and RTP/SAVPF [RFC5124].  The secure variants of the profiles do
      not directly interoperate with possibility that the non-secure variants, due end-point supports some
   useful feedback format for congestion control purpose such as TMMBR
   [RFC5104].  Implementations that are required to interwork with such
   end-points MUST ensure that they stay within the
      presence of additional header fields in addition RTP circuit breaker
   [I-D.perkins-avtcore-rtp-circuit-breakers] constraints to any
      cryptographic transformation of the packet content.  As WebRTC
      requires the usage of limit the SAVPF profile only a single profile will
      need to be signalled.  Interworking functions
   congestion they can cause, but may transform this
      into SAVP for a legacy use case by indicating to find that they can achieve better
   congestion response depending on the amount of feedback that is
   available.

8.  WebRTC end-
      point a SAVPF end-point and limiting the usage Use of the a=rtcp
      attribute to indicate RTP: Performance Monitoring

   RTCP does contains a trr-int value basic set of 4 seconds.

   Transport Information:  Source and destination address(s) and ports
      for RTP flow monitoring metrics like
   packet loss and RTCP MUST jitter.  There are a number of extensions that could
   be signalled for each RTP session.  In
      WebRTC these end-points will included in the set to be provided by ICE that signalls
      candidates and arrive at nominated candidate pairs.  If supported.  However, in most cases which
   RTP and
      RTCP multiplexing [RFC5761] monitoring that is needed depends on the application, which makes
   it difficult to be used, such that a single port select which to include when the set of applications
   is used for RTP and RTCP flows, this MUST very large.

   Exposing some metrics in the WebRTC API should be signalled (see
      Section 4.5).  If several RTP sessions are considered allowing
   the application 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 gather the RTP
      payload formats measurements of interest.  However,
   security implications for the different data sets exposed will need
   to be used) and the RTP payload type numbers must considered in this.

   (tbd: If any RTCP XR metrics should be signalled.  Each media type may also have added is still an open
   question, but possible to extend at a number later stage)

9.  WebRTC Use of media
      type parameters RTP: Future Extensions

   It is possible that must also be signalled to configure the codec
      and RTP payload format (the "a=fmtp:" line from SDP). core set of RTP Extensions:  The protocols and RTP extensions one intends to use need to be
      agreed upon, including any parameters for each respective
      extension.  At the very least,
   specified in this memo will help avoiding using
      bandwidth prove insufficient for features that the other end-point will ignore.  But
      for certain mechanisms there is requirement for future needs
   of WebRTC applications.  In this case, future updates to happen as
      interoperability failure otherwise happens.

   RTCP Bandwidth:  Support for exchanging RTCP Bandwidth values to the
      end-points will this memo
   MUST be necessary, as described in "Session Description
      Protocol (SDP) Bandwidth Modifiers made following the Guidelines for Writers of RTP Payload
   Format Specifications [RFC2736] and Guidelines for Extending the RTP
   Control Protocol (RTCP)
      Bandwidth" [RFC3556], or something semantically equivalent.  This
      also ensures [RFC5968], and SHOULD take into account any future
   guidelines for extending RTP and related protocols that the end-points have a common view of the RTCP
      bandwidth, this is important as too different view been
   developed.

   Authors of the
      bandwidths may lead to failure to interoperate.

   These parameters future extensions 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 somehow, and provided urged to consider the wide range of
   environments in which RTP implementation.  We note is used when recommending extensions, since
   extensions that in the WebRTC context it will depend on the signalling model and
   API how these parameters need to be configured but they will are applicable in some scenarios can be need
   to either set problematic
   in the API or explicitly signalled between the peers.

11.  WebRTC API Considerations

   The following sections describe how others.  Where possible, the WebRTC API features map onto
   the framework should adopt RTP mechanisms described in this memo.

11.1.  API MediaStream
   extensions that are of general utility, to RTP Mapping

   The enable easy gatewaying to
   other applications using RTP, rather than adopt mechanisms that are
   narrowly targeted at specific WebRTC API and its media function have use cases.

10.  Signalling Considerations

   RTP is built with the concept assumption of a
   MediaStream an external signalling channel
   that consists can be used to configure the RTP sessions and their features.
   The basic configuration of zero or more tracks.  Where a track is an individual stream of media from any type RTP session consists of media source like a
   microphone or a camera, but also coneptual sources, like a audio mix
   or a video composition. the following
   parameters:

   RTP Profile:  The tracks within a MediaStream are expected
   to be synchronized.

   A track correspondes to name of 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 a media stream profile to be used in session.  The
      RTP/AVP [RFC3551] and its timing.

   Thus a MediaStream is a collection 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 SSRCs carrying the different
   media included 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 synchornized aggregate.  Thus also the
   synchronization state associated with packet content.  As WebRTC
      requires the included SSRCs are part usage of
   concept.  One important thing to consider is that there the RTP/SAVPF profile this can be
   multiple different MediaStreams containing inferred
      as there is only a given Track (SSRC).
   Thus single profile, but in SDP this is still
      required information to avoid unnecessary duplication of media at transport level one
   need be signalled.  Interworking functions may
      transform this into RTP/SAVP for a legacy use case by indicating
      to do the binding of which MediaStreams WebRTC end-point a given SSRC is
   associated with at signalling level.

   A proposal for how the binding between MediaStreams RTP/SAVPF end-point and SSRC can be
   done exist in "Cross Session Stream Identification in the Session
   Description Protocol" [I-D.alvestrand-rtcweb-msid].

12.  RTP Implementation Considerations

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

   This section discusses and RTCP MUST be signalled for each RTP functionality 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 part of used for RTP and RTCP flows, this MUST be
      signalled (see Section 4.5).  If several RTP sessions are to be
      multiplexed onto a single transport layer flow, this MUST also be
      signalled (see Section 4.4).

   RTP Payload Types, media formats, and media format
   parameters:  The mapping between media type names (and hence the RTP
   standard, required by decisions made, or
      payload formats to enable use cases raised be used) and their motivations.  This discussion is done from an WebRTC end-
   point perspective.  It will occassional go into central nodes, but as the specification is for an end-point RTP payload type numbers MUST
      be signalled.  Each media type MAY also have a number of media
      type parameters that is where the focus lies.
   For more discussion on MUST also be signalled to configure the central nodes codec
      and details about RTP
   topologies please reveiw Appendix A. payload format (the "a=fmtp:" line from SDP).

   RTP Extensions:  The section will touch on RTP extensions to be used SHOULD be agreed upon,
      including any parameters for each respective extension.  At the relation with certain RTP/RTCP
   extensions, but
      very least, this will focus on the RTP core functionality.  The
   definition of what functionalities and help avoiding using bandwidth for features
      that the level of requirement on
   implementing it other end-point will ignore.  But for certain mechanisms
      there is defined 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 Section 2.

12.1.  RTP Sessions and PeerConnection

   An RTP session is an association among
      "Session Description Protocol (SDP) Bandwidth Modifiers for RTP nodes, which
      Control Protocol (RTCP) Bandwidth" [RFC3556], or something
      semantically equivalent.  This also ensures that the end-points
      have one a common SSRC space.  An view of the RTCP bandwidth, this is important as too
      different view of the bandwidths may lead to failure to
      interoperate.

   These parameters are often expressed in SDP messages conveyed within
   an offer/answer exchange.  RTP session can include any number of end-
   points and nodes sourcing, sinking, manipulating does not depend on SDP or reporting on the
   media streams being sent within
   offer/answer model, but does require all the RTP session.  A PeerConnection
   being a point necessary parameters to point association between an end-point and another
   node.  That peer node may
   be both an end-point or centralized
   processing node of some type, thus agreed upon, and provided to the RTP session may terminate
   immediately on implementation.  We note that
   in the far end of WebRTC context it will depend on the PeerConnection, signalling model and API
   how these parameters need to be configured but it may also
   continue as further discused below they will be need to
   either set in Multiparty (Section 12.3) and
   Multiple RTP End-points (Section 12.7).

   A PeerConnection can contain one the API or more RTP session depending on how
   it is setup and explicitly signalled between the peers.

11.  WebRTC API Considerations

   The following sections describe how many UDP flows it uses.  A common usage has been the WebRTC API features map onto
   the RTP mechanisms described in this memo.

11.1.  API MediaStream to have one RTP session per media type, e.g. one for audio Mapping

   The WebRTC API and one
   for Video, each sent over different UDP flows.  However, its media function have the default
   usage in 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 will
   MediaStream are expected to be synchronized.

   A track correspond to use one RTP session for all the media types.
   This usage then uses only received with one UDP flow, as also particular SSRC.
   There might be additional SSRCs associated with that SSRC, like for
   RTP and RTCP
   multiplexing is mandated (Section 4.5). retransmission or Forward Error Correction.  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 SSRC
   will identify an RTP
   session per media type is used.  Currently each RTP session must use stream and its own UDP flow.  Discussion are ongoing if timing.

   As a solution enabling
   multiple RTP sessions over result, a single UDP flow, see Section 4.4.

   The multi-unicast or mesh based multi-party topology (Figure 1) WebRTC MediaStream is
   best to raise a collection of SSRCs carrying
   the different media included in this section as it concers the relation between RTP
   sessions and PeerConnections.  In this topology, each participant
   sends individual unicast RTP/UDP/IP flows synchronised aggregate.
   Therefore, also the synchronization state associated with the
   included SSRCs are part of concept.  It is important to each consider that
   there can be multiple different WebRTC MediaStreams containing a
   given Track (SSRC).  To avoid unnecessary duplication of media at the other
   participants using independent PeerConnections
   transport level in such cases, a full mesh. 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
   topology has the benefit of not requiring central nodes. text must be improved and achieved consensus on.  Interim
   meeting in June 2012 shows large differences in opinions.)

12.  RTP Implementation Considerations

   The
   downside is that it increases following provide some guidance on the used bandwidth at each sender by
   requiring one copy implementation of the media streams for each participant RTP
   features described in this memo.

   This section discusses RTP functionality that are is part of the same session beyond the sender itself.  Hence, this
   topology is limited RTP
   standard, required by decisions made, or to scenarios with few participants unless the
   media enable use cases raised
   and their motivations.  This discussion is very low bandwidth.

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

                          Figure 1: Multi-unicast

   The multi-unicast topology could be implemented from an WebRTC end-point
   perspective.  It will occasionally talk about central nodes, but as a single
   this specification is for an end-point, this is where the focus lies.
   For more discussion on the central nodes and details about RTP
   session, spanning multiple peer-to-peer transport layer connections,
   or as several pairwise
   topologies please see Appendix A.

   The section will touch on the relation with certain RTP/RTCP
   extensions, but will focus on the RTP sessions, one between each pair core functionality.  The
   definition of peers.
   To maintain a coherent mapping between what functionalities and the relation between level of requirement on
   implementing it is defined in Section 2.

12.1.  RTP
   sessions Sessions and PeerConnections we recommend that one implements this as
   individual PeerConnection

   An RTP session is an association among RTP nodes, which have one
   common SSRC space.  An RTP sessions.  The only downside is that end-point A will
   not learn of the quality of session can include any transmission happening between B number of end-
   points and
   C based nodes sourcing, sinking, manipulating or reporting 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
   differnt peers, thus not forcing B to endure the same quality
   reductions if there are limiations in streams being sent within the transport from A to C as C
   will.

12.2.  Multiple Sources RTP session.  A WebRTC end-point may have multiple cameras, microphones or audio
   inputs thus
   PeerConnection being a single point-to-point association between an end-
   point and another node.  That peer node may be both an end-point can source multiple media streams
   concurrently or
   centralized processing node of some type; thus, the same media type.  In addition RTP session may
   terminate immediately on the above far end of the PeerConnection, but it
   may also continue as further discussed
   criteria to support multiple media types below in Multiparty
   (Section 12.3) and Multiple RTP End-points (Section 12.7).

   A PeerConnection can contain one single or more RTP session
   results that also an end-point that 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 audio and RTP session for all media types.
   This usage then uses only one video
   source still need two transmit using two SSRCs concurrently.  As
   multi-party conferences are supported, UDP flow, as discussed below in
   Section 12.3, 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 will need needs to be capable support a mode of
   receiving, decoding and playout multiple operation where
   one RTP session per media streams of the same type concurrently.

   Open Issue:Are any mechanism needed to signal limiations in the
   number of SSRC that an end-point can handle?

12.3.  Multiparty

   There exist numerous situations and clear is used.  Currently, each RTP session
   must use cases for WebRTC
   supporting sessions supoprting multi-party.  This can be realized in its own UDP flow.  Discussions are ongoing if a number of ways using solution
   enabling multiple RTP sessions over a number of different implementations
   strategies.  This focus on the different set of WebRTC end-point
   requirements that arise from different sets of multi-party
   topologies. single UDP flow, see
   Section 4.4.

   The multi-unicast mesh multi-unicast- or mesh-based multi-party topology (Figure 1) based multi-party topoology
   discussed above provides is a non-centralized solution but can easily
   tax
   good example for this section as it concerns the end-points outgoing paths.  It may also consume large amount
   of encoding resources if relation between RTP
   sessions and PeerConnections.  In this topology, each outgoing stream is specifically
   encoded.  If an encoding is transmitted participant
   sends individual unicast RTP/UDP/IP flows to multiple parties, either
   as in each of the mesh case or when other
   participants using relaying independent PeerConnections in a full mesh.  This
   topology has the benefit of not requiring central nodes (see below)
   a requirement on nodes.  The
   downside is that it increases the end-point becomes to be able to create used bandwidth at each sender by
   requiring one copy of the RTP media streams suitable to multiple destinations requirements.  These
   requirements may both be dependent on transport path and for each participant that
   are part of the
   different end-points preferences related same session beyond the sender itself.  Hence, this
   topology is limited to playout of scenarios with few participants unless the media.
   media is very low bandwidth.

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

                          Figure 2: 1: Multi-unicast

   The multi-unicast topology could be implemented as a single RTP Mixer with Only Unicast Paths
   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 Mixer (Figure 2) is an RTP to C as C
   will.

12.2.  Multiple Sources

   A WebRTC end-point that optimizes the
   transmission of may have multiple cameras, microphones or audio
   inputs and thus a single end-point can source multiple RTP media
   streams from certain perspectives, either by
   only sending some of the received same media stream to any given receiver
   or by providing a combined type concurrently.  Even if an end-point
   does not have multiple media stream out of a set of contributing
   streams.  There exist various methods of implementation as discussed
   in Appendix A.3.  A common aspect is that these central nodes a
   number sources of tools to control the same media encoding provided by a WebRTC
   end-point.  This includes functions like requesting breaking the
   encoding chain and have type it will
   be required to support transmission using multiple SSRCs concurrently
   in the encoder produce a so called Intra frame.
   Another same RTP session.  This is limiting the bit-rate of a given stream due to better suit the
   mixer view of the requirement on an WebRTC
   end-point to support multiple down-streams.  Others are controling media types in one RTP session.  For
   example, one audio and one video source can result in the
   most suitable frame-rate, picture resultion, end-point
   sending with two different SSRCs in the trade-off between
   frame-rate and spatial quality.

   A mixer gets a significant responsibility to correctly perform
   congestion control, identity management, manage synchronization while
   providing same RTP session.  As multi-
   party conferences are supported, as discussed below in Section 12.3,
   a for the application suitable media optimization.

   Mixers also WebRTC end-point will need to be a trusted node when it comes to security as it
   manipulates either be capable of receiving, decoding and
   playout multiple RTP or the media itself before sending it on
   towards streams of the end-point(s) thus must be able to decrypt and then
   encrypt it before sending it out.  There exist one same type of central
   node, concurrently.

   tbd: Are any mechanism needed to signal limitations in the relay number of
   SSRC that one doesn't need to trust with 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 keys to following, the
   media.  The relay operates only focus is on the IP/UDP level different set of the transport.
   It is configured so
   WebRTC end-point requirements that it would forward any RTP/RTCP packets arise from A
   to the other participants B-D.

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

             Figure 3: DTLS-SRTP host and RTP Relay Separated

   To accomplish the security properties different sets of
   multi-party topologies.

   The multi-unicast mesh (Figure 1)-based multi-party topology
   discussed above using provides a relay
   one need to have non-centralized solution but may incur a separate key handling server and also support for
   distribute
   heavy tax on the different keys such as Encrypted Key Transport
   [I-D.ietf-avt-srtp-ekt].  The relay end-points' outgoing paths.  It may also creates a situation where
   there consume
   large amount of encoding resources if each outgoing stream is
   specifically encoded.  If an encoding is transmitted to multiple end-points visible
   parties, as in some implementations of the RTCP reporting and any
   feedback events.  Thus becoming yet another situation in addition to
   Mesh where mesh case, a requirement
   on the end-point will have becomes to have logic be able to create RTP media streams
   suitable for merging
   different multiple destinations requirements.  These requirements
   may both be dependent on transport path and preferences.  This is more detail
   discussed in Section 12.7.

                          +---+ the different end-points
   preferences related to playout of the media.

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

                Figure 4: MediaStream Forwarding

   The above Figure 4 depicts a possible scenario where 2: RTP Mixer with Only Unicast Paths

   A Mixer (Figure 2) is an WebRTC end-
   point (A) sends a RTP end-point that optimizes the
   transmission of RTP media stream to B. B decides to forward streams from certain perspectives, either
   by only sending some of the received RTP media stream to C. This can either be realized in B (WebRTC end-point)
   using any given
   receiver or by providing a simple relay functionality creating similar consideration and combined RTP media stream out of a set of
   contributing streams.  There are various methods of implementation requirements.  Another implmentation strategy as
   discussed in B
   could be to select Appendix A.3.  A common aspect is that these central
   nodes may use a number of tools to transcode 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 media from A bit-rate of a given
   stream to C, thus breaking
   most better suit the mixer view of the dependecies multiple down-streams.
   Others are controlling the most suitable frame-rate, picture
   resolution, the trade-off between A frame-rate and C. In that case spatial quality.

   A is not
   required 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 aware of B forwarding 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 must be able to C. 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. i.e., resolve the
   conflict when two different end-points uses use the same SSRC value.  This
   requirement
   applies also applies to WebRTC end-points.  There exist are several
   scenarios where SSRC collisions may occur.

   In a point to point point-to-point session where each SSRC are is associated with either
   of the two end-points and where the main media carrying SSRC
   identifier will be announced in the signalling there channel, a collision
   is less likely to occur due to the information about used SSRCs
   provided by Source-
   Specific Source-Specific SDP Attributes [RFC5576].  Still if both
   end-points starts start uses an new SSRC identifier prior to having
   signalled it to the peer and received acknowledgement on the
   signalling message 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 unsignalled SSRCs, this may be considered a
   bug. SSRCs.  This is more likely than
   it appears as certain RTP
   functionalities functions need extra SSRCs to provide
   functionality related to another (the "main") SSRC, for example example, SSRC
   multiplexed RTP retransmission [RFC4588].  In those cases cases, an end-point end-
   point can create a new SSRC which that strictly don't doesn't need to be
   announced over the signalling channel to function correctly on both
   RTP and PeerConnection level.

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

   Another scenario is when the central node manage manages to connect an end-
   points
   point's PeerConnection to another PeerConnectio PeerConnection the end-point it has.
   Thus
   already has, thus forming a loop where the end-point will receive its
   own traffic.
   This must be  While is is clearly considered a bug, but still if it occurs it is important
   that the end-point can is able to recognise and handle the situation. case when it
   occurs.

12.5.  Contributing Sources

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

   (tbd: should the API 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 is dependent depends on the CRSRC list
   being used, that is used: the Mixer to client Mixer-to-client audio level [RFC6465],
   that which enhances
   the information provided by the CSRC to actual energy levels for
   audio for each contributing source.

12.6.  Media Synchronization

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

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

12.7.  Multiple RTP End-points

   A number of

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

   The topologies relevant to WebRTC when this can occur are centralized
   relay and

   RTP Mixers may create a situation where an end-point forwarding experiences a media stream.
   situation in-between a session with only two end-points and multiple
   end-points.  Mixers are expected to not forward media stream RTCP reports
   regarding RTP media streams across itself themselves.  This is due to the
   difference in the RTP media stream streams provided to different end-points which the different end-
   points.  The original media source lacks information about a mixer's
   manipulations prior to sending it the mixers
   manipulation. different receivers.  This
   setup 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 fulfill the receivers
   request.  This will not necessarily be explicitly visible any RTP and
   RTCP traffic, but the interactions and the time to complete them will
   indicate such dependencies.

   The topologies in which an end-point receives receiver reports from
   multiple other end-points are the centralized relay, multicast and an
   end-point forwarding an RTP media stream.  Having multiple RTP nodes
   receive ones 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  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 challange.
   challenge.  In the mixer based topologies an mixer-based topologies, end-points source
   authentication is based on on, firstly, verifying that media comes from
   the mixer by cryptographic verification and secondly and, secondly, trust in the
   mixer to correctly identify any source towards the end-point.  In RTP
   sessions where multiple end-
   points end-points are directly visible to an end-point end-
   point, all end-points will have knowledge about each others others' master
   keys, and can thus inject packets claimed to come from another end-point end-
   point in the session.  Any node performing relay can perform non-cryptographic non-
   cryptographic mitigation by preventing forwarding of packets that has
   have SSRC fields that has
   previously come fields that came from other end-points. end-points before.  For
   cryptographic verification of the source SRTP will 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 version versions of the
   same media source.  In the WebRTC context that appears to context, this could be most
   easily accomplished by establishing mutliple
   in two ways.  One is to establish multiple PeerConnection all being
   feed the same set of WebRTC MediaStreams.  Each PeerConnection  Another method is then to use
   multiple WebRTC MediaStreams that are differently configured when it
   comes to deliver a particular media quality and thus the media bit-
   rate. parameters.  This will work well as long as would result in that multiple
   different RTP Media Streams (SSRCs) being in used with different
   encoding based on the end-point implements
   indepdentent same media encoding for each PeerConnection and not share source (camera, microphone).

   When intending to use simulcast it is important that this is made
   explicit so that the
   encoder.  Simulcast will fail if end-points don't automatically try to optimize
   away the end-point uses different encodings and provide a single common encoder
   instance version.
   Thus, some explicit indications that the intent really is to multiple PeerConnections.

   Thus it have
   different media encodings is likely required.  It should be considered to explicitly signal which of the two
   implementation strategies noted
   that are desired and which will it might be done.
   At least making the application and possible the a central node
   interested in receiving simulcast of node, rather than an end-points WebRTC end-point that
   would benefit from receiving simulcasted media streams sources.

   tbd: How to perform simulcast needs to be aware if it will function determined and the
   appropriate API or not. signalling for its usage needs to be defined.

12.9.  Differentiated Treatment of Flows

   There exist 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 for
   sending the media, which controls, both which RTP media streams that should
   will be sent, there and their allocation of bit-rate out of the current
   available aggregate as determined by the congestion control.

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

   Diffserv:  The end-point could mark the marks a packet with a diffserv code point to
      indicate to the network how that the WebRTC application and
      browser would like this particular packet treated. belongs to a particular
      class.

   Flow based:  Prioritization of all packets belonging to  Packets that shall be given a particular
      media flow or RTP session by keeping them in separated UDP flows.
      Thus enabling either end-point initiated or network initiated
      prioritization treatment are
      identified using a combination of the flow. 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 due to that because multiple different type types of flows are aggregated and thus it
   becomes more difficult to apply analysis on. analyse them.  The flow based flow-based
   differentiation will provide the same treatment to all packets within
   the flow.  Thus flow, i.e., relative prioritization is not possible.  In addition  Moreover,
   if the resources are limited it may not be possible to provide
   differential treatment compared to best-effort for all the flows in a
   WebRTC application.

   When flow based 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 is based on assumes that either the end-point or a classifier can mark
   the packets with an appropriate DSCP so that the packets is are treated
   according to that marking.  If the end-point is to mark the traffic
   there exist
   two requirements arise in the WebRTC context. context: 1) The first is
   that the WebRTC
   application or browser knows has to know which DSCP to use and that it can
   use them on some set of RTP media streams.  Secondly the 2) The information needs
   to be propagated to the operating system when transmitting the
   packet.

   Open Issue:

   tbd: The model for providing differentiated treatment needs to be
   evolved.  This includes:

   1.  How will the WebRTC application and/or browser know that
   differentiated treatment is desired and available and ensure that it
   gets can prioritize MediaStreamTracks differently
       in the information required to correctly configure API

   2.  How the WebRTC
   multimedia conference. browser or application determine availability of
       transport differentiation

   3.  How to learn about any configuration information for transport
       differentiation, such as DSCPs.

13.  IANA Considerations

   This memo makes no request of IANA.

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

14.  Security Considerations

   RTP and its various extensions each have their own security
   considerations.  These should be taken into account when considering
   the security properties of the complete suite.  We currently don't
   think this suite creates any additional security issues or
   properties.  The use of SRTP [RFC3711] will provide protection or
   mitigation against all most of the fundamental issues by offering
   confidentiality, integrity and partial source authentication.  A
   mandatory to implement media security solution will be required to be
   picked.  We currently don't discuss the key-management aspect of SRTP
   in this memo, that needs to be done taking the WebRTC communication
   model into account.

   Privacy concerns are under discussion and the generation of non-
   trackable CNAMEs are under discussion.

   The guidelines in [I-D.ietf-avtcore-srtp-vbr-audio] [RFC6562] apply when using variable bit rate (VBR)
   audio codecs, for example Opus or the Mixer audio level header
   extensions.

   Security considerations for the WebRTC work are discussed in
   [I-D.ietf-rtcweb-security].

15.  Acknowledgements

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

16.  References

16.1.  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-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-01 (work in
              progress), October 2011.

   [I-D.ietf-avtcore-srtp-vbr-audio]
              Perkins, C.

   [I-D.ietf-avtext-multiple-clock-rates]
              Petit-Huguenin, M. and J. Valin, "Guidelines G. Zorn, "Support for the use of
              Variable Bit Rate Audio with Secure RTP",
              draft-ietf-avtcore-srtp-vbr-audio-04 Multiple
              Clock Rates in an RTP Session",
              draft-ietf-avtext-multiple-clock-rates-05 (work in
              progress),
              December 2011. May 2012.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-03 draft-ietf-rtcweb-overview-04 (work
              in progress), March June 2012.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for RTC-Web",
              draft-ietf-rtcweb-security-02 (work in progress),
              March 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
              draft-ietf-rtcweb-security-03 (work in progress),
              March
              June 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.perkins-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
              Breakers for Unicast Sessions",
              draft-perkins-avtcore-rtp-circuit-breakers-00 (work in
              progress), March 2012.

   [I-D.westerlund-avtcore-multiplex-architecture]
              Westerlund, M., Burman, B., and C. Perkins, "RTP
              Multiplexing Architecture",
              draft-westerlund-avtcore-multiplex-architecture-01 (work
              in progress), March 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-02 (work
              in progress), March 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.
              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.

   [RFC5109]  Li, A., "RTP Payload Format for Generic Forward Error
              Correction", Feedback (AVPF)", RFC 5109, December 2007. 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.

   [RFC6222]  Begen, A., Perkins, C., and D. Wing, "Guidelines for
              Choosing RTP Control Protocol (RTCP) Canonical Names
              (CNAMEs)", RFC 6222, April 2011.

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

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

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

16.2.  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.begen-mmusic-redundancy-grouping]
              Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
              Semantics in the Session Description Protocol",
              draft-begen-mmusic-redundancy-grouping-03 (work in
              progress), March 2012.

   [I-D.cbran-rtcweb-data]
              Bran, C. and C. Jennings, "RTC-Web Non-Media Data
              Transport Requirements", draft-cbran-rtcweb-data-00 (work
              in progress), July 2011.

   [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-fecframe-framework]
              Watson, M., Begen, A.,

   [I-D.ietf-rtcweb-use-cases-and-requirements]
              Holmberg, C., Hakansson, S., and V. Roca, "Forward Error
              Correction (FEC) Framework",
              draft-ietf-fecframe-framework-15 G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements",
              draft-ietf-rtcweb-use-cases-and-requirements-09 (work in
              progress), June 2011.

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, 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., Bolot, J., Vega-Garcia, A., Burman, B., and S. Fosse-
              Parisis, C. Perkins, "RTP Payload for Redundant Audio Data", RFC 2198,
              September 1997.
              Multiplexing Architecture",
              draft-westerlund-avtcore-multiplex-architecture-01 (work
              in progress), March 2012.

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

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

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

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

   [RFC4867]  Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
              "RTP Payload Format and File Storage Format for the
              Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
              (AMR-WB) Audio Codecs", RFC 4867, April 2007.

   [RFC5117]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
              January 2008.

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

   [RFC5404]  Westerlund, M. and I. Johansson, "RTP Payload Format for
              G.719", RFC 5404, January 2009.

   [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 [RFC5117], and we further
   describe those expected to be useful for WebRTC in the following.  We
   also goes into important RTP session aspects that the topology or
   implementation variant can place on a WebRTC end-point.

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

A.1.  Point to Point

   The point-to-point RTP topology (Figure 5) 3) is the simplest scenario
   for WebRTC applications.  This is going to be very common for user to
   user calls.

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

                         Figure 5: 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 must be opened to allow RTP traffic.  To reduce these costs and
   session setup 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
   that these limitations will significantly affect the scenarios
   targetted
   targeted by WebRTC, but they may impact interoperability with legacy
   systems.

   An RTP session have good support for simultanously 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 transmitt 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 are required.

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

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

              Figure 6: 4: Point to Point: Multiple RTP sessions

   As can be seen above in the Point to Point: Multiple RTP sessions
   (Figure 6) 4) the single Peer Connection contains two RTP sessions over
   different UDP flows UDP 1 and UDP 2, i.e. their 5-tuples will be
   different, normally on source and destination ports.  The first RTP
   session (RTP1) carries audio, one stream in each direction AA1 and
   BA1.  The second RTP session contains two video streams from A (AV1
   and AV2) and one from B to A (BV1).

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

               Figure 7: 5: Point to Point: Single RTP session.

   In (Figure 7) 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 8); 6); unfortunately not discussed in the RTP
   Topologies RFC [RFC5117].  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 8: 6: Multi-unicast

   This topology has the benefit of not requiring central nodes.  The
   downside is that it increases the used bandwidth at each sender by
   requiring one copy of the RTP media streams for each participant that
   are part of the same session beyond the sender itself.  Hence, this
   topology is limited to scenarios with few participants unless the
   media is very low bandwidth.  The multi-unicast topology could be
   implemented as a single RTP session, spanning multiple peer-to-peer
   transport layer connections, or as several pairwise RTP sessions, one
   between each pair of peers.  To maintain a coherent mapping between
   the relation between RTP sessions and PeerConnections we recommend
   that one implements this as individual RTP sessions.  The only
   downside is that end-point A will not learn of the quality of any
   transmission happening between B and C based on RTCP.  This has not
   been seen as a significant downside as now one has yet seen a need
   for why A would need to know about the B's and C's communication.  An
   advantage of using separate RTP sessions is that it enables using
   different media bit-rates to the differnt peers, thus not forcing B
   to endure the same quality reductions if there are limiations 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 9: 7: Session strcuture structure for Multi-Unicast Setup

   Lets review how the RTP sessions looks from A's perspective by
   considering both how the media is a handled and what PeerConnections
   and RTP sessions that are setup in Figure 9. 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 costrained will use a different implementation strategy
   where the encoder is shared between the different PeerConnections as
   shown below Figure 10. 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 10: 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 prefered resolution to receive
   based on dispaly area available is another aspect requiring
   consideration.  The need for this type of descion logic does arise in
   several different topologies and implementation.

A.3.  Mixer Based

   An mixer (Figure 11) 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 required in
   stead of 3 in the depicted scenario (Figure 11). 9).

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

                Figure 11: 9: RTP Mixer with Only Unicast Paths

   Mixers has two downsides, the first is that the mixer must be a
   trusted node as they either performs media operations or at least
   repacketize 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 integegrity
   protect them.  This applies to all types of mixers described below.

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

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

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

A.3.1.  Media Mixing

   This type of mixer is one which clearly can be called RTP mixer is
   likely the one that most thinks of when they hear the term mixer.

   Its basic patter of operation is that it will receive the different
   participants RTP media stream.  Select which that are to be included
   in a media domain mix of the incomming 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
   N decodings.  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 may
   produce a Mix of all N streams for the group that are curently not
   included in the mix, thus N+1 mixes.  These audio streams are then
   encoded again, RTP packetized and sent out.

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

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

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

            Figure 12: 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 12. 10.  The mixer present one SSRC towards the
   peer client, e.g.  MA1 to Peer A, which is the media mix of the other
   particpants.  As each peer receives a different version produced by
   the mixer there are no actual relation between the different RTP
   sessions in the actual media or the transport level information.
   There is however one connection between RTP1-RTP3 in this figure.  It
   has to do with the SSRC space and the identity information.  When A
   receives the MA1 stream which is a combination of BA1 and CA1 streams
   in the other PeerConnections RTP could enable the mixer to include
   CSRC information in the MA1 stream to identify the contributing
   source BA1 and CA1.

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

A.3.1.1.  RTP Session Termination

   There exist an possible implementation choice to have the RTP
   sessions being separated between the different legs in the multi-
   party communication session and only generate RTP media streams in
   each without carrying on RTP/RTCP level any identity information
   about the contributing sources.  This removes both the functionaltiy
   that CSRC can provide and the possibility to use any extensions that
   build on CSRC and the loop detection.  It may appear a simplification
   if SSRC collision would occur between two different end-points as
   they can be avoide to be resolved and instead remapped between the
   independent sessions if at all exposed.  However, SSRC/CSRC remapping
   requiresthat 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
   missreferencing.  Also SSRC collision handling will still be needed
   as it may occur between the mixer and the end-point.

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

A.3.2.  Media Switching

   An RTP Mixer based on media switching avoids the media decoding and
   encoding cycle in the mixer, but not the decryption and re-encryption
   cycle as one rewrites RTP headers.  This both reduces the amount of
   computational resources needed in the mixer and increases the media
   quality per transmitted bit.  This is achieve by letting the mixer
   have a number of SSRCs that represents conceptual or functional
   streams the mixer produces.  These streams are created by selecting
   media from one of the by the mixer received RTP media streams and
   forward the media using the mixers own SSRCs.  The mixer can then
   switch between available sources if that is required 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 must be set to the value of the Mixer's SSRC.
   Secondly, the sequence number must be 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 may 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
   required.

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

      +-A-------------+             +-MIXER--------------------------+
      | +-PeerC1------|             |-PeerC1--------+                |
      | | +-UDP1------|             |-UDP1--------+ |                |
      | | | +-RTP1----|             |-RTP1------+ | |        +-----+ |
      | | | | +-Video-|             |-Video---+ | | |        |     | |
      | | | | |    AV1|------------>|---------+-+-+-+------->|     | |
      | | | | |       |<------------|MV1 <----+-+-+-+-BV1----|     | |
      | | | | |       |<------------|MV2 <----+-+-+-+-EV1----|     | |
      | | | | +-------|             |---------+ | | |        |     | |
      | | | +---------|             |-----------+ | |        |     | |
      | | +-----------|             |-------------+ |        |  S  | |
      | +-------------|             |---------------+        |  W  | |
      +---------------+             |                        |  I  | |
                                    |                        |  T  | |
      +-B-------------+             |                        |  C  | |
      | +-PeerC2------|             |-PeerC2--------+        |  H  | |
      | | +-UDP2------|             |-UDP2--------+ |        |     | |
      | | | +-RTP2----|             |-RTP2------+ | |        |  M  | |
      | | | | +-Video-|             |-Video---+ | | |        |  A  | |
      | | | | |    BV1|------------>|---------+-+-+-+------->|  T  | |
      | | | | |       |<------------|MV3 <----+-+-+-+-AV1----|  R  | |
      | | | | |       |<------------|MV4 <----+-+-+-+-EV1----|  I  | |
      | | | | +-------|             |---------+ | | |        |  X  | |
      | | | +---------|             |-----------+ | |        |     | |
      | | +-----------|             |-------------+ |        |     | |
      | +-------------|             |---------------+        |     | |
      +---------------+             |                        |     | |
                                    :                        :     : :
                                    :                        :     : :
      +-F-------------+             |                        |     | |
      | +-PeerC6------|             |-PeerC6--------+        |     | |
      | | +-UDP6------|             |-UDP6--------+ |        |     | |
      | | | +-RTP6----|             |-RTP6------+ | |        |     | |
      | | | | +-Video-|             |-Video---+ | | |        |     | |
      | | | | |    CV1|------------>|---------+-+-+-+------->|     | |
      | | | | |       |<------------|MV11 <---+-+-+-+-AV1----|     | |
      | | | | |       |<------------|MV12 <---+-+-+-+-EV1----|     | |
      | | | | +-------|             |---------+ | | |        |     | |
      | | | +---------|             |-----------+ | |        |     | |
      | | +-----------|             |-------------+ |        +-----+ |
      | +-------------|             |---------------+                |
      +---------------+             +--------------------------------+

                   Figure 13: 11: Media Switching RTP Mixer

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

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

   Also in this type of mixer one could consider to terminate the RTP
   sessions fully between the different PeerConnection.  The same
   arguments and conisderations 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 recieves media from that end-point.
   This is similar to the media switching Mixer but have some important
   differences in RTP details.

      +-A-------------+             +-MIXER--------------------------+
      | +-PeerC1------|             |-PeerC1--------+                |
      | | +-UDP1------|             |-UDP1--------+ |                |
      | | | +-RTP1----|             |-RTP1------+ | |        +-----+ |
      | | | | +-Video-|             |-Video---+ | | |        |     | |
      | | | | |    AV1|------------>|---------+-+-+-+------->|     | |
      | | | | |       |<------------|BV1 <----+-+-+-+--------|     | |
      | | | | |       |<------------|CV1 <----+-+-+-+--------|     | |
      | | | | |       |<------------|DV1 <----+-+-+-+--------|     | |
      | | | | |       |<------------|EV1 <----+-+-+-+--------|     | |
      | | | | |       |<------------|FV1 <----+-+-+-+--------|     | |
      | | | | +-------|             |---------+ | | |        |     | |
      | | | +---------|             |-----------+ | |        |     | |
      | | +-----------|             |-------------+ |        |  S  | |
      | +-------------|             |---------------+        |  W  | |
      +---------------+             |                        |  I  | |
                                    |                        |  T  | |
      +-B-------------+             |                        |  C  | |
      | +-PeerC2------|             |-PeerC2--------+        |  H  | |
      | | +-UDP2------|             |-UDP2--------+ |        |     | |
      | | | +-RTP2----|             |-RTP2------+ | |        |  M  | |
      | | | | +-Video-|             |-Video---+ | | |        |  A  | |
      | | | | |    BV1|------------>|---------+-+-+-+------->|  T  | |
      | | | | |       |<------------|AV1 <----+-+-+-+--------|  R  | |
      | | | | |       |<------------|CV1 <----+-+-+-+--------|  I  | |
      | | | | |       | :    :    : |: :  : : : : : :  :  : :|  X  | |
      | | | | |       |<------------|FV1 <----+-+-+-+--------|     | |
      | | | | +-------|             |---------+ | | |        |     | |
      | | | +---------|             |-----------+ | |        |     | |
      | | +-----------|             |-------------+ |        |     | |
      | +-------------|             |---------------+        |     | |
      +---------------+             |                        |     | |
                                    :                        :     : :
                                    :                        :     : :
      +-F-------------+             |                        |     | |
      | +-PeerC6------|             |-PeerC6--------+        |     | |
      | | +-UDP6------|             |-UDP6--------+ |        |     | |
      | | | +-RTP6----|             |-RTP6------+ | |        |     | |
      | | | | +-Video-|             |-Video---+ | | |        |     | |
      | | | | |    CV1|------------>|---------+-+-+-+------->|     | |
      | | | | |       |<------------|AV1 <----+-+-+-+--------|     | |
      | | | | |       | :    :    : |: :  : : : : : :  :  : :|     | |
      | | | | |       |<------------|EV1 <----+-+-+-+--------|     | |
      | | | | +-------|             |---------+ | | |        |     | |
      | | | +---------|             |-----------+ | |        |     | |
      | | +-----------|             |-------------+ |        +-----+ |
      | +-------------|             |---------------+                |
      +---------------+             +--------------------------------+
                     Figure 14: 12: Media Projecting Mixer

   So in this six participant conference depicted above in (Figure 14) 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 transmitt 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 consequitvely 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 indepdentently also thus working around any SSRC collisions
   by having remapping tables between the RTP sessions.  However the
   related WebRTC MediaStream signalling must be correspondlingly
   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 comming from an end-point and decide if it can act on it
   locally or needs to translate the request into the RTP session that
   contains the media source.  Both end-points and the mixer will need
   to implement conference related codec control functionalities to
   provide a good experience.  Full Intra Request to request from the
   media source to provide switching points between the sources,
   Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer
   to aggregate congestion control response towards the media source and
   have it adjust its bit-rate in case the limitation is not in the
   source to mixer link.

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

A.4.  Translator Based

   There is also a variety of translators.  The core commonality is that
   they do not need to make themselves visible in the RTP level by
   having an SSRC themselves.  Instead they sit between one or more end-
   point and perform translation at some level.  It can be media
   transcoding, protocol translation or covering missing functionality
   for a legacy device 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 should at all be relevant.  In certain legacy deployment
   media transcoder will be necessary to ensure both codecs and bit-rate
   falls within the envelope the legacy device end-point supports.

   As transcoding requires access to the media the transcoder must
   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 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 should send towards the source
   end-point.  For example receving a 2.5 mbps video stream and then
   send out a 250 kbps video stream after transcoding is a vaste 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 futher than what is needed to represent the
   incoming quality accurate is also wasted resources.

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

                        Figure 15: 13: Media Transcoder

   Figure 15 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 represtented 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 soruced 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 is required is not
   supported by an end-point wants to participate in session.  This RTP
   translator in Figure 16 14 takes on the role of ensuring that from the
   perspective of participant A, participant B appears as a fully
   compliant WebRTC end-point (that is, it is the combination of the
   Translator and participant B that looks like a WebRTC end point).

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

       Figure 16: 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
   keymanagement.  On RTP level the main functions that may be missing
   in a legacy implementation that otherswise support RTP are RTCP in
   general, SRTP implementation, congestion control and feedback
   messages required 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 17: 15: RTP/RTCP Protocol Translator

   The legacy gateway may be implemented in several ways and what it
   need to change is higly dependent on what functions it need to proxy
   for the legacy end-point.  One possibility is depicted in Figure 17 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 may need the capabilities
   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
   required in this type of translator if it operates on the RTP/RTCP
   packet content or media.  In fact one of the more likley 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 integirty protection operation to resolve missmatch 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 flavors, 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 18. 16.

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

         Figure 18: 16: RTP Translator (Relay) with Only Unicast Paths

   The first kind is straight forward and is likely to exist in WebRTC
   context when an legacy end-point is compatible with the exception for
   ICE, and thus needs a gateway that terminates the ICE and then
   forwards all the RTP/RTCP traffic and keymanagment 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
   particpants, and is sometimes called a transport-layer translator.
   In Figure 18, 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 19. 17.

      +-A-------------+             +-RELAY--------------------------+
      | +-PeerC1------|             |-PeerC1--------+                |
      | | +-UDP1------|             |-UDP1--------+ |                |
      | | | +-RTP1----|             |-RTP1-------------------------+ |
      | | | | +-Video-|             |-Video---+                    | |
      | | | | |    AV1|------------>|---------------------------+  | |
      | | | | |       |<------------|BV1 <--------------------+ |  | |
      | | | | |       |<------------|CV1 <------------------+ | |  | |
      | | | | +-------|             |---------+             | | |  | |
      | | | +---------|             |-------------------+   ^ ^ V  | |
      | | +-----------|             |-------------+ |   |   | | |  | |
      | +-------------|             |---------------+   |   | | |  | |
      +---------------+             |                   |   | | |  | |
                                    |                   |   | | |  | |
      +-B-------------+             |                   |   | | |  | |
      | +-PeerC2------|             |-PeerC2--------+   |   | | |  | |
      | | +-UDP2------|             |-UDP2--------+ |   |   | | |  | |
      | | | +-RTP2----|             |-RTP1--------------+   | | |  | |
      | | | | +-Video-|             |-Video---+             | | |  | |
      | | | | |    BV1|------------>|-----------------------+ | |  | |
      | | | | |       |<------------|AV1 <----------------------+  | |
      | | | | |       |<------------|CV1 <--------------------+ |  | |
      | | | | +-------|             |---------+             | | |  | |
      | | | +---------|             |-------------------+   | | |  | |
      | | +-----------|             |-------------+ |   |   V ^ V  | |
      | +-------------|             |---------------+   |   | | |  | |
      +---------------+             |                   |   | | |  | |
                                    :                   |   | | |  | |
                                    :                   |   | | |  | |
      +-C-------------+             |                   |   | | |  | |
      | +-PeerC3------|             |-PeerC3--------+   |   | | |  | |
      | | +-UDP3------|             |-UDP3--------+ |   |   | | |  | |
      | | | +-RTP3----|             |-RTP1--------------+   | | |  | |
      | | | | +-Video-|             |-Video---+             | | |  | |
      | | | | |    CV1|------------>|-------------------------+ |  | |
      | | | | |       |<------------|AV1 <----------------------+  | |
      | | | | |       |<------------|BV1 <------------------+      | |
      | | | | +-------|             |---------+                    | |
      | | | +---------|             |------------------------------+ |
      | | +-----------|             |-------------+ |                |
      | +-------------|             |---------------+                |
      +---------------+             +--------------------------------+

                  Figure 19: 17: Transport Multi-party Relay

   As the Relay RTP and RTCP packets between the UDP flows as indicated
   by the arrows for the media flow a given WebRTC end-point, like A
   will see the remote sources BV1 and CV1.  There will be also two
   different network paths between A, and B or C. This results in that
   the client A must be capable of handlilng that when determining
   congestion state that there might exist multiple destinations on the
   far side of a PeerConnection and that these paths shall 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 incosistent and without explicit
   information about it being blocked must be interpret 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 must be agreed such so
   that the relay doesn't get it, e.g. no DTLS-SRTP handshake with the
   relay, instead some other method must be used.  Secondly, the keying
   structure must 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 20. 18.  The first one could
   be achieved using MIKEY messages in SDP.

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

             Figure 20: 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 21) 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 21: 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 approache will result in that the WebRTC end-points will have
   to have the same capabilities as being discussed in Relay
   (Appendix A.4.3).  Thus A will see an RTP session that is extended
   beyond the PeerConnection and see two different receiving end-points
   with different path characteristics (B and C).  Thus A's congestion
   control needs to be capable of handling this.  The security solution
   can either support mechanism that allows A to inform C about the key
   A is using despite B and C having agreed on another set of keys.
   Alternatively B will decrypt and then re-encrypt using a new key.
   The relay based approach has the advantage that B does not need to
   transcode the media thus both maintaining the quality of the encoding
   and reducing B's complexity requirements.  If the right security
   solutions are supported then also C will be able to verify the
   authenticity of the media comming 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 may 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 must 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 mutliple
   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 9. 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 10. 8.  If a single encoder
   instance is feed to multiple PeerConnections the intention of
   performing simulcast will fail.

   Thus it should 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