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Versions: 00 01 02 03 draft-ietf-rtcweb-rtp-usage

Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Intended status: Informational                             M. Westerlund
Expires: December 7, 2011                                       Ericsson
                                                                  J. Ott
                                                        Aalto University
                                                            June 5, 2011

                      RTP Requirements for RTC-Web


   This memo discusses use of RTP in the context of the RTC-Web
   activity.  It discusses important features of RTP that need to be
   considered by other parts of the RTC-Web framework, describes which
   RTP profile to use in this environment, and outlines what RTP
   extensions should 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
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   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 7, 2011.

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
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must

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   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1.  Expected Topologies  . . . . . . . . . . . . . . . . . . .  3
   2.  Requirements from RTP  . . . . . . . . . . . . . . . . . . . .  6
     2.1.  RTP Multiplexing Points  . . . . . . . . . . . . . . . . .  6
     2.2.  Signalling for RTP sessions  . . . . . . . . . . . . . . .  8
     2.3.  (Lack of) Signalling for Payload Format Changes  . . . . .  9
   3.  RTP Profile  . . . . . . . . . . . . . . . . . . . . . . . . . 10
   4.  RTP and RTCP Guidelines  . . . . . . . . . . . . . . . . . . . 10
   5.  RTP Optimizations  . . . . . . . . . . . . . . . . . . . . . . 11
     5.1.  RTP and RTCP Multiplexing  . . . . . . . . . . . . . . . . 11
     5.2.  Reduced Size RTCP  . . . . . . . . . . . . . . . . . . . . 11
     5.3.  Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 11
     5.4.  Generation of the RTCP Canonical Name (CNAME)  . . . . . . 12
   6.  RTP Extensions . . . . . . . . . . . . . . . . . . . . . . . . 12
     6.1.  RTP Conferencing Extensions  . . . . . . . . . . . . . . . 12
       6.1.1.  RTCP Feedback Message: Full Intra Request  . . . . . . 13
       6.1.2.  RTCP Feedback Message: Picture Loss Indicator  . . . . 13
       6.1.3.  RTCP Feedback Message: Temporary Maximum Media
               Stream Bit Rate Request  . . . . . . . . . . . . . . . 14
     6.2.  RTP Header Extensions  . . . . . . . . . . . . . . . . . . 14
     6.3.  Rapid Synchronisation Extensions . . . . . . . . . . . . . 15
   7.  Improving RTP Transport Robustness . . . . . . . . . . . . . . 15
     7.1.  RTP Retransmission . . . . . . . . . . . . . . . . . . . . 15
     7.2.  Forward Error Correction (FEC) . . . . . . . . . . . . . . 15
   8.  RTP Rate Control and Media Adaptation  . . . . . . . . . . . . 16
   9.  RTP Performance Monitoring . . . . . . . . . . . . . . . . . . 16
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 16
   11. Security Considerations  . . . . . . . . . . . . . . . . . . . 16
   12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 17
   13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 17
     13.1. Normative References . . . . . . . . . . . . . . . . . . . 17
     13.2. Informative References . . . . . . . . . . . . . . . . . . 19
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 19

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1.  Introduction

   This memo discusses the Real-time Transport Protocol (RTP) [RFC3551]
   in the context of the RTC-Web activity.  The work in the IETF Audio/
   Video Transport Working Group, and it's successors, has been about
   providing building blocks for real-time multimedia transport, and has
   not specified who should use which building blocks.  The selection of
   building blocks and functionalities can really only be done in the
   context of some application, for example RTC-Web.  We have selected a
   set of RTP features and extensions that are suitable for a number of
   applications that fits the RTC-Web context.  Thus applications such
   as VoIP, audio and video conferencing, and on-demand multimedia
   streaming are considered.  Applications that rely on IP multicast
   have not been considered likely to be applicable to RTC-Web, thus
   extensions related to multicast have been excluded.  We believe that
   RTC-Web will greatly benefit in interoperability if a reasonable set
   of RTP functionalities and extensions are selected.  This memo is
   intended as a starting point for discussion of those features in the
   RTC-Web framework.

   This memo is structured into different topics.  For each topic, one
   or several recommendations from the authors are done.  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 RTC-Web specification:

   REQUIRED:  Functionality that is absolutely needed to make the RTC-
      Web 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

   When this memo discusses RTP, it includes the RTP Control Protocol
   (RTCP) unless explicitly stated otherwise.  RTCP is a fundamental and
   integral part of the RTP protocol, and is REQUIRED to be implemented.

1.1.  Expected Topologies

   As RTC-Web is focused on peer to peer connections established from
   clients in web browsers the following topologies further discussed in
   RTP Topologies [RFC5117] are primarily considered.  The topologies
   are depicted and briefly explaind here for ease of the reader.

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                            +---+         +---+
                            | 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 and is unfortunately not discussed in the RTP Topologies
   [RFC5117].  This topology has the benefit of not requiring central
   nodes.  On the downside is that it increase the used bandwidth by
   requiring one copy of the media streams for each participant 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.

   It needs to be noted that if this topology is to be supported by the
   RTC-Web framework it needs to be possible to connect one RTP session
   to multiple established peer to peer flows that are individually

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

                Figure 3: RTP Mixer with Only Unicast Paths

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   An RTP mixer (Figure 3) is a centralized point that selects or mixes
   content in a conference to optimize 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 optimized in different ways.  These optimizations
   include methods like only chosing 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 transmitts.

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

               Figure 5: Translator towards Legacy end-point

   To support legacy end-point (B) that don't fulfill the requiremetns
   of RTC-Web 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 transparant 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 RTC-Web
   results in such issues and how to handle them.

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2.  Requirements from RTP

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

2.1.  RTP Multiplexing Points

   There are three fundamental points of multiplexing within the RTP

   Use of separate RTP Sessions:  The first, and the most important,
      multiplexing point is the RTP session.  This multiplexing point
      does not have an identifier within the RTP protocol itself, but
      instead relies on the lower layer to separate the different RTP
      sessions.  This is most often done by separating different RTP
      sessions onto different UDP ports, or by sending to different IP
      multicast addresses.  The distinguishing feature of an RTP session
      is that it has a separate SSRC identifier space; a single RTP
      session can span multiple transport connections provided packets
      are gatewayed such that participants are known to each other.
      Different RTP sessions are used to separate different types of
      media within a multimedia session.  For example, audio and video
      flows are sent on separate RTP sessions.

   Multiplexing using the SSRC within an RTP session:  The second
      multiplexing point is the SSRC that separates different sources of
      media within a single RTP session.  An example might be different
      participants in a multiparty teleconference, or different camera
      views of a presentation.  In most cases, each participant within
      an RTP session has a single SSRC, although this may change over
      time if collisions are detected.  However, in some more complex
      scenarios participants may generate multiple media streams of the
      same type simultaneously (e.g., if they have two cameras, and so
      send two video streams at once) and so will have more than one
      SSRC in use at once.  The RTCP CNAME can be used to distinguish
      between a single participant using two SSRC values (where the RTCP
      CNAME will be the same for each SSRC), and two participants (who
      will have different RTCP CNAMEs).

   Multiplexing using the Payload Type within an RTP session:  If
      different media encodings of the same type are to be used at
      different times within an RTP session, for example a single
      participant that can switch between two different audio codecs,
      the payload type is used to identify how the media from that
      particular source is encoded.  When changing media formats within
      an RTP Session, the SSRC of the sender remains unchanged, but the
      RTP Payload Type changes to indicate the change in media format.

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   These multiplexing points area fundamental part of the design of RTP
   and are discussed in Section 5.2 of [RFC3550].  Of special importance
   is the need to separate different RTP sessions using a multiplexing
   mechanism at some lower layer than RTP, rather than trying to combine
   several RTP sessions into one lower layer flow.

   The processing that can happen in an RTP mixer, translator or in an
   end-point is dependent on the purpose and media type of the stream,
   as determined by the RTP session on which it arrives.  Hence, it is
   important to separate such RTP session from each other.  This could
   of course be achieved by other methods, like tagging SSRC values with
   their purpose (this is not defined in any known specification), but
   there are reasons why this method isn't defined.  First of all it is
   not the simple solution, as this require additional signalling, and
   possibly synchronization between session peers.  In addition,
   combining RTP sessions into a single lower-layer flow complicates
   quality of service and traffic engineering between the media flows in
   different RTP sessions.  By using different transport layer ports,
   QoS mechanism that are capable of operating on the 5-tuple (Source
   address, port, destination address, port, and protocol) can be used
   without modification on RTP.

   There are also various other RTP mechanism that become problematic if
   one doesn't have a clear separation of RTP sessions:

   Scalabilty:  RTP was built with media scalability in consideration.
      The simplest way of achieving separation between different
      scalability layers are placing them in different RTP sessions, and
      using the same SSRC and CNAME in each session to bind them
      together.  This is most commonly done in multicast, and not
      particular applicable to RTC-Web, but gatewaying of such a session
      would then require more alterations and likely stateful

   RTP Retransmission in Session Multiplexing mode:  RTP Retransmission
      [RFC4588] does have a mode for session multiplexing.  This would
      not be the main mode used in RTC-Web, but for interoperability and
      reduced cost in translation support for different RTP Sessions are

   Forward Error Correction:  The "An RTP Payload Format for Generic
      Forward Error Correction" [RFC2733] and its update [RFC5109] can
      only be used on media formats that produce RTP packets that are
      smaller than half the MTU if the FEC flow and media flow being
      protected are to be sent in the same RTP session, this is due to
      "RTP Payload for Redundant Audio Data" [RFC2198].  This is because
      the SSRC value of the original flow is recovered from the FEC
      packets SSRC field.  So for anything that desires to use these

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      format with RTP payloads that are close to MTU needs to put the
      FEC data in a separate RTP session compared to the original

   RTCP behavior also becomes a factor in why overloading RTP sessions
   is problematic.  The extension mechanisms used in RTCP depends on the
   media streams.  For example the Extended RTCP report block for VoIP
   is of suitable for conversational audio, but clearly not useful for
   Video.  This has three impacts, either one get unusable reports if
   they are generated for streams where there are little purpose.  This
   is maybe less likely for the VoIP report, but for example the more
   detailed media agnostic reports it may occur.  It otherwise makes the
   implementation of RTCP more complex as the SSRC purpose tagging needs
   not only to be one the media side, but also on the RTCP reporting.
   Also the RTCP reporting interval and transmission scheduling will be

   Due to these design principle implementors of various services or
   applications using RTP have not commonly violated this model.  If one
   choses to violate it today, one fails to achieve interoperability
   with a number of existing services, applications and implementations.

   As a conclusion not ensuring that RTP sessions are used for its
   intended purpose as a multiplexing point does violate the RTP design
   philosophy.  It prevents the use of certain RTP extensions.  It will
   require additional extensions to function and will significantly
   increase the complexity of the implementation.  At the same time it
   will significantly reduce the interoperability with current
   implementations.  Thus the authors considered it REQUIRED that RTP
   sessions are multiplexed using a mechanism outside of RTP.  The
   RECOMMENDED mechanism to accomplish that would be to use unique UDP
   flows.  If the WG comes to a consensus that due to NAT/Firewall
   traversal aspects would be greately simplified with a single flow
   between peers and accept that flow based QoS can only be done on the
   aggreage of all RTP sessions then the authors RECOMMEND that some
   type of multiplexing layer is inserted between UDP flow and the RTP/
   RTCP header to separate the RTP sessions.

2.2.  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

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   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 transoformation 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.

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

   RTP Extensions:  The RTP extensions one intendeds to use needs to be
      agreed on, including any parameters for that extension.  In some
      case just to avoid spending bit-rate on 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 negotiated somehow, and provided to the RTP implementation.

2.3.  (Lack of) Signalling for Payload Format Changes

   As discussed in Section 2.2, 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

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   endpoint, that endpoint is REQUIRED to be prepared to receive data
   encoded in any of those formats at any time.  RTP does not require
   advance signalling for changes between formats that were signalled
   during the session setup.  This is needed for rapid rate adaptation.

3.  RTP Profile

   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 utilize 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 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

4.  RTP and RTCP Guidelines

   RTP and RTCP are two flexible and extensible protocols that allow, on
   the one hand, choosing from a variety of building blocks and
   combining those to meet application needs, and on the other hand,
   create extensions where existing mechanisms are not sufficient: from
   new payload formats to RTP extension headers to additional RTCP
   control packets.

   Different informational documents provide guidelines to the use and
   particularly the extension of RTP and RTCP, including the following:
   Guidelines for Writers of RTP Payload Format Specifications [RFC2736]
   and Guidelines for Extending the RTP Control Protocol [RFC5968].

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5.  RTP Optimizations

   This section discusses some optimizations 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
   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.

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

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

   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

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   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 registred 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.

   Using Symmetric RTP and RTCP [RFC4961] is REQURIED.

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 Synchronization 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 within 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 the RTP CNAME as specified in "Guidelines for Choosing RTP
   Control Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is
   RECOMMENDED, since this addresses both concerns.

6.  RTP Extensions

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

6.1.  RTP Conferencing Extensions

   RTP is inherently defined for group communications, whether using IP
   multicast, multi-unicast, or based on a centralised server.  In
   today's practice, however, overlay-based conferencing dominates,
   typically using one or a few so-called conference bridges or servers
   to connect endpoints in a star or flat tree topology.  Quite diverse

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   conferencing topologies can be created using the basic elements of
   RTP mixers and translators as defined in RFC 3550.

   An number of conferencing topologies are defined in [RFC5117] out of
   the which the following ones are the more common (and most likely in
   practice workable) ones:

   1) RTP Translator (Relay) with Only Unicast Paths (RFC 5117, section 

   2) RTP Mixer with Only Unicast Paths (RFC 5117, section 3.4)

   3) Point to Multipoint Using a Video Switching MCU (RFC 5117, section 

   4) Point to Multipoint Using Content Modifying MCUs (RFC 5117,
   section 3.6)

   We note that 3 and 4 are not well utilizing the functions of RTP and
   in some cases even violates the RTP specifications.  Thus we
   recommend that one focus on 1 and 2.

   RTP protocol extensions to be used with conferencing are included
   because they are important in the context of centralized
   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.  RTCP Feedback Message: 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 the currently
   distributed 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

6.1.2.  RTCP Feedback Message: Picture Loss Indicator

   The Picture Loss Indicator is defined in Section 6.3.1 of AVPF
   [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

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

6.1.3.  RTCP Feedback Message: Temporary Maximum Media Stream Bit Rate

   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.  RTP 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 signaled (e.g., as defined by the payload type).  Valid examples
   might include metadata that is additional to the usual RTP

   The RTP rapid synchronisation header extension is recommended, as
   discussed in Section 6.3.

   Currently no other header extensions are recommended.  But we do
   include a list of the available ones for consideration below:

   Transmission Time offsets:  [RFC5450] defines a format for including
      an RTP timestamp offset of the actual transmission time of the RTP
      packet in relation to capture/display timestamp present in the RTP
      header.  This can be used to improve jitter determination and
      buffer management.

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   Associating Time-Codes with RTP Streams:  [RFC5484] defines how to
      associate SMPTE times codes with the RTP streams.

   Audio Levels indications:  There is ongoing work to define RTP header
      extensions for providing audio levels both from a media sender to
      an mixer [I-D.ietf-avtext-client-to-mixer-audio-level], and from a
      mixer to a receiver[I-D.ietf-avtext-mixer-to-client-audio-level].

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.

7.  Improving RTP Transport Robustness

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

7.1.  RTP 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 prioritize 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.

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.

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   (tbd: add further discussion here)

8.  RTP Rate Control and Media Adaptation

   It is REQUIRED to have an RTP Rate Control mechanism using Media
   adaptation to ensure that the generated RTP flows are network
   friendly, and maintain the user experience in the presence of network

   The biggest issue is that there are no standardized and ready to use
   mechanism that can simply be included in RTC-Web.  Thus there will be
   need for the IETF to produce such a specification.  A potential
   starting point for defining a solution is "RTP with TCP Friendly Rate

9.  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.

10.  IANA Considerations

   This memo makes no request of IANA.

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

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 will provide protection or mitigation
   against all the fundamental issues by offering confidentiality,
   integrity and partial source authentication.  We don't discuss the
   key-management aspect of SRTP in this memo, that needs to be done
   taking the RTC-Web communication model into account.

   In the context of RTC-Web the actual security properties required

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   from RTP are currently not fully understood.  Until security goals
   and requirements are specified it will be difficult to determine what
   security features in addition to SRTP and a suitable key-management,
   if any, that are needed.

12.  Acknowledgements

13.  References

13.1.  Normative References

              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-00 (work in
              progress), February 2011.

              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-00 (work in
              progress), February 2011.

   [RFC2733]  Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
              for Generic Forward Error Correction", RFC 2733,
              December 1999.

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

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

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

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

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

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              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

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

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

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

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

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

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

   [RFC5450]  Singer, D. and H. Desineni, "Transmission Time Offsets in
              RTP Streams", RFC 5450, March 2009.

   [RFC5484]  Singer, D., "Associating Time-Codes with RTP Streams",
              RFC 5484, March 2009.

   [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.

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

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   [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.

13.2.  Informative References

   [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.

              Gharai, L., "RTP with TCP Friendly Rate Control
              (draft-gharai-avtcore-rtp-tfrc-00)", March 2011.

Authors' Addresses

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

   Email: csp@csperkins.org

   Magnus Westerlund
   Farogatan 6
   SE-164 80 Kista

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

   Joerg Ott
   Aalto University
   School of Electrical Engineering
   Espoo  02150

   Email: jorg.ott@aalto.fi

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