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In: MissingRef
Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Standards Track                           M. Westerlund
Expires: May 3, 2012                                            Ericsson
                                                                  J. Ott
                                                        Aalto University
                                                        October 31, 2011


  Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
                     draft-ietf-rtcweb-rtp-usage-01

Abstract

   The Web Real-Time Communication (WebRTC) framework aims to provide
   support for direct interactive rich communication using audio, video,
   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 May 3, 2012.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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



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


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Media Transport in WebRTC  . . . . . . . . . . . . . . . . . .  4
     3.1.  Expected Topologies  . . . . . . . . . . . . . . . . . . .  4
     3.2.  Requirements from RTP  . . . . . . . . . . . . . . . . . .  7
       3.2.1.  Signalling for RTP sessions  . . . . . . . . . . . . .  7
       3.2.2.  (Lack of) Signalling for Payload Format Changes  . . .  8
   4.  WebRTC Use of RTP: Core Protocols  . . . . . . . . . . . . . .  8
     4.1.  RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . .  8
     4.2.  Choice of RTP Profile  . . . . . . . . . . . . . . . . . .  9
     4.3.  Choice of RTP Payload Formats  . . . . . . . . . . . . . . 10
     4.4.  RTP Session Multiplexing . . . . . . . . . . . . . . . . . 10
   5.  WebRTC Use of RTP: Optimisations . . . . . . . . . . . . . . . 10
     5.1.  RTP and RTCP Multiplexing  . . . . . . . . . . . . . . . . 10
     5.2.  Reduced Size RTCP  . . . . . . . . . . . . . . . . . . . . 11
     5.3.  Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 11
     5.4.  Generation of the RTCP Canonical Name (CNAME)  . . . . . . 12
   6.  WebRTC Use of RTP: Extensions  . . . . . . . . . . . . . . . . 12
     6.1.  Conferencing Extensions  . . . . . . . . . . . . . . . . . 12
       6.1.1.  Full Intra Request . . . . . . . . . . . . . . . . . . 13
       6.1.2.  Picture Loss Indication  . . . . . . . . . . . . . . . 13
       6.1.3.  Slice Loss Indication  . . . . . . . . . . . . . . . . 13
       6.1.4.  Reference Picture Selection Indication . . . . . . . . 14
       6.1.5.  Temporary Maximum Media Stream Bit Rate Request  . . . 14
     6.2.  Header Extensions  . . . . . . . . . . . . . . . . . . . . 14
     6.3.  Rapid Synchronisation Extensions . . . . . . . . . . . . . 15
     6.4.  Mixer Audio Level Extensions . . . . . . . . . . . . . . . 15
       6.4.1.  Client to Mixer Audio Level  . . . . . . . . . . . . . 15
       6.4.2.  Mixer to Client Audio Level  . . . . . . . . . . . . . 15
   7.  WebRTC Use of RTP: Improving Transport Robustness  . . . . . . 16
     7.1.  Retransmission . . . . . . . . . . . . . . . . . . . . . . 16
     7.2.  Forward Error Correction (FEC) . . . . . . . . . . . . . . 17
       7.2.1.  Basic Redundancy . . . . . . . . . . . . . . . . . . . 17
       7.2.2.  Block Based FEC  . . . . . . . . . . . . . . . . . . . 19
       7.2.3.  Recommendations for FEC  . . . . . . . . . . . . . . . 20
   8.  WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 20
     8.1.  Rate Control Requirements  . . . . . . . . . . . . . . . . 21
     8.2.  RTCP Limiations  . . . . . . . . . . . . . . . . . . . . . 21
     8.3.  Legacy Interop Limitations . . . . . . . . . . . . . . . . 22



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   9.  WebRTC Use of RTP: Performance Monitoring  . . . . . . . . . . 23
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 23
   11. Security Considerations  . . . . . . . . . . . . . . . . . . . 24
   12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 24
   13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24
     13.1. Normative References . . . . . . . . . . . . . . . . . . . 24
     13.2. Informative References . . . . . . . . . . . . . . . . . . 27
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 28











































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1.  Introduction

   The Real-time Transport Protocol (RTP) [RFC3550] was designed to
   provide a framework for delivery of audio and video teleconferencing
   data and other real-time media applications.  This memo describes how
   RTP is to be used in the context of the Web Real-Time Communication
   (WebRTC) framework, a new activity that aims to provide support for
   direct, interactive, and rich communication using audio, video,
   collaboration, games, etc. between two peers' web-browsers.

   Previous work in the IETF Audio/Video Transport Working Group, and
   it's successors, has been about providing a framework for real-time
   multimedia transport, but has not specified how the pieces of this
   framework should be combined.  This is because the choice of building
   blocks and protocol features can really only be done in the context
   of some application.  This memo proposes a set of RTP features and
   extensions to be implemented by applications that fit within the
   WebRTC application context.  This includes applications such as voice
   over IP (VoIP), video teleconferencing, and on-demand multimedia
   streaming, delivered in the context of the WebRTC browser-based
   infrastructure.


2.  Terminology

   This memo is structured into different topics.  For each topic, one
   or several recommendations from the authors are given.  When it comes
   to the importance of extensions, or the need for implementation
   support, we use three requirement levels to indicate the importance
   of the feature to the WebRTC specification:

   REQUIRED:  Functionality that is absolutely needed to make the WebRTC
      solution work well, or functionality of low complexity that
      provides high value.

   RECOMMENDED:  Should be included as its brings significant benefit,
      but the solution can potentially work without it.

   OPTIONAL:  Something that is useful in some cases, but not always a
      benefit.


3.  Media Transport in WebRTC

3.1.  Expected Topologies

   As WebRTC is focused on peer to peer connections established from
   clients in web browsers the following topologies further discussed in



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   RTP Topologies [RFC5117] are primarily considered.  The topologies
   are depicted and briefly explained here for ease of the reader.

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

                         Figure 1: Point to Point

   The point to point topology (Figure 1) is going to be very common in
   any single user to single user applications.

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

                          Figure 2: Multi-unicast

   For small multiparty sessions it is practical enough to create RTP
   sessions by letting every participant send individual unicast RTP/UDP
   flows to each of the other participants.  This is called multi-
   unicast (Figure 2), and is unfortunately not discussed in the RTP
   Topologies [RFC5117].  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 media streams for each
   participant that are part of the same session beyond the sender
   itself.  Thus this is limited to scenarios with few end-points unless
   the media is very low bandwidth.

   This topology may 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.  The later
   approach simplifies rate adaptation, but reduces the effectiveness of
   RTCP for debugging purposes, by limiting the ability to send third-
   party RTCP reports.









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

                Figure 3: RTP Mixer with Only Unicast Paths

   An RTP mixer (Figure 3) 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 also
   reduces the bit-rate needs 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.  The downside of
   the mixer is that someone is required to provide the actual mixer.

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

         Figure 4: RTP Translator (Relay) with Only Unicast Paths

   If one wants a less complex central node it is possible to use an
   relay (called an Transport Translator) (Figure 4) that takes on the
   role of forwarding the media to the other end-points but doesn't
   perform any media processing.  It simply forwards the media from all
   other to all the other.  Thus one endpoint A will only need to send a
   media once to the relay, but it will still receive 3 RTP streams with
   the media if B, C and D all currently transmits.

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

               Figure 5: Translator towards Legacy end-point




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   To support legacy end-point (B) that don't fulfil the requirements of
   WebRTC it is possible to insert a Translator (Figure 5) that takes on
   the role to ensure that from A's perspective B looks like a fully
   compliant end-point.  Thus it is the combination of the Translator
   and B that looks like the end-point B. The intention is that the
   presence of the translator is transparent to A, however it is not
   certain that is possible.  Thus this case is include so that it can
   be discussed if any mechanism specified to be used for WebRTC results
   in such issues and how to handle them.

3.2.  Requirements from RTP

   This section discusses some requirements RTP and RTCP [RFC3550] place
   on their underlying transport protocol, the signalling channel, etc.

3.2.1.  Signalling for RTP sessions

   RTP is built with the assumption of an external to RTP/RTCP
   signalling channel to configure the RTP sessions and its functions.
   The basic configuration of an RTP session consists of the following
   parameters:

   RTP Profile:  The name of the RTP profile to be used in session.  The
      RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate
      on basic level, as can their secure variants RTP/SAVP [RFC3711]
      and RTP/SAVPF [RFC5124].  The secure variants of the profiles do
      not directly interoperate with the non-secure variants, due to the
      presence of additional header fields in addition to any
      cryptographic transformation of the packet content.

   Transport Information:  Source and destination address(s) and ports
      for RTP and RTCP MUST be signalled for each RTP session.  If RTP
      and RTCP multiplexing [RFC5761] is to be used, such that a single
      port is used for RTP and RTCP flows, this MUST be signalled (see
      Section 5.1).  If several RTP sessions are to be multiplexed onto
      a single transport layer flow, this MUST also be signalled (see
      Section 4.4).

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







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

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

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

3.2.2.  (Lack of) Signalling for Payload Format Changes

   As discussed in Section 3.2.1, the mapping between media type name,
   and its associated RTP payload format, and the RTP payload type
   number to be used for that format must be signalled as part of the
   session setup.  An endpoint may signal support for multiple media
   formats, or multiple configurations of a single format, each using a
   different RTP payload type number.  If multiple formats are signalled
   by an endpoint, that endpoint is REQUIRED to be prepared to receive
   data encoded in any of those formats at any time (this is slightly
   modified if several RTP sessions are multiplexed onto one transport
   layer connection, such that an endpoint must be prepared for a source
   to switch between formats of the same media type at any time; see
   Section 4.4).  RTP does not require advance signalling for changes
   between formats that were signalled during the session setup.  This
   is needed for rapid rate adaptation.


4.  WebRTC Use of RTP: Core Protocols

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



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   control protocol (RTCP).  RTCP is a fundamental and integral part of
   the RTP protocol, and is REQUIRED to be implemented.

   RTP and RTCP are flexible and extensible protocols that allow, on the
   one hand, choosing from a variety of building blocks and combining
   those to meet application needs, but on the other hand, offer the
   ability to create extensions where existing mechanisms are not
   sufficient.  This memo requires a number of RTP and RTCP extensions
   that have been shown to be provide important functionality in the
   WebRTC context be implemented.  It is possible that future extensions
   will be needed: several documents provide guidelines for the use and
   extension of RTP and RTCP, including Guidelines for Writers of RTP
   Payload Format Specifications [RFC2736] and Guidelines for Extending
   the RTP Control Protocol [RFC5968], and should be consulted before
   extending this memo.

4.2.  Choice of 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/AVPF
   feedback profile [RFC4585], and the secure RTP/SAVP profile
   [RFC3711].

   The RTP/AVPF part of RTP/SAVPF is required to get the improved RTCP
   timer model, that allows more flexible transmission of RTCP packets
   in response to events, rather than strictly according to bandwidth.
   This also saves RTCP bandwidth and will commonly only use the full
   amount when there is a lot of events on which to send feedback.  This
   functionality is needed to make use of the RTP conferencing
   extensions discussed in Section 6.1.  The improved RTCP timer model
   defined by RTP/AVPF is backwards compatible with legacy systems that
   implement only the RTP/AVP profile given some constraints on
   parameter configuration such as RTCP banwidth and "trr-int".

   The RTP/SAVP part of RTP/SAVPF is for support for Secure RTP (SRTP)
   [RFC3711].  This provides media encryption, integrity protection,
   replay protection and a limited form of source authentication.  It
   does not contain a specific keying mechanism, so that, and the set of
   security transforms, will be required to be chosen.  It is possible
   that a security mechanism operating on a lower layer than RTP can be
   used instead and that should be evaluated.  However, the reasons for
   the design of SRTP should be taken into consideration in that
   discussion.  A mandatory to implement media security mechanism
   including keying must be required so that confidentialtiy, integrity
   protection and source authentication of the media stream can be



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   provided when desired by the user.

4.3.  Choice of RTP Payload Formats

   (tbd: say something about the choice of RTP Payload Format for
   WebRTC.  If there is a mandatory to implement set of codecs, this
   should reference them.  In any case, it should reference a discussion
   of signalling for the choice of codec, once that discussion reaches
   closure.)

4.4.  RTP Session Multiplexing

   An association amongst a set of participants communicating with RTP
   is known as an RTP session.  A participant may 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 a different transport connection 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 of multimedia sessions in this way, for
   compatibility with legacy systems.

   In today's networks, however, with the prolific use of Network
   Address Translators (NAT) and Firewalls (FW), there is a desire to
   reduce the number of transport layer ports 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].)  WebRTC
   implementations of RTP are REQUIRED to support multiplexing of
   multimedia sessions 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.


5.  WebRTC Use of RTP: Optimisations

   This section discusses some optimisations that makes RTP/RTCP work
   better and more efficient and therefore are considered.

5.1.  RTP and RTCP Multiplexing

   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



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   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] is REQUIRED.
   Supporting this specification is generally a simplification in code,
   since it relaxes the tests in [RFC3550].

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

5.2.  Reduced Size RTCP

   RTCP packets are usually sent as compound RTCP packets; and RFC 3550
   demands that those compound packets always start with an SR or RR
   packet.  However, especially when using frequent feedback messages,
   these general statistics are not needed in every packet and
   unnecessarily increase the mean RTCP packet size and thus limit the
   frequency at which RTCP packets can be sent within the RTCP bandwidth
   share.

   RFC5506 "Support for Reduced-Size Real-Time Transport Control
   Protocol (RTCP): Opportunities and Consequences" [RFC5506] specifies
   how to reduce the mean RTCP message and allow for more frequent
   feedback.  Frequent feedback, in turn, is essential to make real-time
   application quickly aware of changing network conditions and allow
   them to adapt their transmission and encoding behaviour.

   Support for RFC5506 is REQUIRED.

5.3.  Symmetric RTP/RTCP

   RTP entities choose the RTP and RTCP transport addresses, i.e., IP
   addresses and port numbers, to receive packets on and bind their
   respective sockets to those.  When sending RTP packets, however, they
   may use a different IP address or port number for RTP, RTCP, or both;
   e.g., when using a different socket instance for sending and for
   receiving.  Symmetric RTP/RTCP requires that the IP address and port
   number for sending and receiving RTP/RTCP packets are identical.

   The reasons for using symmetric RTP is primarily to avoid issues with
   NAT and Firewalls by ensuring that the flow is actually bi-
   directional and thus kept alive and registered as flow the intended
   recipient actually wants.  In addition it saves resources in the form
   of ports at the end-points, but also in the network as NAT mappings
   or firewall state is not unnecessary bloated.  Also the number of QoS
   state are reduced.



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   Using Symmetric RTP and RTCP [RFC4961] is REQUIRED.

5.4.  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 may change if a collision is
   detected, or when the RTP application is restarted, it's RTCP CNAME
   is meant to stay unchanged, so that RTP endpoints can be uniquely
   identified and associated with their RTP media streams.  For proper
   functionality, RTCP CNAMEs should be unique among the participants of
   an RTP session.

   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME, but these are not sufficient in the presence of NAT
   devices.  In addition, some may find long-term persistent identifiers
   problematic from a privacy viewpoint.  Accordingly, support for
   generating a short-term persistent RTCP CNAMEs following method (b)
   as specified in Section 4.2 of "Guidelines for Choosing RTP Control
   Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is RECOMMENDED,
   since this addresses both concerns.


6.  WebRTC Use of RTP: Extensions

   There are a number of RTP extensions that could be very useful in the
   WebRTC context.  One set is related to conferencing, others are more
   generic in nature.

6.1.  Conferencing Extensions

   RTP is inherently a group communication protocol.  Groups can be
   implemented using a centralised server, multi-unicast, or 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 [RFC5117], of which the following are the most common:

   1.  RTP Translator (Relay) with Only Unicast Paths ([RFC5117],
       section 3.3)

   2.  RTP Mixer with Only Unicast Paths ([RFC5117], section 3.4)

   3.  Point to Multipoint Using a Video Switching MCU ([RFC5117],
       section 3.5)





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   4.  Point to Multipoint Using Content Modifying MCUs ([RFC5117],
       section 3.6)

   As discussed in [RFC5117] section 3.5, 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 [RFC5117]
   section 3.6, the use of a content modifying MCU with RTCP termination
   breaks RTP loop detection and removes the ability for receivers to
   identify active senders.  According only the first two options are
   recommended.

   RTP protocol extensions to be used with conferencing are included
   because they are important in the context of centralised
   conferencing, where one RTP Mixer (Conference Focus) receives a
   participants media streams and distribute them to the other
   participants.  These messages are defined in the Extended RTP Profile
   for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
   AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
   Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
   usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].

6.1.1.  Full Intra Request

   The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of CCM
   [RFC5104].  It is used to have the mixer request from a session
   participants a new Intra picture.  This is used when switching
   between sources to ensure that the receivers can decode the video or
   other predicted media encoding with long prediction chains.  It is
   RECOMMENDED that this feedback message is supported.

6.1.2.  Picture Loss Indication

   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 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.  It is RECOMMENDED that this feedback message is supported as
   a loss tolerance mechanism.

6.1.3.  Slice Loss Indication

   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.





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6.1.4.  Reference Picture Selection Indication

   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, a
   known-as-correct reference picture can be used for future coding.
   The RPSI message allows this to be signalled.  The use of this RTCP
   feedback message is OPTIONAL as a loss tolerance mechanism.

6.1.5.  Temporary Maximum Media Stream Bit Rate Request

   This feedback message is defined in Section 3.5.4 and 4.2.1 in CCM
   [RFC5104].  This message and its notification message is used by a
   media receiver, to inform the sending party that there is a current
   limitation on the amount of bandwidth available to this receiver.
   This can be for various reasons, and can for example be used by an
   RTP mixer to limit the media sender being forwarded by the mixer
   (without doing media transcoding) to fit the bottlenecks existing
   towards the other session participants.  It is RECOMMENDED that this
   feedback message is supported.

6.2.  Header Extensions

   The RTP specification [RFC3550] provides a capability to extend the
   RTP header with in-band data, but the format and semantics of the
   extensions are poorly specified.  Accordingly, if header extensions
   are to be used, it is REQUIRED that they be formatted and signalled
   according to the general mechanism of RTP header extensions defined
   in [RFC5285].

   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.

   The RTP rapid synchronisation header extension [RFC6051] is
   recommended, as discussed in Section 6.3 we also recommend the client
   to mixer audio level [I-D.ietf-avtext-client-to-mixer-audio-level],
   and consider the mixer to client audio level
   [I-D.ietf-avtext-mixer-to-client-audio-level] as optional feature.



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   It is REQUIRED that the mechanism to encrypt header extensions
   [I-D.ietf-avtcore-srtp-encrypted-header-ext] is implemented when the
   client-to-mixer or mixer-to-client audio level indications are in use
   in SRTP encrypted sessions, since the information contained in these
   header extensions may be considered sensitive.

6.3.  Rapid Synchronisation Extensions

   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.

6.4.  Mixer Audio Level Extensions

6.4.1.  Client to Mixer Audio Level

   The Client to Mixer Audio Level
   [I-D.ietf-avtext-client-to-mixer-audio-level] is an RTP header
   extension used by a client to inform a mixer about the level of audio
   activity in the packet the header is attached to.  This enables a
   central node to make mixing or selection decisions without decoding
   or detailed inspection of the payload.  Thus reducing the needed
   complexity in some types of central RTP nodes.

   Assuming that the Client to Mixer Audio Level
   [I-D.ietf-avtext-client-to-mixer-audio-level] is published as a
   finished specification prior to RTCWEB's first RTP specification then
   it is RECOMMENDED that this extension is included.

6.4.2.  Mixer to Client Audio Level

   The Mixer to Client Audio Level header extension
   [I-D.ietf-avtext-mixer-to-client-audio-level] provides the client
   with the audio level of the different sources mixed into a common mix
   from the RTP mixer.  Thus enabling a user interface to indicate the
   relative activity level of a session participant, rather than just
   being included or not based on the CSRC field.  This is a pure
   optimisations of non critical functions and thus optional
   functionality.

   Assuming that the Mixer to Client Audio Level
   [I-D.ietf-avtext-client-to-mixer-audio-level] is published as a
   finished specification prior to RTCWEB's first RTP specification then



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   it is OPTIONAL that this extension is included.


7.  WebRTC Use of RTP: Improving Transport Robustness

   There are some tools that can make RTP flows robust against Packet
   loss and reduce the impact on media quality.  However they all add
   extra bits compared to a non-robust stream.  These extra bits needs
   to be considered and the aggregate bit-rate needs to be rate
   controlled.  Thus improving robustness might require a lower base
   encoding quality but has the potential to give that quality with
   fewer errors in it.

7.1.  Retransmission

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

   The retransmission scheme in RTP allows flexible application of
   retransmissions.  Only selected missing packets can be requested by
   the receiver.  It also allows for the sender to prioritise between
   missing packets based on senders knowledge about their content.
   Compared to TCP, RTP retransmission also allows one 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 for
   RTCWEB.  We here consider these issues and explain why they are in
   fact not a reason to exclude RTP retransmission from the tool box
   available to RTCWEB media sessions.

   The additional latency added by [RFC4588] will exceed the latency
   threshold for interactive voice and video:  RTP Retransmission will
      require at least one round trip time for a retransmission request
      and repair packet to arrive.  Thus the general suitability of
      using retransmissions will depend on the actual network path
      latency between the end-points.  In many of the actual usages the
      latency between two end-points will be low enough for RTP
      retransmission to be effective.  Interactive communication with
      end-to-end delays of 400 ms still provide a fair quality.  Even
      removing half of that in end-point delays allows functional
      retransmission between end-points on the same continent.  In
      addition, some 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.




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   The undesirable increase in packet transmission at the point when
   congestion occurs:  Congestion loss will impact the rate controls
      view of available bit-rate for transmission.  When using
      retransmission one will have to prioritise between performing
      retransmissions and the quality one can achieve with ones
      adaptable codecs.  In many 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 RTCWEB end-point implementations will need to both select when to
   enable RTP retransmissions based on API settings and measurements of
   the actual round trip time.  In addition for each NACK request that a
   media sender receives it will need to make a prioritisation based on
   the importance of the requested media, the probability that the
   packet will reach the receiver in time for being usable, the
   consumption of available bit-rate and the impact of the media quality
   for new encodings.

   To conclude, the issues raised are implementation concerns that an
   implementation needs to take into consideration, they are not
   arguments against including a highly versatile and efficient packet
   loss repair mechanism.

7.2.  Forward Error Correction (FEC)

   Support of some type of FEC to combat the effects of packet loss is
   beneficial, but is heavily application dependent.  However, some FEC
   mechanisms are encumbered.

   The main benefit from FEC is the relatively low additional delay
   needed to protect against packet losses.  The transmission of any
   repair packets should preferably be done with a time delay that is
   just larger than any loss events normally encountered.  That way the
   repair packet isn't also lost in the same event as the source data.

   The amount of repair packets needed varies depending on 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).

7.2.1.  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



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   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.




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7.2.2.  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.






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7.2.3.  Recommendations for FEC

   (tbd)


8.  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 to specific paths.

   The above estalish a clear need based on 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.



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   The biggest issue is that there are no standardised and ready to use
   mechanism that can simply be included in WebRTC Thus there will be
   need for the IETF to produce such a specification.  Therefore the
   suggested way forward is to specify requirements on any solution for
   the media adaptation.  These requirements is for now proposed to 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.

8.1.  Rate Control Requirements

   Note: This section does not yet have WG consensus.

   This section provides a number of requirements on an media
   adaptation/congestion control solution for WebRTC.

   1.  All WebRTC media streams MUST be congestion-controlled.  (The
       same requirement apply to data streams)

   2.  The congestion algorithms used MUST cause WebRTC streams to act
       reasonably fairly with TCP and other congestion-controlled flows,
       such as DCCP and TFRC, and other WebRTC flows.  Note that WebRTC
       involves multiple data flows which "normally" would be separately
       congestion-controlled.

   3.  The congestion control mechanism MUST be possible to realize
       between two indendently implemented WebRTC end-points.

   4.  The congestion control algorithm SHOULD attempt to minimize the
       media-stream end-to-end delays between the participants, by
       controlling bandwidth appropriately.

   5.  The congestion control SHOULD allow for prioritization and
       shifting of banwidth between media flows.  In other words, if one
       flow on the same path as another has to adjust its bit-rate the
       other flow can perform that adjustment instead, or divided
       between the flows.

   Thus it is REQUIRED to have an implementation of an RTP Rate Control
   mechanism fulfilling the above requirements.

8.2.  RTCP Limiations

   Experience with the congestion control algorithms of TCP [RFC5681],
   TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown
   that feedback on packet arrivals needs to be sent roughly once per
   round trip time.  We note that the capabilities of real-time media
   traffic to adapt to changing path conditions may be less rapid than



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   for the elastic applications TCP was designed for, but frequent
   feedback is still required to allow the congestion control algorithm
   to track the path dynamics.

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

   Interactive communication may not be able to afford waiting for
   packet losses to occur to indicate congestion, because an increase in
   playout delay due to queuing (most prominent in wireless networks)
   may easily lead to packets being dropped due to late arrival at the
   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 report size again.  For
   example, different RTCP XR report blocks (jointly) provide the
   necessary details to implement a variety of congestion control
   algorithms, but the (compound) report size grows quickly.

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

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

8.3.  Legacy Interop Limitations

   Congestion control interoperability with most type 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 of 5s:  RTP [RFC3550] under the RTP/AVP
      profile specifies a recommended minimal fixed interval of 5
      seconds.  Sending RTCP report blocks as seldom as 5 seconds makes
      it very difficult for a sender to use these reports and react to
      any congestion event.







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   RTP/AVP Scaled Minimal Interval:  If a legacy device uses the scaled
      minimal RTCP compound interval, the "RECOMMENDED value for the
      reduced minimum in seconds is 360 divided by the session bandwidth
      in kilobits/second" ([RFC3550], section 6.2).  The minimal
      interval drops below a second, still several times the RTT in
      almost all paths in the Internet, when the session bandwidht
      becomes 360 kbps.  A session bandwidth of 1 Mbps still has a
      minimal interval of 360 ms.  Thus, with the exception for rather
      high bandwidth sessions, getting frequent enough RTCP Report
      Blocks to report on the order of the RTT is very difficult as long
      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 for
      frequent reporting, such as the "trr-int" parameter, and the
      possibility that the end-point supports some useful feedback
      format for congestion control purpose such as TMMBR [RFC5104].

   It has been suggested on the RTCWEB mailing list that if
   interoperating with really 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 the Internet when communicating
   with a legacy node not providing sufficient feedback for effective
   congestion control.  This warrants further discussion 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.


9.  WebRTC Use of RTP: Performance Monitoring

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

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


10.  IANA Considerations

   This memo makes no request of IANA.

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



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   RFC.


11.  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 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.

   The guidelines in [I-D.ietf-avtcore-srtp-vbr-audio] 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].


12.  Acknowledgements

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


13.  References

13.1.  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-00 (work in
              progress), June 2011.

   [I-D.ietf-avtcore-srtp-vbr-audio]



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              Perkins, C. and J. Valin, "Guidelines for the use of
              Variable Bit Rate Audio with Secure RTP",
              draft-ietf-avtcore-srtp-vbr-audio-03 (work in progress),
              July 2011.

   [I-D.ietf-avtext-client-to-mixer-audio-level]
              Lennox, J., Ivov, E., and E. Marocco, "A Real-Time
              Transport Protocol (RTP) Header Extension for Client-to-
              Mixer Audio Level Indication",
              draft-ietf-avtext-client-to-mixer-audio-level-03 (work in
              progress), July 2011.

   [I-D.ietf-avtext-mixer-to-client-audio-level]
              Ivov, E., Marocco, E., and J. Lennox, "A Real-Time
              Transport Protocol (RTP) Header Extension for Mixer-to-
              Client Audio Level Indication",
              draft-ietf-avtext-mixer-to-client-audio-level-03 (work in
              progress), July 2011.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for RTC-Web",
              draft-ietf-rtcweb-security-01 (work in progress),
              October 2011.

   [I-D.lennox-rtcweb-rtp-media-type-mux]
              Lennox, J. and J. Rosenberg, "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.westerlund-avtcore-multiplex-architecture]
              Westerlund, M., Burman, B., and C. Perkins, "RTP
              Multiplexing Architecture",
              draft-westerlund-avtcore-multiplex-architecture-00 (work
              in progress), October 2011.

   [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.




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Internet-Draft               RTP for WebRTC                 October 2011


   [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", RFC 5109, December 2007.

   [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.

   [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.




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13.2.  Informative References

   [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-01 (work in
              progress), June 2011.

   [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-fecframe-framework]
              Watson, M., Begen, A., and V. Roca, "Forward Error
              Correction (FEC) Framework",
              draft-ietf-fecframe-framework-15 (work in progress),
              June 2011.

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              September 1997.

   [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.

   [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.



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Internet-Draft               RTP for WebRTC                 October 2011


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


Authors' Addresses

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

   Email: csp@csperkins.org


   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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


   Joerg Ott
   Aalto University
   School of Electrical Engineering
   Espoo  02150
   Finland

   Email: jorg.ott@aalto.fi









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