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Versions: (draft-perkins-avt-rapid-rtp-sync) 00 01 02 03 04 05 06 07 08 09 10 11 12 RFC 6051

Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Updates: RFC3550                                              T. Schierl
(if approved)                                             Fraunhofer HHI
Intended status: Standards Track                           June 30, 2009
Expires: January 1, 2010


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

Status of this Memo

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   Copyright (c) 2009 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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Abstract

   This memo outlines how RTP sessions are synchronised, and discusses



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   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 synchronisation
   delay.  This increase in delay can be unacceptable to some
   applications that use layered and/or multi-description codecs.

   This memo introduces three mechanisms to reduce the synchronisation
   delay for such sessions.  First, it updates the RTP Control Protocol
   (RTCP) timing rules to reduce the initial synchronisation delay for
   SSM sessions.  Second, 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.  Finally,
   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.



































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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Synchronisation of RTP Flows . . . . . . . . . . . . . . . . .  5
     2.1.  Initial Synchronisation Delay  . . . . . . . . . . . . . .  6
       2.1.1.  Unicast Sessions . . . . . . . . . . . . . . . . . . .  6
       2.1.2.  Source Specific Multicast (SSM) Sessions . . . . . . .  7
       2.1.3.  Any Source Multicast (ASM) Sessions  . . . . . . . . .  8
       2.1.4.  Discussion . . . . . . . . . . . . . . . . . . . . . .  9
     2.2.  Synchronisation for Late Joiners . . . . . . . . . . . . . 10
   3.  Reducing RTP Synchronisation Delays  . . . . . . . . . . . . . 10
     3.1.  Reduced Initial RTCP Interval for SSM Senders  . . . . . . 11
     3.2.  Rapid Resynchronisation Request  . . . . . . . . . . . . . 11
     3.3.  In-band Delivery of Synchronisation Metadata . . . . . . . 12
   4.  Application to Decoding Order Recovery in Layered Codecs . . . 14
     4.1.  Problem description  . . . . . . . . . . . . . . . . . . . 14
     4.2.  In-band Synchronisation for Decoding Order Recovery  . . . 15
     4.3.  Timestamp based decoding order recovery  . . . . . . . . . 16
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 19
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 20
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 20
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 21
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22


























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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 RTP flow
   to allow the receiver to establish mappings between the media clock
   used for each RTP flow, and the common (NTP-format) reference 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.  Three backwards compatible extensions to the basic
   RTP synchronisation mechanism are proposed:

   o  The RTCP transmission timing rules are relaxed for SSM senders, to
      reduce the initial synchronisation latency for large SSM groups.
      See Section 3.1.

   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, in the presence of
      packet loss, or when video switching MCUs are employed.  See
      Section 3.2.

   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 flows can be synchronising, although it
      does not eliminate the need for RTCP.  See Section 3.3.

   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 [9]).  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].



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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
   NTP-format timestamp and the RTP timestamp).  Synchronisation
   requires that a common reference clock MUST be used to generate the
   NTP-format timestamps in a set of flows that are to be synchronised.
   Furthermore, to achieve more rapid and accurate synchronisation, it
   is RECOMMENDED that senders and receivers use a common reference
   clock where possible (recognising that this is often not possible
   when RTP is used outside of controlled environments); the means by
   which that common reference clock is distributed are outside the
   scope of this memo.

   For multimedia sessions, 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 or signalled out of band [10].  For layered
   media, different layers can be sent in different RTP sessions, or
   using different SSRC values within a single RTP session; in both
   cases, the CNAME is used to identify flows to be synchronised.  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 [11]).  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



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

   The initial synchronisation delay for layered sessions is similar to
   that for multimedia sessions.  The layers cannot be synchronised
   until the RTCP SR and CNAME information has been received for each
   layer in the session.

2.1.1.  Unicast Sessions

   For unicast multimedia or layered 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. [12]) and/or security keying (e.g. [13],[14]) 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.






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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 or layered 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 average synchronisation delays
   shown in Figure 1.


        Session| Number of receivers:
      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  0.70  0.70  0.70  0.70  0.70  0.70  0.70
       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 in seconds for an RTP
                          Session with 1 sender.

   These numbers assume a source specific 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



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   members in the session.

2.1.3.  Any Source Multicast (ASM) Sessions

   For ASM sessions, the fraction of members that are senders plays an
   important role, and causes more variation in average RTCP reporting
   interval.  This is illustrated in Figure 2 and Figure 3, which show
   the RTCP reporting interval for the same session bandwidths and
   receiver populations as the SSM session described in Figure 1, but
   for sessions with 2 and 10 senders respectively.  It can be seen that
   the initial synchronisation delay scales with the number of senders
   (this is to ensure that the total RTCP traffic from all group members
   does not grow without bound) and can be significantly larger than for
   single source groups.  Despite this, the initial synchronisation time
   remains acceptable for lip-synchronisation in typical small-to-medium
   sized group conferencing scenarios.


        Session| Number of receivers:
      Bandwidth|  2     3     4     5     10   100   1000  10000
             --+------------------------------------------------
         8 kbps| 2.73  4.10  5.47  6.84 10.94 10.94 10.94 10.94
        16 kbps| 2.50  2.50  2.73  3.42  5.47  5.47  5.47  5.47
        32 kbps| 2.50  2.50  2.50  2.50  2.73  2.73  2.73  2.73
        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  0.70  0.70  0.70  0.70  0.70  0.70  0.70
       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 2: Average RTCP reporting interval in seconds for an RTP
                          Session with 2 senders.

















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        Session| Number of receivers:
      Bandwidth|  2     3     4     5     10   100   1000  10000
             --+------------------------------------------------
         8 kbps| 2.73  4.10  5.47  6.84 13.67 54.69 54.69 54.69
        16 kbps| 2.50  2.50  2.73  3.42  6.84 27.34 27.34 27.34
        32 kbps| 2.50  2.50  2.50  2.50  3.42 13.67 13.67 13.67
        64 kbps| 2.50  2.50  2.50  2.50  2.50  6.84  6.84  6.84
       128 kbps| 1.41  1.41  1.41  1.41  1.41  3.42  3.42  3.42
       256 kbps| 0.70  0.70  0.70  0.70  0.70  1.71  1.71  1.71
       512 kbps| 0.35  0.35  0.35  0.35  0.35  0.85  0.85  0.85
         1 Mbps| 0.18  0.18  0.18  0.18  0.18  0.43  0.43  0.43
         2 Mbps| 0.09  0.09  0.09  0.09  0.09  0.21  0.21  0.21
         4 Mbps| 0.04  0.04  0.04  0.04  0.04  0.11  0.11  0.11

      Figure 3: Average RTCP reporting interval in seconds for an RTP
                         Session with 10 senders.

   Note that multi-sender groups implemented using multi-unicast with a
   central RTP translator (Topo-Translator in the terminology of [15])
   or mixer (Topo-Mixer), or some forms of video switching MCU (Topo-
   Video-switch-MCU) distribute RTCP packets to all members of the
   group, and so scale in the same way as an ASM group with regards to
   initial synchronisation latency.

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
   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 is 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.
   For ASM sessions, implosion of responses is a concern, so no change
   is proposed to the RTCP timing rules.

   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 3.2 defines a new RTP/AVPF
   transport layer feedback message to request an RTCP SR be generated,



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   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 isn'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 with
       high accuracy given sufficient reports.  Collecting sufficient
       RTCP SR data to perform this estimation, however, may require
       reception of several RTCP reports, further increasing the
       synchronisation delay.

   3.  Many media codecs have the notion of periodic access points, such
       that a newly joined receiver often cannot start decoding a media
       stream until the packets corresponding to the access point have
       been received.  These access points may be sent less often than
       RTCP SR packets, and so may be the limiting factor in starting
       synchronised media playout for late joiners.

   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

   Three backwards compatible RTP extensions are defined to reduce the
   possible synchronisation delay: a reduced initial RTCP interval for
   SSM senders, a rapid resynchronisation request message, and RTP
   header extensions that can convey synchronisation metadata in-band.



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3.1.  Reduced Initial RTCP Interval for SSM Senders

   In SSM sessions where the initial synchronisation delay is important,
   the RTP sender MAY set the delay before sending the initial compound
   RTCP packet to zero, and send its first RTCP packet immediately upon
   joining the SSM session.  RTP receivers in an SSM session, sending
   unicast RTCP feedback, MUST NOT send RTCP packets with zero initial
   delay; the timing rules defined in [4] apply unchanged to receivers.

3.2.  Rapid Resynchronisation Request

   The general format of an RTP/AVPF transport layer feedback message is
   shown in Figure 4 (see [2] for details).


       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)                 :
      :                                                               :

            Figure 4: RTP/AVP Transport Layer Feedback Message

   A new feedback message type, RTCP-SR-REQ, is defined with FMT = 5.
   The Feedback Control Information (FCI) part of the feedback message
   MUST be empty.  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 length MUST equal 2.

   This feedback message MAY be sent by a receiver to indicate that it's
   unable to synchronise some media streams, and desires that the media
   source transmit an RTCP SR packet as soon as possible (within the
   constraints of the RTCP timing rules for early feedback).  When it
   receives such an indication, the media source SHOULD generate an RTCP
   SR packet as soon as possible within the RTCP early feedback rules.
   If the use of non-compound RTCP [5] was previously negotiated, both
   the feedback request and the RTCP SR response may be sent as non-
   compound RTCP packets.  The RTCP-SR-REQ packet MAY be repeated once
   per RTCP reporting interval if no RTCP SR packet is forthcoming.

   When using SSM sessions with unicast feedback, is possible that the



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   feedback target and media source are not co-located.  If a feedback
   target receives an RTCP-SR-REQ feedback message in such a case, the
   request should be forwarded to the media source.  The mechanism to be
   used for forwarding such requests is not defined here.

3.3.  In-band Delivery of Synchronisation Metadata

   The RTP header extension mechanism defined in [6] can be adopted to
   carry an OPTIONAL NTP format 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.  Provided it has knowledge of the SSRC to CNAME
   mapping, either from prior receipt of an RTCP CNAME packet or via
   out-of-band signalling [10], the receiver can use the information
   provided as input to the synchronisation algorithm, in exactly the
   same way as if an additional RTCP SR packet was 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 [7].  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/64-bit NTP RTP header extension (length: 16 bytes):

       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/56-bit NTP RTP header extension (length: 12 bytes):

       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 further specified in Section 4.2.  This header
   extension SHOULD 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 in multimedia sessions,
   and in video switching scenarios.

   Note: The inclusion of an RTP header extension will reduce the
   efficiency of RTP header compression, if it is used.  Furthermore,
   middle boxes which do not understand the header extensions may remove
   them or may not update the content according to this memo.

   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], and robustness in the face of middleboxes
   (RTP translators) that might strip RTP header extensions.  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 Variant
   B/56-bit NTP RTP header extension (although this may generally be
   inferred from context).




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   When the SDP is used, the use of the RTP header extensions defined
   above MUST be indicated as specified in [6].  Therefore the following
   URIs MUST be used:

   o  The URI used for signaling the use of Variant A/64-bit NTP RTP
      header extension in SDP is "urn:ietf:params:rtp-hdrext:ntp-64".

   o  The URI used for signaling the use of Variant B/56-bit NTP RTP
      header extension in SDP is "urn:ietf:params:rtp-hdrext:ntp-56".


4.  Application to Decoding Order Recovery in Layered Codecs

   Based on the timestamp contained in each RTP data packet, and the
   mapping to an NTP format timestamp, a decoding order recovery process
   may be 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 multiple 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 in NTP-format clock, it may not be
   possible to derive exact matching NTP timestamps using the NTP format
   clock 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, while having a skew between those samples in the RTP flows
   RTCP sender reports.  If the RTCP SR packets are not send
   synchronously in the multiple RTP flows, they therefore do not
   contain the same NTP-format timestamp.  If there is a skew present in
   the clock used for NTP-format timestamp generation, using different
   NTP-format timestamps for the same sampling instance in the RTP flow
   inevitably leads to non-matching NTP timestamps generated from RTP
   timestamps and NTP-format timestamps 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



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   exact inter-flow alignment of NTP timestamps can be guaranteed, if an
   RTP header extension containing an NTP timestamp 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.  This is called
   synchronous insertion of RTP header extensions in the following.

4.2.  In-band Synchronisation for Decoding Order Recovery

   The RTP header extension to convey an NTP timestamp 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 this header extension is inserted for
   RTP flows transporting samples or parts of samples of the same
   layered media, it SHALL be included at least once in each of the RTP
   flows of the same media for the sampling time instance of an
   insertion of an RTP header extension.  Such synchronously inserted
   RTP header extensions SHALL contain the same NTP format timestamp.
   The frequency of inserting the header extensions in the RTP flows is
   up to the sender, but it should be noticed that higher insertion
   frequencies obviously lead to higher synchronization frequencies.
   For use cases where the same clock source has been used to generate
   the RTP timestamps in the multiple RTP flows, an application MAY rely
   on the RTP timestamps only for decoding order recovery starting from
   the point of synchronous insertion of the RTP header extensions
   containing NTP timestamps.

   Note: If the decoding order of RTP flows is given by any means (as
   e.g., for RTP session by mechanism defined in [8]), 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.  There
   may be future mechanism to allow indication of dependencies of RTP
   flows transported as RTP streams using SSRC multiplexing

   It is RECOMMENDED that the receiver uses for timestamp-based decoding
   order recovery the NTP timestamps provided in the RTP header
   extensions only, if such extensions are present for the RTP flows.
   Section 4.3 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



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   solves the possible problem of clock skews identified for the NI-T
   mode as defined in [9].  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 RTP 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
   RTP header extension containing the NTP timestamp 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, i.e. 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-
   format timestamp provided in the RTCP sender report packets.

   In order to guarantee the exact alignment of those derived NTP
   presentation timestamps, the RTP header extension as defined in this
   memo in Section 3.3 allows the receiver to start the decoding order
   recovery before the reception of a RTCP sender report if the RTP
   header extension is earlier provided in the RTP flow.  Using the RTP
   header extensions may be the only way to allow correct decoding order
   recovery based on exact matching of NTP timestamps in the presence of
   clock skew in the clock used for generating the NTP format clock.

   Furthermore, some use cases may allow to use synchronously inserted
   RTP header extensions containing NTP timestamps to align the RTP
   timestamps of the multiple RTP flows, i.e. use cases where the RTP
   timestamps of the multiple RTP flows are generated from the same
   clock source.  In such use cases, starting from a synchronous
   insertion of the RTP header extensions, the application may use the
   detected difference of RTP random offset values in the multiple
   sessions to align the media samples of parts of samples.

   Since typically for layered video codecs as, e.g.  SVC [9], 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



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   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 [8] 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
   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 [9].

   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 same sample 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.




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   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 timestamp provided in an RTP header
      extension.  The order of RTP flows may e.g. be indicated by
      mechanisms as defined in [8] 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 [8] 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
   media coding 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 the full video samples with their
   corresponding RTP timestamps "(x)".  The video samples are already
   re-ordered according to their RTP sequence number order.  The figure
   indicates for the received sample in decoding order within each RTP
   flow, as well as the associated NTP media timestamps ("TS[..]").
   These timestamps may be derived using the NTP format timestamp
   provided in the RTCP sender reports or as shown in the figure
   directly from the NTP timestamp contained in the RTP header
   extensions as indicate by the timestamp in "<x>".  Note that the
   timestamps are not in increasing order since, in this example, the
   decoding order is different from the output/presentation order.

   The process first proceeds to the sample parts associated with the
   first available synchronous insertion of NTP timestamp into RTP
   header extensions at NTP media timestamp TS=[8] and starts in the
   highest RTP flow C and removes/ignores all preceding sample parts (in
   decoding order) to sample parts with TS=[8] 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=[8]) 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 [8] (in the example for TS[8]: first flow B and
   then flow C into the video sample VS(TS[8]) associated with NTP media
   timestamp TS=[8].  Then the next media timestamp TS=[6] (RTP



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   timestamp=(4)) 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=[5]).  The decoding order recovery
   process could be also started after receiving all RTP sender reports
   RTP timestamp to NTP-format timestamp mapping (indicated as
   timestamps "(x){y}") assuming that there is no clock skew in the
   source used for the NTP-format timestamp generation.



   C:-(0)----(2)----(7)<8>--(5)----(4)----(6)-----(11)----(9){10}-
      |      |      |       |      |      |       |       |
   B:-(3)----(5)---(10)<8>--(8)----(7)----(9){7}--(14)----(12)----
                    |       |                     |       |
   A:---------------(3)<8>--(1)-------------------(7){12}-(5)-----

   ---------------------------------------decoding/transmission order->
   TS:[1]    [3]    [8]=<8> [6]    [5]    [7]    [12]    [10]


   Key:
   A, B, C               - RTP flows
   Integer values in "()"- video sample with its RTP timestamp as
                          indicated in its RTP packet.
   "|"                   - indicates corresponding samples / parts of
                          sample of the same video sample VS(TS[..])
                          in the RTP flows.
   Integer values in "[]"- NTP media timestamp TS, sampling time
                          as derived from the NTP timestamp associated
                          with the video sample AU(TS[..]), consisting
                          of sample parts in the flows above.
   Integer values in "<>"- NTP media timestamp TS as directly
                         taken from the NTP RTP header extensions.
   Integer values in "{}"- NTP media timestamp TS as provided in the
                          RTCP sender reports.



5.  Security Considerations

   The security considerations of the RTP specification [1], the
   Extended RTP profile for RTCP-Based Feedback [2], and the General
   Mechanism for RTP Header Extensions [6] apply.  The extensions we
   define in this memo are not believed to introduce any additional
   security considerations.





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6.  IANA Considerations

   NOTE TO RFC EDITOR: Please replace "RFC XXXX" in the following with
   the RFC number assigned to this memo, and delete this note.

   The IANA is requested to register one new value in the table of FMT
   Values for RTPFB Payload Types [2] as follows:


      Name:          RTCP-SR-REQ
      Long name:     RTCP Rapid Resynchronisation Request
      Value:         5
      Reference:     RFC XXXX

   The IANA is also requested to register two new RTP Compact Header
   Extensions [6], according to the following:


      Extension URI: urn:ietf:params:rtp-hdrext:ntp-64
      Description:   Synchronisation metadata: 64-bit timestamp format
      Contact:       Thomas Schierl <Thomas.Schierl@hhi.fraunhofer.de>
                     IETF Audio/Video Transport Working Group
      Reference:     RFC XXXX

      Extension URI: urn:ietf:params:rtp-hdrext:ntp-56
      Description:   Synchronisation metadata: 56-bit timestamp format
      Contact:       Thomas Schierl <Thomas.Schierl@hhi.fraunhofer.de>
                     IETF Audio/Video Transport Working Group
      Reference:     RFC XXXX


7.  Acknowledgements

   This memo has benefitted from discussions with numerous members of
   the IETF AVT working group, including Jonathan Lennox, Magnus
   Westerlund, Randell Jesup, Gerard Babonneau, Ingemar Johansson, Ali
   C. Begen, Ye-Kui Wang, Roni Even, Michael Dolan, and Art Allison.
   The header extension format of Variant A in Section 3.3 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.



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   [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]   Schooler, E., Ott, J., and J. Chesterfield, "RTCP Extensions
         for Single-Source Multicast Sessions with Unicast Feedback",
         draft-ietf-avt-rtcpssm-18 (work in progress), March 2009.

   [5]   Johansson, I. and M. Westerlund, "Support for Reduced-Size
         Real-Time Transport Control Protocol (RTCP): Opportunities and
         Consequences", RFC 5506, April 2009.

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

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

   [8]   Schierl, T. and S. Wenger, "Signaling media decoding dependency
         in Session Description Protocol (SDP)",
         draft-ietf-mmusic-decoding-dependency-08 (work in progress),
         April 2009.

8.2.  Informative References

   [9]   Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, "RTP
         Payload Format for SVC Video", draft-ietf-avt-rtp-svc-18 (work
         in progress), March 2009.

   [10]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific Media
         Attributes in the Session Description Protocol (SDP)",
         draft-ietf-mmusic-sdp-source-attributes-02 (work in progress),
         October 2008.

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

   [12]  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.

   [13]  McGrew, D. and E. Rescorla, "Datagram Transport Layer Security
         (DTLS) Extension to Establish Keys for  Secure Real-time



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         Transport Protocol (SRTP)", draft-ietf-avt-dtls-srtp-05 (work
         in progress), September 2008.

   [14]  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.

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


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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