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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: Standards Track                           M. Westerlund
Expires: February 29, 2012                                      Ericsson
                                                                  J. Ott
                                                        Aalto University
                                                         August 28, 2011


                      RTP Requirements for RTC-Web
                   draft-perkins-rtcweb-rtp-usage-03

Abstract

   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.

   This document is a candidate to become a work item of the RTCWEB
   working group as <WORKING GROUP DRAFT "MEDIA TRANSPORTS">.

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 February 29, 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



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


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1.  Expected Topologies  . . . . . . . . . . . . . . . . . . .  3
   2.  Requirements from RTP  . . . . . . . . . . . . . . . . . . . .  6
     2.1.  Signalling for RTP sessions  . . . . . . . . . . . . . . .  6
     2.2.  (Lack of) Signalling for Payload Format Changes  . . . . .  7
   3.  RTP Profile  . . . . . . . . . . . . . . . . . . . . . . . . .  7
   4.  RTP and RTCP Guidelines  . . . . . . . . . . . . . . . . . . .  8
   5.  RTP Optimisations  . . . . . . . . . . . . . . . . . . . . . .  8
     5.1.  RTP and RTCP Multiplexing  . . . . . . . . . . . . . . . .  8
     5.2.  Reduced Size RTCP  . . . . . . . . . . . . . . . . . . . .  8
     5.3.  Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . .  9
     5.4.  Generation of the RTCP Canonical Name (CNAME)  . . . . . .  9
   6.  RTP Extensions . . . . . . . . . . . . . . . . . . . . . . . . 10
     6.1.  RTP Conferencing Extensions  . . . . . . . . . . . . . . . 10
       6.1.1.  Full Intra Request . . . . . . . . . . . . . . . . . . 11
       6.1.2.  Picture Loss Indication  . . . . . . . . . . . . . . . 11
       6.1.3.  Slice Loss Indication  . . . . . . . . . . . . . . . . 11
       6.1.4.  Reference Picture Selection Indication . . . . . . . . 11
       6.1.5.  Temporary Maximum Media Stream Bit Rate Request  . . . 11
     6.2.  RTP Header Extensions  . . . . . . . . . . . . . . . . . . 12
     6.3.  Rapid Synchronisation Extensions . . . . . . . . . . . . . 13
     6.4.  Client to Mixer Audio Level  . . . . . . . . . . . . . . . 13
     6.5.  Mixer to Client Audio Level  . . . . . . . . . . . . . . . 13
   7.  Improving RTP Transport Robustness . . . . . . . . . . . . . . 13
     7.1.  RTP Retransmission . . . . . . . . . . . . . . . . . . . . 14
     7.2.  Forward Error Correction (FEC) . . . . . . . . . . . . . . 15
       7.2.1.  Basic Redundancy . . . . . . . . . . . . . . . . . . . 15
       7.2.2.  Block Based  . . . . . . . . . . . . . . . . . . . . . 16
       7.2.3.  Recommendations for FEC  . . . . . . . . . . . . . . . 17
   8.  RTP Rate Control and Media Adaptation  . . . . . . . . . . . . 17
   9.  RTP Performance Monitoring . . . . . . . . . . . . . . . . . . 18
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 18
   11. Security Considerations  . . . . . . . . . . . . . . . . . . . 18
   12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18
   13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
     13.1. Normative References . . . . . . . . . . . . . . . . . . . 19
     13.2. Informative References . . . . . . . . . . . . . . . . . . 21
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 21



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1.  Introduction

   This memo discusses the Real-time Transport Protocol (RTP) [RFC3550]
   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 fit 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 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 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
      benefit.

   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 explained 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.  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.

   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
   established.


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

                Figure 3: RTP Mixer with Only Unicast Paths



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   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

   To support legacy end-point (B) that don't fulfil the requirements 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 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 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.  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.

   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 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



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   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.

2.2.  (Lack of) Signalling for Payload Format Changes

   As discussed in Section 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.  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 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 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.





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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].


5.  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
   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
   share.

   RFC5506 "Support for Reduced-Size Real-Time Transport Control



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   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.

   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.




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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
   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 
   3.3)

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

   3) Point to Multipoint Using a Video Switching MCU (RFC 5117, section 
   3.5)

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

   We note that 3 and 4 are not well utilising 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 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].







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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.

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.



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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 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.

   It is RECOMMENDED that the mechanism to encrypt header extensions
   [I-D.ietf-avtcore-srtp-encrypted-header-ext] is implemented when the
   client-to-mixer and 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.

   Currently the other header extensions are not recommended to be
   included at this time.  But we do include a list of the available
   ones for information 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.

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







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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.  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.

6.5.  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
   it is OPTIONAL that this extension is included.


7.  Improving RTP 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



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   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.  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 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.

   "RTC-Web 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 continent.  In addition
      in some applications one may accept temporary delay spikes to
      allow for retransmission of crucial codec information such an
      parameter sets, intra picture etc, rather than getting no media at
      all.

   The undesirable increase in 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.



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   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 are also highly dynamically and
   depends on two main factors, the amount and pattern of lost packets
   to be recovered and the mechanism one use 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 an
   some time earlier sent packet.  This is relatively simple in theory,
   i.e. one saves any outgoing source (original) packet in a buffer
   marked with a timestamp of actual transmission, some X ms later one
   transmit this packet again.  Where X is selected to be longer than
   the common loss events.  Thus any loss events shorter than X can be
   recovered assuming that one doesn't get an another loss event before
   all the packets lost in the first event has been received.

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



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   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 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 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.

7.2.2.  Block Based

   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



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   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.

7.2.3.  Recommendations for FEC

   (tbd)


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
   problems.

   The biggest issue is that there are no standardised and ready to use
   mechanism that can simply be included in RTC-Web.  Thus there will be



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   need for the IETF to produce such a specification.  A potential
   starting point for defining a solution is "RTP with TCP Friendly Rate
   Control" [rtp-tfrc].


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

   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
   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





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13.  References

13.1.  Normative References

   [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]
              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.

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

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

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

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

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



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              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.

   [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.

   [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.

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

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

   [rtp-tfrc]
              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







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   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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