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Versions: 00 01 02 03 draft-ietf-avt-rapid-rtp-sync

Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Updates: RFC3550                                              T. Schierl
(if approved)                                             Fraunhofer HHI
Intended status: Standards Track                           March 9, 2009
Expires: September 10, 2009


                   Rapid Synchronisation of RTP Flows
                draft-perkins-avt-rapid-rtp-sync-03.txt

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   Copyright (c) 2009 IETF Trust and the persons identified as the
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Abstract

   This memo outlines how RTP multimedia sessions are synchronised, and



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   discusses how rapidly such synchronisation can occur.  We show that
   most RTP sessions can be synchronised immediately, but that the use
   of video switching multipoint conference units (MCUs) or large source
   specific multicast (SSM) groups can greatly increase the initial
   synchronisation delay.  This increase in delay can be unacceptable to
   some applications that use layered and/or multi-description codecs.

   This memo updates the RTP Control Protocol (RTCP) timing rules to
   reduce the initial synchronisation delay for SSM sessions.  A new
   feedback packet is defined for use with the Extended RTP Profile for
   RTCP-based Feedback (RTP/AVPF), allowing video switching MCUs to
   rapidly request resynchronisation.  Two new RTP header extensions are
   defined to allow rapid synchronisation of late joiners, and guarantee
   correct timestamp based decoding order recovery for layered codecs in
   the presence of clock skew.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Synchronisation of RTP Flows . . . . . . . . . . . . . . . . .  3
     2.1.  Initial Synchronisation Delay  . . . . . . . . . . . . . .  4
       2.1.1.  Unicast Sessions . . . . . . . . . . . . . . . . . . .  5
       2.1.2.  Source Specific Multicast (SSM) Sessions . . . . . . .  5
       2.1.3.  Any Source Multicast (ASM) Sessions  . . . . . . . . .  6
       2.1.4.  Discussion . . . . . . . . . . . . . . . . . . . . . .  6
     2.2.  Synchronisation for Late Joiners . . . . . . . . . . . . .  7
   3.  Reducing RTP Synchronisation Delays  . . . . . . . . . . . . .  8
     3.1.  Rapid Resynchronisation Request  . . . . . . . . . . . . .  8
     3.2.  In-band Delivery of Synchronisation Metadata . . . . . . .  9
     3.3.  Signalling . . . . . . . . . . . . . . . . . . . . . . . . 10
   4.  Application to Decoding Order Recovery in Layered Codecs . . . 11
     4.1.  Problem description  . . . . . . . . . . . . . . . . . . . 11
     4.2.  Use of RTP Header Extensions for Synchronisation . . . . . 11
     4.3.  Timestamp based decoding order recovery  . . . . . . . . . 12
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 15
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 16
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 16
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 16
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 16
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17









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1.  Introduction

   When using RTP to deliver multimedia content it's often necessary to
   synchronise playout of audio and video components of a presentation.
   This is achieved using information contained in RTP Control Protocol
   (RTCP) Sender Report (SR) packets [1].  These are sent periodically,
   and the components of a multimedia session cannot be synchronised
   until sufficient RTCP SR packets have been received for each flow to
   allow the receiver to establish mappings between the media clock used
   for each flow, and the common (NTP-format) clock used to establish
   synchronisation.

   Recently, concern has been expressed that this synchronisation delay
   is problematic for some applications, for example those using layered
   or multi-description video coding.  This memo reviews the operations
   of RTP synchronisation, and describes the synchronisation delay that
   can be expected.  Two backwards compatible extensions to the basic
   RTP synchronisation mechanism are proposed:

   o  An enhancement to the Extended RTP Profile for RTCP-based Feedback
      (RTP/AVPF) [2] is defined to allow receivers to request additional
      RTCP SR packets, providing the metadata needed to synchronise RTP
      flows.  This can reduce the synchronisation delay when joining
      sessions with large RTCP reporting intervals, or in the presence
      of packet loss.

   o  Two RTP header extensions are defined, to deliver synchronisation
      metadata in-band with RTP data packets.  These extensions provide
      synchronisation metadata that is aligned with RTP data packets,
      and so eliminate the need to estimate clock-skew between flows
      before synchronisation.  They can also reduce the need to receive
      RTCP SR packets before synchronising flows.

   The immediate use-case for these extensions is to reduce the delay
   due to synchronisation when joining a layered video session (e.g. an
   H.264/SVC session in NI-T mode [7]).  The extensions are not specific
   to layered coding, however, and can be used in any environment when
   synchronisation latency is an issue.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [3].


2.  Synchronisation of RTP Flows

   RTP flows are synchronised by receivers based on information that is
   contained in RTCP SR packets generated by senders (specifically, the



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   NTP and RTP timestamps).  Each type of media (e.g. audio or video) is
   sent in a separate RTP session, and the receiver associates RTP flows
   to be synchronised by means of the canonical end-point identifier
   (CNAME) item included in the RTCP Source Description (SDES) packets
   generated by the sender.  To ensure synchronisation, an RTP sender
   MUST therefore send periodic compound RTCP packets following Section
   6 of RFC 3550 [1].

   The timing of these periodic compound RTCP packets will depend on the
   number of members in each RTP session, the fraction of those that are
   sending data, the session bandwidth, the configured RTCP bandwidth
   fraction, and whether the session is multicast or unicast (see RFC
   3550 Section 6.2 for details).  In summary, RTCP control traffic is
   allocated a small fraction, generally 5%, of the session bandwidth,
   and of that fraction, one quarter is allocated to active RTP senders,
   while receivers use the remaining three quarters (these fractions can
   be configured via SDP [8]).  Each member of an RTP session derives an
   RTCP reporting interval based on these fractions, whether the session
   is multicast or unicast, the number of members it has observed, and
   whether it is actively sending data or not.  It then sends a compound
   RTCP packet on average once per reporting interval (the actual packet
   transmission time is randomised in the range [0.5 ... 1.5] times the
   reporting interval to avoid synchronisation of reports).

   A minimum reporting interval of 5 seconds is RECOMMENDED, except that
   the delay before sending the initial report "MAY be set to half the
   minimum interval to allow quicker notification that the new
   participant is present" [1].  Also, for unicast sessions, "the delay
   before sending the initial compound RTCP packet MAY be zero" [1].  In
   addition, for unicast sessions, and for active senders in a multicast
   session, the fixed minimum reporting interval MAY be scaled to "360
   divided by the session bandwidth in kilobits/second.  This minimum is
   smaller than 5 seconds for bandwidths greater than 72 kb/s." [1]

2.1.  Initial Synchronisation Delay

   A multimedia session comprises a set of concurrent RTP sessions among
   a common group of participants, using one RTP session for each media
   type.  For example, a videoconference (which is a multimedia session)
   might contain an audio RTP session and a video RTP session.  To allow
   a receiver to synchronise the components of a multimedia session, a
   compound RTCP packet containing an RTCP SR packet and an RTCP SDES
   packet with a CNAME item MUST be sent to each of the RTP sessions in
   the multimedia session.  A receiver cannot synchronise playout across
   the multimedia session until such RTCP packets have been received on
   all of the component RTP sessions.  If there is no packet loss, this
   gives an expected initial synchronisation delay equal to the average
   time taken to receive the first RTCP packet in the RTP session with



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   the longest RTCP reporting interval.  This will vary between unicast
   and multicast RTP sessions.

2.1.1.  Unicast Sessions

   For unicast multimedia sessions, senders SHOULD transmit an initial
   compound RTCP packet (containing an RTCP SR packet and an RTCP SDES
   packet with a CNAME item) immediately on joining each RTP session in
   the multimedia session.  The individual RTP sessions are considered
   to be joined once any in-band signalling for NAT traversal (e.g. [9])
   and/or security keying (e.g. [10],[11]) has concluded, and the media
   path is open.  This implies that the initial RTCP packet is sent in
   parallel with the first data packet following the guidance in RFC
   3550 that "the delay before sending the initial compound RTCP packet
   MAY be zero" and, in the absence of any packet loss, flows can be
   synchronised immediately.

   Note that NAT pinholes, firewall holes, quality-of-service, and media
   security keys should have been negotiated as part of the signalling,
   whether in-band or out-of-band, before the first RTCP packet is sent.
   This should ensure that any middleboxes are ready to accept traffic,
   and reduce the likelihood that the initial RTCP packet will be lost.

2.1.2.  Source Specific Multicast (SSM) Sessions

   For multicast sessions, the delay before sending the initial RTCP
   packet, and hence the synchronisation delay, varies with the session
   bandwidth and the number of members in the session.  For a multicast
   multimedia session, the average synchronisation delay will depend on
   the slowest of the component RTP sessions; this will generally be the
   session with the lowest bandwidth (assuming all the RTP sessions have
   the same number of members).

   When sending to a multicast group, the reduced minimum RTCP reporting
   interval of 360 seconds divided by the session bandwidth in kilobits
   per second [1] should be used when synchronisation latency is likely
   to be an issue.  Also, as usual, the reporting interval is halved for
   the first RTCP packet.  Depending on the session bandwidth and the
   number of members, this gives the following average synchronisation
   delays:











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        Session| Number of receivers (single sender assumed):
      Bandwidth|  2     3     4     5     10   100   1000  10000
             --+------------------------------------------------
         8 kbps| 2.73  4.10  5.47  5.47  5.47  5.47  5.47  5.47
        16 kbps| 2.50  2.50  2.73  2.73  2.73  2.73  2.73  2.73
        32 kbps| 2.50  2.50  2.50  2.50  2.50  2.50  2.50  2.50
        64 kbps| 2.50  2.50  2.50  2.50  2.50  2.50  2.50  2.50
       128 kbps| 1.41  1.41  1.41  1.41  1.41  1.41  1.41  1.41
       256 kbps| 0.70  0.07  0.07  0.07  0.07  0.07  0.07  0.07
       512 kbps| 0.35  0.35  0.35  0.35  0.35  0.35  0.35  0.35
         1 Mbps| 0.18  0.18  0.18  0.18  0.18  0.18  0.18  0.18
         2 Mbps| 0.09  0.09  0.09  0.09  0.09  0.09  0.09  0.09
         4 Mbps| 0.04  0.04  0.04  0.04  0.04  0.04  0.04  0.04

   Figure 1: Average RTCP Reporting Interval (seconds)

   These numbers assume a single-source multicast channel with a single
   active sender, which the rules in RFC 3550 section 6.3 give a fixed
   fraction of the RTCP bandwidth irrespective of the number of
   receivers.  It can be seen that they are sufficient for lip-
   synchronisation without excessive delay, but might be viewed as
   having too much latency for synchronising parts of a layered video
   stream.

   The RTCP interval is randomised in the usual manner, so the minimum
   synchronisation delay will be half these intervals, and the maximum
   delay will be 1.5 times these intervals.  Note also that these RTCP
   intervals are calculated assuming perfect knowledge of the number of
   members in the session.  In practice, an implementation will have
   only limited knowledge of the size of the session when joining, and
   will likely send its initial report early compared to these values,
   following the RTCP reconsideration rules.

2.1.3.  Any Source Multicast (ASM) Sessions

   (tbd)

   For ASM sessions, the fraction of members that are senders plays an
   important role, and imply more variation in average RTCP reporting
   interval.

2.1.4.  Discussion

   For unicast sessions, the existing RTCP SR-based mechanism allows for
   immediate synchronisation, provided the initial RTCP packet is not
   lost.

   For SSM sessions, the initial synchronisation delay is sufficient for



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   lip-synchronisation, but may be larger than desired for some layered
   codecs.  The rationale for not sending immediate RTCP packets for
   multicast groups is to avoid implosion of requests when large numbers
   of members simultaneously join the group ("flash crowd").  This is
   not an issue for SSM senders, since there can be at most one sender,
   so it might be desirable to allow SSM senders to send an immediate
   RTCP SR on joining a session (as is currently allowed for unicast
   sessions, which also don't suffer from the implosion problem).  SSM
   receivers using unicast feedback would not be allowed to send
   immediate RTCP.  This would be a change to RFC 3550, if accepted.

   For ASM session... (tbd)

   In all cases, it is possible that the initial RTCP SR packet is lost.
   In this case, the receiver will not be able to synchronise the media
   until the reporting interval has passed, and the next RTCP SR packet
   is sent.  This is undesirable.  Section 3.1 defines a new RTP/AVPF
   transport layer feedback message to request an RTCP SR be generated,
   allowing rapid resynchronisation in the case of packet loss.

2.2.  Synchronisation for Late Joiners

   Synchronisation between RTP sessions is potentially slower for late
   joiners, than for participants present at the start of the session.
   The reasons for this are two-fold:

   1.  Many of the optimisations that allow rapid transmission of RTCP
       SR packets apply only at the start of a session.  This implies
       that a new participant may have to wait a complete RTCP reporting
       interval for each session before receiving the necessary data to
       synchronise media streams.  This might potentially take several
       seconds, depending on the configured session bandwidth and the
       number of participants.

   2.  Additional synchronisation delay comes from the nature of the
       RTCP timing rules.  Packets are generated on average once per
       reporting interval but with the exact transmission times being
       randomised +/- 50% to avoid synchronisation of reports.  This is
       important to avoid network congestion in multicast sessions, but
       does mean that the timing of RTCP SR reports for different RTP
       sessions aren't synchronised.  Accordingly, a receiver must
       estimate the skew on the NTP-format clock in order to align RTP
       timestamps across sessions.  This estimation is an essential part
       of an RTP synchronisation implementation, and can be done exactly
       given sufficient reports.  Collecting sufficient RTCP SR data to
       perform this estimation, however, may require several reports,
       further increasing the synchronisation delay.




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   These delays are likely an issue for tuning in to an ongoing
   multicast RTP session, or for video switching MCUs.


3.  Reducing RTP Synchronisation Delays

   Two backwards compatible RTP extensions are defined to reduce the
   possible synchronisation delay: a rapid resynchronisation request
   message, and RTP header extensions that can convey synchronisation
   metadata in-band.

3.1.  Rapid Resynchronisation Request

   The general format of an RTP/AVPF transport layer feedback message is
   shown below.


       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |V=2|P|   FMT   | PT=RTPFB=205  |          length               |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                  SSRC of packet sender                        |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                  SSRC of media source                         |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      :            Feedback Control Information (FCI)                 :
      :                                                               :

   A new feedback message type, RTCP-SR-REQ, is defined with FMT = XXX.
   (the next available FMT is 5?)  This MAY be sent to indicate that a
   receiver is unable to synchronise media streams, and desires that the
   media source send an RTCP SR packet as soon as possible (within the
   constraints of the RTCP early feedback rules).  On receipt of this,
   the media source SHOULD generate an RTCP SR packet as soon as
   possible within the RTCP early feedback rules.  That RTCP SR packet
   MAY be sent as a non-compound RTCP packet, if this has been
   negotiated.

   The Feedback Control Information (FCI) part of the packet is emtpy.
   The SSRC of packet sender indicates the member that is unable to
   synchronise media streams, while the SSRC of media source indicates
   the sender of the media it is unable to synchronise.  The lenght MUST
   equal 2.

   (tbd: discuss what happens if the feedback target is not co-located
   with the sender)




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3.2.  In-band Delivery of Synchronisation Metadata

   The RTP header extension mechanism defined in [4] can be adopted to
   carry an OPTIONAL NTP format wall clock timestamp in RTP data
   packets.  If such a timestamp is included, it MUST correspond to the
   same time instant as the RTP timestamp in the packet's header, and
   MUST be derived from the same clock used to generate the NTP format
   timestamps included in RTCP SR packets.  The receiver can use the
   information provided as input to the synchronisation algorithm, as-if
   an RTCP SR packet had been received for the flow.

   Two variants are defined for this header extension.  The first
   variant extends the RTP header with a 64 bit NTP timestamp format
   timestamp as defined in [5].  The second variant carries the lower 24
   bit part of the Seconds of a NTP timestamp format timestamp and the
   32 bit of the Fraction of a NTP timestamp format timestamp.  The
   formats of the two variants are shown below.


       Variant A (16 byte) of the NTP header extension:

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |V=2|P|1|  CC   |M|     PT      |       sequence number         |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+R
      |                           timestamp                           |T
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+P
      |           synchronization source (SSRC) identifier            |
      +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
      |       0xBE    |    0xDE       |           length=3            |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E
      |  ID-A | L=7   |   NTP timestamp format - Seconds (bit 0-23)   |x
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t
      |NTP Sec.(24-31)|   NTP timestamp format - Fraction(bit 0-23)   |n
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |NTP Frc.(24-31)|    0 (pad)    |    0 (pad)    |    0 (pad)    |
      +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
      |                         payload data                          |
      |                             ....                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+










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       Variant B (12 byte) of the NTP header extension:

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |V=2|P|1|  CC   |M|     PT      |       sequence number         |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+R
      |                           timestamp                           |T
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+P
      |           synchronization source (SSRC) identifier            |
      +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
      |       0xBE    |    0xDE       |           length=2            |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E
      |  ID-B | L=6   |  NTP timestamp format - Seconds (bit 8-31)    |x
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t
      |           NTP timestamp format - Fraction (bit 0-31)          |n
      +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
      |                         payload data                          |
      |                             ....                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   An NTP timestamp format timestamp MAY be included on any RTP packets
   the sender chooses, but it is RECOMMENDED when performing timestamp
   based decoding order recovery for layered codecs transported in
   multiple RTP flows, as discussed in Section 4.  This header extension
   MAY be also sent on the RTP packets corresponding to a video random
   access point, and on the associated audio packets, to allow rapid
   synchronisation for late joiners and in video switching scenarios.
   Note that the inclusion of an RTP header extension will reduce the
   efficiency of RTP header compression, if it is used.

   In all cases, irrespective of whether in-band NTP timestamp format
   timestamps are included or not, regular RTCP SR packets MUST be sent
   to provide backwards compatibility with receivers that synchronize
   RTP flows according to [1].  The sender reports are also required to
   receive the upper 8 bit of the Seconds of the NTP timestamp format
   timestamp not included in the NTP header extension.

3.3.  Signalling

   The signaling of using the NTP header extension defined in
   Section 3.2 MUST be applied as defined in [4].

   (tbd - URI, ID-A and ID-B for the NTP header extension need to be
   defined, e.g.  URI: "a=extmap:ID-A
   urn:ietf:params:rtp-hdrext:ntp-64"] "a=extmap:ID-B
   urn:ietf:params:rtp-hdrext:ntp-56"]




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4.  Application to Decoding Order Recovery in Layered Codecs

   Based on the timestamp contained each RTP data packet, and the
   mapping to an NTP-format wall-clock time, a decoding order recovery
   process is applied if a media as result of a layered coding process
   is transported in multiple RTP flows.  This recovers the decoding
   order of media frames or samples at the receiver.  Especially when
   transporting layered video, the decoding order recovery process is
   not straight forward.  In this section, we provide guidance on how to
   use RTP/NTP timing information for decoding order recovery.

4.1.  Problem description

   One option for decoding order recovery in layered codecs is to use
   the NTP (sample presentation) timestamps to reorder data of the same
   layered media transported in different RTP flows.  For a timestamp-
   based decoding order recovery process, it is crucial to allow exact
   alignment of media frames respectively samples using the NTP timing
   information.  In the presence of clock skew, it may not be possible
   to derive exact matching NTP timestamps using the NTP wallclock in
   each RTP flow's RTCP sender reports.  This is due to the fact that
   RTCP sender reports are not sent at the same point of time in the
   multiple RTP flows transporting data of the same layered media.  If
   the RTCP SR packets are not send at the same time, they therefore do
   not contain the same NTP wallclock timestamp.  If there is a skew
   present in the clock used for NTP wallclock timestamp generation,
   using different wallclock timestamps for the same sampling instance
   in the RTP flow inevitably leads to non-matching NTP timestamps
   generated from RTP timestamps and wallclock timestamp in the multiple
   RTP flows.  In order to allow a common and straight forward
   timestamp-based decoding order recovery process, it is important to
   guarantee exact matching of NTP timestamps.  Thus in the presence of
   non-perfect clocks, which should be the normal case, an additional
   mechanism shall be used.  An exact inter-flow alignment of NTP
   timestamps can be guaranteed, if the NTP header extension is always
   inserted at the same timing position in all the RTP flows in question
   and if those NTP header extensions are used to update the NTP-RTP
   relation in all RTP flows at the same point of time.

4.2.  Use of RTP Header Extensions for Synchronisation

   The NTP header extension SHOULD be used with a layered, multi-
   description, or multi-view codec, to provide exact matching of NTP
   timestamps between layers, descriptions, or views transported in
   different RTP flows to allow timestamp-based decoding order recovery.
   If the NTP header extension is inserted for RTP flows transporting
   samples or parts of samples of the same layered media, the NTP header
   extension SHALL be included at least once in each of the RTP flows of



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   the same media for the sampling time instance of an insertion of a
   NTP header extension and such synchronously inserted NTP header
   extensions SHALL contain the same NTP timestamp.  The frequency of
   inserting NTP header extensions in the RTP flows is up to the sender.

   Note: If the decoding order of RTP flows is given by any means (as
   e.g., by mechanism defined in [6]), the NTP timestamp provided by the
   header extension allows to collect data of the same sample from the
   RTP flows, forming the sample decoding order.

   It is RECOMMENDED that the receiver uses for timestamp-based decoding
   order recovery the NTP timestamps provided in the RTP NTP header
   extensions only, if such extensions are present for the RTP flows.
   Section 4 gives further details about the timestamp-based decoding
   order recovery.

   Note: Using the RTP header extensions described above allows the
   receiver to find the corresponding sample of the layered media, or
   parts thereof, in all RTP flows at the instant the RTP header
   extension is inserted into the flows.  This guarantees that any clock
   skew present in the NTP timestamp generation process based on RTCP
   sender reports is avoided, and so allows direct comparison of NTP
   timestamps across multiple RTP flows.  Furthermore, this approach
   solves the possible problem of clock skews identified for the NI-T
   mode as defined in [7].  To ensure the absence of clock skew, a
   header extension containing the NTP timestamp MUST be inserted into
   the RTP flows comprising a layered media stream at the same instant
   in each RTP flow.  This may require the insertion of extra packets in
   some of the RTP flows, since in layered video codecs not all sampling
   instances may be present in all the flows.  If such a header
   extension is included in all flows at a sampling time instance, the
   NTP timestamps for samples following in decoding order the NTP header
   insertion point can be constructed using the RTP timestamps and
   identical reference NTP timestamps in the NTP header extension in all
   RTP flows.  It should be noted that the frequency of inserting the
   NTP header extension is crucial in presence of clock skew, since the
   points of insertion may be the only points for a receiver to start
   the decoding order recovery.

4.3.  Timestamp based decoding order recovery

   If parts or complete samples as result of a layered coding process
   are transported as different RTP flows as different RTP streams
   and/or as different RTP sessions, a decoding order recovery process
   is required to reorder the samples or parts of samples received.
   Such mechanism may be based on the NTP presentation timestamp which
   can be derived from the RTP timestamp using the NTP wallclock
   provided in the RTCP sender report packets.  In order to guarantee



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   the exact NTP alignment, the RTP NTP header extension as defined in
   this memo in Section 3.2 allows the receiver to tune in before the
   reception of such a sender report if the header extension is earlier
   provided in the RTP flow or it may be the only way to allow correct
   decoding order recovery based on exact matching of NTP timestamps in
   case of the presence of clock skew in the clock used for generating
   the NTP wallclock.

   Since typically for layered video codecs as, e.g.  SVC [7], the
   decoding order is not equal to the presentation order of the media
   samples, media samples or parts of media samples cannot be simply
   ordered according to the presentation timestamp order.  For this
   reason, if transporting media samples or parts of media samples of a
   layered, multi-view or multi description codec in different RTP
   flows, the following rules SHOULD be kept for sending such flows:

   Note: The following rules are typically kept for layered audio
   codecs, which allows using the same algorithm for decoding order
   recovery of audio samples.

   Terminology: Following the decoding order of RTP flows as described
   above, an RTP flow containing sample data which is required to be
   accessed and/or decoded before decoding a second sample data of
   another RTP flow is called a lower RTP flow with respect to the
   second RTP flow.  A second RTP flow, which requires for the decoding
   process accessing and/or decoding the sample data of the lower RTP
   flow is called the higher RTP flow.  The lowest RTP flow is the flow,
   which does not require the presence of any other data.

   o  The decoding order of media samples or part of the media samples
      transported in different RTP flows SHOULD be derivable by any
      means.  This can be accomplished, e.g. by using the mechanisms
      defined in [6] if the sample data or parts of the sample data are
      transported in different RTP sessions or by any other means.

   o  For each two RTP flows the following rules SHOULD be true in order
      to allow decoding order recovery based on matching NTP timestamps
      present in the different RTP flows:

      1.  The order of the RTP samples within an RTP flow is equal to
          the decoding order.

      2.  A higher RTP flow contains all the same sampling instances of
          the lower RTP flow.  A higher RTP flow may contain additional
          sampling instances.

   Note: In some cases, it may be required to add packets in higher RTP
   flows in order to satisfy the second rule above.  This may be



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   achieved by placing empty RTP packets (containing padding data only)
   or by other payload means as, e.g. the Empty NAL unit packet as
   defined in [7].

   If a packet must be inserted for satisfying the above rule, the NTP
   timestamp of such an inserted packet MUST be set equal to the NTP
   timestamp of a packet of the access unit present in any lower RTP
   flow and the lowest RTP flow.  This is easy to accomplish if the
   packet can be inserted at the time of the RTP flow generation, since
   the NTP timestamp must be the same for the inserted packet and the
   packet of the corresponding sample.

   The above rules allow the receiver to process the data of the RTP
   flows as follows:

   o  Go through all received RTP flows starting with the highest RTP
      flow and aggregate the sample data or parts of the sample data
      with the same NTP timestamp in the order of RTP flows, starting
      from the lowest RTP flow up to the highest RTP flow received, to
      the sample with the NTP timestamp present in the highest RTP flow.
      The NTP timestamps MAY be derived using RTCP sender reports or MAY
      be directly taken from the NTP header extension.  The order of RTP
      flows may e.g. be indicated by mechanisms as defined in [6] or any
      other implicit or explicit means.  Repeat the aforementioned
      process for each different NTP timestamp present in the highest
      RTP flow.

   Informative example: The example shown in Figure 3 refers to three
   RTP flows A, B and C containing a layered, a multi-view or a multi-
   description media stream.  In the example, the dependency signalling
   as defined in [6] indicates that flow A is the lowest RTP flow, B is
   the first higher RTP flow and depends on A, and C is the second
   higher RTP flow corresponding to flow A and depends on A and B. A
   picture coding prediction structure is used that results in samples
   present in higher flows but not present in all lower flows.  Flow A
   has the lowest frame rate and Flow B and C have the same but higher
   frame rate.  The figure shows parts of video samples contained in RTP
   packets which are stored in the de-jittering buffer at the receiver
   for de-packetization.  The parts of the video samples are already re-
   ordered according to their RTP sequence number order.  The figure
   indicates for the received sample parts the decoding order within the
   sessions, as well as the associated media (NTP) timestamps
   ("TS[..]").  Parts share the same media timestamp TS, which is shown
   at the bottom of the figure.  Note that the timestamps are not in
   increasing order since, in this example, the decoding order is
   different from the output/display order.

   The process first proceeds to the sample parts associated with the



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   first media timestamp TS[1] present in the highest flow C and
   removes/ignores all preceding (in decoding order) sample parts to
   sample parts with TS[1] in each of the de-jittering buffers of RTP
   flows A, B, and C. Then, starting from flow C, the first media
   timestamp available in decoding order (TS [1]) is selected and sample
   parts starting from RTP flow A, and flow B and C are placed in order
   of the RTP flow dependency as indicated by mechanisms defined in [6]
   (in the example for TS[1]: first flow B and then flow C into the
   video sample AU(TS[1]) associated with media timestamp TS[1].  Then
   the next media timestamp TS[3] in order of appearance in the highest
   RTP flow C is processed and the process described above is repeated.
   Note that there may be video samples with no sample parts present,
   e.g., in the lowest RTP flow A (see, e.g., TS[1]).  With TS[8], the
   first video sample with sample parts present in all the RTP flows
   appears in the buffers.


    C: ------------(1)----(2)---(3)---(4)----(5)--(6)---(7)----(8)----
         |     |     |     |     |     |      |    |     |      |
    B: -(1)---(2)--(3)----(4)---(5)---(6)----(7)--(8)---(9)---(10)----
         |     |                 |     |                 |      |
    A: -------(1)---------------(2)---(3)---------------(4)----(5)----
    --------------------------------------------------decoding order-->

    TS: [4]   [2]   [1]   [3]   [8]   [6]   [5]   [7]   [12]   [10]


    Key:
    A, B, C               - RTP sessions
    Integer values in "()"- Video sample/part of video sample decoding
                            order within RTP session
    "|"                   - indicates corresponding samples / parts of
                            sample of the same video sample AU(TS[..])
                            in the RTP flows
    Integer values in "[]"- media timestamp TS, sampling time
                            as derived from the NTP timestamp associated
                            with the video sample AU(TS[..]), consisting
                            of sample parts in the sessions above.


5.  Security Considerations

   The security considerations of the RTP specification [1] and RTP/AVPF
   profile [2] apply.  No additional security considerations apply due
   to the RTP/AVPF rapid resynchronisation mechanism defined in
   Section 3.1.





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6.  IANA Considerations

   (tbd - this needs to register the new RTP/AVPF transport layer
   feedback packet type)


7.  Acknowledgements

   This memo has benefitted from discussions with numerous members of
   the IETF AVT working group, including Magnus Westerlund, Randell
   Jesup, Jonathan Lennox, Gerard Babonneau, Ingemar Johansson, and Roni
   Even.  The header extension format of Variant A in Section 3.2 was
   suggested by Dave Singer, matching a similar mechanism specified by
   ISMA.


8.  References

8.1.  Normative References

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

   [2]   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.

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

   [4]   Singer, D. and H. Desineni, "A General Mechanism for RTP Header
         Extensions", RFC 5285, July 2008.

   [5]   Mills, D., "Network Time Protocol (Version 3) Specification,
         Implementation", RFC 1305, March 1992.

   [6]   Schierl, T. and S. Wenger, "Signaling media decoding dependency
         in Session Description Protocol (SDP)",
         draft-ietf-mmusic-decoding-dependency-05 (work in progress),
         November 2008.

8.2.  Informative References

   [7]   Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, "RTP
         Payload Format for SVC Video", draft-ietf-avt-rtp-svc-16 (work
         in progress), December 2008.




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

   [9]   Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
         Protocol for Network Address  Translator (NAT) Traversal for
         Offer/Answer Protocols", draft-ietf-mmusic-ice-19 (work in
         progress), October 2007.

   [10]  McGrew, D. and E. Rescorla, "Datagram Transport Layer Security
         (DTLS) Extension to Establish Keys for  Secure Real-time
         Transport Protocol (SRTP)", draft-ietf-avt-dtls-srtp-05 (work
         in progress), September 2008.

   [11]  Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media Path
         Key Agreement for Secure RTP", draft-zimmermann-avt-zrtp-13
         (work in progress), January 2009.


Authors' Addresses

   Colin Perkins
   University of Glasgow
   Department of Computing Science
   Sir Alwyn Williams Building
   Lilybank Gardens
   Glasgow  G12 8QQ
   UK

   Email: csp@csperkins.org


   Thomas Schierl
   Fraunhofer HHI
   Einsteinufer 37
   D-10587 Berlin
   Germany

   Phone: +49-30-31002-227
   Email: thomas.schierl@hhi.fraunhofer.de











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