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RFC 6184

 
 
 

RTP Payload Format for H.264 Video

Part 4 of 4, p. 68 to 101
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8.3.  Examples

   An SDP Offer/Answer exchange wherein both parties are expected to
   both send and receive could look like the following.  Only the media-
   codec-specific parts of the SDP are shown.  Some lines are wrapped
   due to text constraints.

      Offerer -> Answerer SDP message:

      m=video 49170 RTP/AVP 100 99 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; packetization-mode=0;
        sprop-parameter-sets=<parameter sets data#0>
      a=rtpmap:99 H264/90000
      a=fmtp:99 profile-level-id=42A01E; packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#1>
      a=rtpmap:100 H264/90000
      a=fmtp:100 profile-level-id=42A01E; packetization-mode=2;
        sprop-parameter-sets=<parameter sets data#2>;
        sprop-interleaving-depth=45; sprop-deint-buf-req=64000;
        sprop-init-buf-time=102478; deint-buf-cap=128000

   The above offer presents the same codec configuration in three
   different packetization formats.  Payload type 98 represents single
   NALU mode, payload type 99 represents non-interleaved mode, and
   payload type 100 indicates the interleaved mode.  In the interleaved
   mode case, the interleaving parameters that the offerer would use if
   the answer indicates support for payload type 100 are also included.
   In all three cases, the parameter sprop-parameter-sets conveys the
   initial parameter sets that are required by the answerer when
   receiving a stream from the offerer when this configuration is
   accepted.  Note that the value for sprop-parameter-sets could be
   different for each payload type.

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      Answerer -> Offerer SDP message:

      m=video 49170 RTP/AVP 100 99 97
      a=rtpmap:97 H264/90000
      a=fmtp:97 profile-level-id=42A01E; packetization-mode=0;
        sprop-parameter-sets=<parameter sets data#3>
      a=rtpmap:99 H264/90000
      a=fmtp:99 profile-level-id=42A01E; packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#4>;
        max-rcmd-nalu-size=3980
      a=rtpmap:100 H264/90000
      a=fmtp:100 profile-level-id=42A01E; packetization-mode=2;
        sprop-parameter-sets=<parameter sets data#5>;
        sprop-interleaving-depth=60;
        sprop-deint-buf-req=86000; sprop-init-buf-time=156320;
        deint-buf-cap=128000; max-rcmd-nalu-size=3980

   As the Offer/Answer negotiation covers both sending and receiving
   streams, an offer indicates the exact parameters for what the offerer
   is willing to receive, whereas the answer indicates the same for what
   the answerer is willing to receive.  In this case, the offerer
   declared that it is willing to receive payload type 98.  The answerer
   accepts this by declaring an equivalent payload type 97; that is, it
   has identical values for the two parameters profile-level-id and
   packetization-mode (since packetization-mode is equal to 0 and sprop-
   deint-buf-req is not present).  As the offered payload type 98 is
   accepted, the answerer needs to store parameter sets included in
   sprop-parameter-sets=<parameter sets data#0> in case the offer
   finally decides to use this configuration.  In the answer, the
   answerer includes the parameter sets in sprop-parameter-
   sets=<parameter sets data#3> that the answerer would use in the
   stream sent from the answerer if this configuration is finally used.

   The answerer also accepts the reception of the two configurations
   that payload types 99 and 100 represent.  Again, the answerer needs
   to store parameter sets included in sprop-parameter-sets=<parameter
   sets data#1> and sprop-parameter-sets=<parameter sets data#2> in case
   the offer finally decides to use either of these two configurations.
   The answerer provides the initial parameter sets for the answerer-to-
   offerer direction, i.e., the parameter sets in sprop-parameter-
   sets=<parameter sets data#4> and sprop-parameter-sets=<parameter sets
   data#5>, for payload types 99 and 100, respectively, that it will use
   to send the payload types.  The answerer also provides the offerer
   with its memory limit for de-interleaving operations by providing a
   deint-buf-cap parameter.  This is only useful if the offerer decides
   on making a second offer, where it can take the new value into

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   account.  The max-rcmd-nalu-size indicates that the answerer can
   efficiently process NALUs up to the size of 3980 bytes.  However,
   there is no guarantee that the network supports this size.

   In the following example, the offer is accepted without level
   downgrading (i.e., the default level, Level 3.0, is accepted), and
   both sprop-parameter-sets and sprop-level-parameter-sets are present
   in the offer.  The answerer must ignore sprop-level-parameter-
   sets=<parameter sets data#1> and store parameter sets in sprop-
   parameter-sets=<parameter sets data#0> for decoding the incoming NAL
   unit stream.  The offerer must store the parameter sets in sprop-
   parameter-sets=<parameter sets data#2> in the answer for decoding the
   incoming NAL unit stream.  Note that in this example, parameter sets
   in sprop-parameter-sets=<parameter sets data#2> must be associated
   with Level 3.0.

      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#0>;
        sprop-level-parameter-sets=<parameter sets data#1>

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#2>

   In the following example, the offer (Baseline profile, Level 1.1) is
   accepted with level downgrading (the accepted level is Level 1b), and
   both sprop-parameter-sets and sprop-level-parameter-sets are present
   in the offer.  The answerer must ignore sprop-parameter-
   sets=<parameter sets data#0> and all parameter sets not for the
   accepted level (Level 1b) in sprop-level-parameter-sets=<parameter
   sets data#1> and must store parameter sets for the accepted level
   (Level 1b) in sprop-level-parameter-sets=<parameter sets data#1> for
   decoding the incoming NAL unit stream.  The offerer must store the
   parameter sets in sprop-parameter-sets=<parameter sets data#2> in the
   answer for decoding the incoming NAL unit stream.  Note that in this
   example, parameter sets in sprop-parameter-sets=<parameter sets
   data#2> must be associated with Level 1b.

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      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A00B; //Baseline profile, Level 1.1
        packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#0>;
        sprop-level-parameter-sets=<parameter sets data#1>

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42B00B; //Baseline profile, Level 1b
        packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#2>;
        use-level-src-parameter-sets=1

   In the following example, the offer (Baseline profile, Level 1.1) is
   accepted with level downgrading (the accepted level is Level 1b), and
   both sprop-parameter-sets and sprop-level-parameter-sets are present
   in the offer.  However, the answerer is a legacy RFC 3984
   implementation and does not understand sprop-level-parameter-sets;
   hence, it does not include use-level-src-parameter-sets (which the
   answerer does not understand either) in the answer.  Therefore, the
   answerer must ignore both sprop-parameter-sets=<parameter sets
   data#0> and sprop-level-parameter-sets=<parameter sets data#1>, and
   the offerer must transport parameter sets in-band.

      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A00B; //Baseline profile, Level 1.1
        packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#0>;
        sprop-level-parameter-sets=<parameter sets data#1>

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42B00B; //Baseline profile, Level 1b
        packetization-mode=1

   In the following example, the offer is accepted without level
   downgrading, and sprop-parameter-sets is present in the offer.
   Parameter sets in sprop-parameter-sets=<parameter sets data#0> must

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   be stored and used by the encoder of the offerer and the decoder of
   the answerer, and parameter sets in sprop-parameter-sets=<parameter
   sets data#1> must be used by the encoder of the answerer and the
   decoder of the offerer.  Note that sprop-parameter-sets=<parameter
   sets data#0> is basically independent of sprop-parameter-
   sets=<parameter sets data#1>.

      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#0>

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#1>

   In the following example, the offer is accepted without level
   downgrading, and neither sprop-parameter-sets nor sprop-level-
   parameter-sets is present in the offer, meaning that there is no out-
   of-band transmission of parameter sets, which then have to be
   transported in-band.

      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1

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   In the following example, the offer is accepted with level
   downgrading and sprop-parameter-sets is present in the offer.  As
   sprop-parameter-sets=<parameter sets data#0> contains level_idc
   indicating Level 3.0, it therefore cannot be used, as the answerer
   wants Level 2.0, and must be ignored by the answerer, and in-band
   parameter sets must be used.

      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1;
        sprop-parameter-sets=<parameter sets data#0>

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A014; //Baseline profile, Level 2.0
        packetization-mode=1

   In the following example, the offer is also accepted with level
   downgrading, and neither sprop-parameter-sets nor sprop-level-
   parameter-sets is present in the offer, meaning that there is no out-
   of-band transmission of parameter sets, which then have to be
   transported in-band.

      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A014; //Baseline profile, Level 2.0
        packetization-mode=1

   In the following example, the offer is accepted with level upgrading,
   and neither sprop-parameter-sets nor sprop-level-parameter-sets is
   present in the offer or the answer, meaning that there is no out-of-
   band transmission of parameter sets, which then have to be
   transported in-band.  The level to use in the offerer-to-answerer
   direction is Level 3.0, and the level to use in the answerer-to-

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   offerer direction is Level 2.0.  The answerer is allowed to send at
   any level up to and including Level 2.0, and the offerer is allowed
   to send at any level up to and including Level 3.0.

      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A014; //Baseline profile, Level 2.0
        packetization-mode=1; level-asymmetry-allowed=1

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1; level-asymmetry-allowed=1

   In the following example, the offerer is a Multipoint Control Unit
   (MCU) in a topology like Topo-Video-switch-MCU [29], offering
   parameter sets received (using out-of-band transport) from three
   other participants (B, C, and D) and receiving parameter sets from
   the participant A, which is the answerer.  The participants are
   identified by their values of canonical name (CNAME), which are
   mapped to different SSRC values.  The same codec configuration is
   used by all four participants.  The participant A stores and
   associates the parameter sets included in <parameter sets data#B>,
   <parameter sets data#C>, and <parameter sets data#D> to participants
   B, C, and D, respectively, and uses <parameter sets data#B> for
   decoding NAL units carried in RTP packets originating from
   participant B only, uses <parameter sets data#C> for decoding NAL
   units carried in RTP packets originating from participant C only, and
   uses <parameter sets data#D> for decoding NAL units carried in RTP
   packets originating from participant D only.

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      Offer SDP:

      m=video 49170 RTP/AVP 98
      a=ssrc:SSRC-B cname:CNAME-B
      a=ssrc:SSRC-C cname:CNAME-C
      a=ssrc:SSRC-D cname:CNAME-D
      a=ssrc:SSRC-B fmtp:98
        sprop-parameter-sets=<parameter sets data#B>
      a=ssrc:SSRC-C fmtp:98
        sprop-parameter-sets=<parameter sets data#C>
      a=ssrc:SSRC-D fmtp:98
        sprop-parameter-sets=<parameter sets data#D>
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1

      Answer SDP:

      m=video 49170 RTP/AVP 98
      a=ssrc:SSRC-A cname:CNAME-A
      a=ssrc:SSRC-A fmtp:98
        sprop-parameter-sets=<parameter sets data#A>
      a=rtpmap:98 H264/90000
      a=fmtp:98 profile-level-id=42A01E; //Baseline profile, Level 3.0
        packetization-mode=1

8.4.  Parameter Set Considerations

   The H.264 parameter sets are a fundamental part of the video codec
   and vital to its operation (see Section 1.2).  Due to their
   characteristics and their importance for the decoding process, lost
   or erroneously transmitted parameter sets can hardly be concealed
   locally at the receiver.  A reference to a corrupt parameter set
   normally has fatal results to the decoding process.  Corruption could
   occur, for example, due to the erroneous transmission or loss of a
   parameter set NAL unit but also due to the untimely transmission of a
   parameter set update.  A parameter set update refers to a change of
   at least one parameter in a picture parameter set or sequence
   parameter set for which the picture parameter set or sequence
   parameter set identifier remains unchanged.  Therefore, the following
   recommendations are provided as a guideline for the implementer of
   the RTP sender.

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   Parameter set NALUs can be transported using three different
   principles:

   A.  Using a session control protocol (out-of-band) prior to the
       actual RTP session.

   B.  Using a session control protocol (out-of-band) during an ongoing
       RTP session.

   C.  Within the RTP packet stream in the payload (in-band) during an
       ongoing RTP session.

   It is recommended to implement principles A and B within a session
   control protocol.  SIP and SDP can be used as described in the SDP
   Offer/Answer model and in the previous sections of this memo.
   Section 8.2.2 includes a detailed discussion on transport of
   parameter sets in-band or out-of-band in SDP Offer/Answer using media
   type parameters sprop-parameter-sets, sprop-level-parameter-sets,
   use-level-src-parameter-sets, and in-band-parameter-sets.  This
   section contains guidelines on how principles A and B should be
   implemented within session control protocols.  It is independent of
   the particular protocol used.  Principle C is supported by the RTP
   payload format defined in this specification.  There are topologies
   like Topo-Video-switch-MCU [29] for which the use of principle C may
   be desirable.

   If in-band signaling of parameter sets is used, the picture and
   sequence parameter set NALUs SHOULD be transmitted in the RTP payload
   using a reliable method of delivering of RTP (see below), as a loss
   of a parameter set of either type will likely prevent decoding of a
   considerable portion of the corresponding RTP packet stream.

   If in-band signaling of parameter sets is used, the sender SHOULD
   take the error characteristics into account and use mechanisms to
   provide a high probability for delivering the parameter sets
   correctly.  Mechanisms that increase the probability for a correct
   reception include packet repetition, FEC, and retransmission.  The
   use of an unreliable, out-of-band control protocol has similar
   disadvantages as the in-band signaling (possible loss) and, in
   addition, may also lead to difficulties in the synchronization (see
   below).  Therefore, it is NOT RECOMMENDED.

   Parameter sets MAY be added or updated during the lifetime of a
   session using principles B and C.  It is required that parameter sets
   be present at the decoder prior to the NAL units that refer to them.
   Update or addition of parameter sets can result in further problems;
   therefore, the following recommendations should be considered.

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   -  When parameter sets are added or updated, care SHOULD be taken to
      ensure that any parameter set is delivered prior to its usage.
      When new parameter sets are added, previously unused parameter set
      identifiers are used.  It is common that no synchronization is
      present between out-of-band signaling and in-band traffic.  If
      out-of-band signaling is used, it is RECOMMENDED that a sender not
      start sending NALUs requiring the added or updated parameter sets
      prior to acknowledgement of delivery from the signaling protocol.

   -  When parameter sets are updated, the following synchronization
      issue should be taken into account.  When overwriting a parameter
      set at the receiver, the sender has to ensure that the parameter
      set in question is not needed by any NALU present in the network
      or receiver buffers.  Otherwise, decoding with a wrong parameter
      set may occur.  To lessen this problem, it is RECOMMENDED either
      to overwrite only those parameter sets that have not been used for
      a sufficiently long time (to ensure that all related NALUs have
      been consumed) or to add a new parameter set instead (which may
      have negative consequences for the efficiency of the video
      coding).

         Informative note: In some topologies like Topo-Video-switch-
         MCU [29], the origin of the whole set of parameter sets may
         come from multiple sources that may use non-unique parameter
         set identifiers.  In this case, an offer may overwrite an
         existing parameter set if no other mechanism that enables
         uniqueness of the parameter sets in the out-of-band channel
         exists.

   -  In a multiparty session, one participant MUST associate parameter
      sets coming from different sources with the source identification
      whenever possible, e.g., by conveying out-of-band transported
      parameter sets, as different sources typically use independent
      parameter set identifier value spaces.

   -  Adding or modifying parameter sets by using both principles B and
      C in the same RTP session may lead to inconsistencies of the
      parameter sets because of the lack of synchronization between the
      control and the RTP channel.  Therefore, principles B and C MUST
      NOT both be used in the same session unless sufficient
      synchronization can be provided.

   In some scenarios (e.g., when only the subset of this payload format
   specification corresponding to H.241 is used) or topologies, it is
   not possible to employ out-of-band parameter set transmission.  In
   this case, parameter sets have to be transmitted in-band.  Here, the
   synchronization with the non-parameter-set-data in the bitstream is
   implicit, but the possibility of a loss has to be taken into account.

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   The loss probability should be reduced using the mechanisms discussed
   above.  In case a loss of a parameter set is detected, recovery may
   be achieved using a Decoder Refresh Point procedure, for example,
   using RTCP feedback Full Intra Request (FIR) [30].  Two example
   Decoder Refresh Point procedures are provided in the informative
   Section 8.5.

   -  When parameter sets are initially provided using principle A and
      then later added or updated in-band (principle C), there is a risk
      associated with updating the parameter sets delivered out-of-band.
      If receivers miss some in-band updates (for example, because of a
      loss or a late tune-in), those receivers attempt to decode the
      bitstream using outdated parameters.  It is therefore RECOMMENDED
      that parameter set IDs be partitioned between the out-of-band and
      in-band parameter sets.

8.5.  Decoder Refresh Point Procedure Using In-Band Transport of
      Parameter Sets (Informative)

   When a sender with a video encoder according to [1] receives a
   request for a decoder refresh point, the encoder shall enter the fast
   update mode by using one of the procedures specified in Sections
   8.5.1 or 8.5.2.  The procedure in Section 8.5.1 is the preferred
   response in a lossless transmission environment.  Both procedures
   satisfy the requirement to enter the fast update mode for H.264 video
   encoding.

8.5.1.  IDR Procedure to Respond to a Request for a Decoder Refresh
        Point

   This section gives one possible way to respond to a request for a
   decoder refresh point.

   The encoder shall, in the order presented here:

   1) Immediately prepare to send an IDR picture.

   2) Send a sequence parameter set to be used by the IDR picture to be
      sent.  The encoder may optionally also send other sequence
      parameter sets.

   3) Send a picture parameter set to be used by the IDR picture to be
      sent.  The encoder may optionally also send other picture
      parameter sets.

   4) Send the IDR picture.

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   5) From this point forward in time, send any other sequence or
      picture parameter sets that have not yet been sent in this
      procedure, prior to their reference by any NAL unit, regardless of
      whether such parameter sets were previously sent prior to
      receiving the request for a decoder refresh point.  As needed,
      such parameter sets may be sent in a batch, one at a time, or in
      any combination of these two methods.  Parameter sets may be
      re-sent at any time for redundancy.  Caution should be taken when
      parameter set updates are present, as described above in Section
      8.4.

8.5.2.  Gradual Recovery Procedure to Respond to a Request for a Decoder
        Refresh Point

   This section gives another possible way to respond to a request for a
   decoder refresh point.

   The encoder shall, in the order presented here:

   1) Send a recovery point SEI message (see Sections D.1.7 and D.2.7 of
      [1]).

   2) Repeat any sequence and picture parameter sets that were sent
      before the recovery point SEI message, prior to their reference by
      a NAL unit.

   The encoder shall ensure that the decoder has access to all reference
   pictures for inter prediction of pictures at or after the recovery
   point, which is indicated by the recovery point SEI message, in
   output order, assuming that the transmission from now on is error-
   free.

   The value of the recovery_frame_cnt syntax element in the recovery
   point SEI message should be small enough to ensure a fast recovery.

   As needed, such parameter sets may be re-sent in a batch, one at a
   time, or in any combination of these two methods.  Parameter sets may
   be re-sent at any time for redundancy.  Caution should be taken when
   parameter set updates are present, as described above in Section 8.4.

9.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [5] and in any appropriate RTP profile (for example,
   [16]).  This implies that confidentiality of the media streams is
   achieved by encryption, for example, through the application of SRTP
   [26].  Because the data compression used with this payload format is

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   applied end-to-end, any encryption needs to be performed after
   compression.  A potential denial-of-service threat exists for data
   encodings using compression techniques that have non-uniform
   receiver-end computational load.  The attacker can inject
   pathological datagrams into the stream that are complex to decode and
   that cause the receiver to be overloaded.  H.264 is particularly
   vulnerable to such attacks, as it is extremely simple to generate
   datagrams containing NAL units that affect the decoding process of
   many future NAL units.  Therefore, the usage of data origin
   authentication and data integrity protection of at least the RTP
   packet is RECOMMENDED, for example, with SRTP [26].

   Note that the appropriate mechanism to ensure confidentiality and
   integrity of RTP packets and their payloads is very dependent on the
   application and on the transport and signaling protocols employed.
   Thus, although SRTP is given as an example above, other possible
   choices exist.

   Decoders MUST exercise caution with respect to the handling of user
   data SEI messages, particularly if they contain active elements, and
   MUST restrict their domain of applicability to the presentation
   containing the stream.

   End-to-end security with either authentication, integrity, or
   confidentiality protection will prevent a MANE from performing media-
   aware operations other than discarding complete packets.  In the case
   of confidentiality protection, it will even be prevented from
   discarding packets in a media-aware way.  To be allowed to perform
   its operations, a MANE is required to be a trusted entity that is
   included in the security context establishment.

10.  Congestion Control

   Congestion control for RTP SHALL be used in accordance with RFC 3550
   [5] and with any applicable RTP profile, e.g., RFC 3551 [16].  If
   best-effort service is being used, an additional requirement is that
   users of this payload format MUST monitor packet loss to ensure that
   the packet loss rate is within acceptable parameters.  Packet loss is
   considered acceptable if a TCP flow across the same network path, and
   experiencing the same network conditions, would achieve an average
   throughput, measured on a reasonable timescale, that is not less than
   the RTP flow is achieving.  This condition can be satisfied by
   implementing congestion control mechanisms to adapt the transmission
   rate (or the number of layers subscribed for a layered multicast
   session) or by arranging for a receiver to leave the session if the
   loss rate is unacceptably high.

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   The bitrate adaptation necessary for obeying the congestion control
   principle is easily achievable when real-time encoding is used.
   However, when pre-encoded content is being transmitted, bandwidth
   adaptation requires the availability of more than one coded
   representation of the same content, at different bitrates, or the
   existence of non-reference pictures or sub-sequences [22] in the
   bitstream.  The switching between the different representations can
   normally be performed in the same RTP session, e.g., by employing a
   concept known as SI/SP slices of the Extended profile or by switching
   streams at IDR picture boundaries.  Only when non-downgradable
   parameters (such as the profile part of the profile/level ID) are
   required to be changed does it become necessary to terminate and
   restart the media stream.  This may be accomplished by using a
   different RTP payload type.

   MANEs MAY follow the suggestions outlined in Section 7.3 and remove
   certain unusable packets from the packet stream when that stream was
   damaged due to previous packet losses.  This can help reduce the
   network load in certain special cases.

11.  IANA Considerations

   The H264 media subtype name specified by RFC 3984 has been updated as
   defined in Section 8.1 of this memo.

12.  Informative Appendix: Application Examples

   This payload specification is very flexible in its use, in order to
   cover the extremely wide application space anticipated for H.264.
   However, this great flexibility also makes it difficult for an
   implementer to decide on a reasonable packetization scheme.  Some
   information on how to apply this specification to real-world
   scenarios is likely to appear in the form of academic publications
   and a test model software and description in the near future.
   However, some preliminary usage scenarios are described here as well.

12.1.  Video Telephony According to Annex A of ITU-T Recommendation
       H.241

   H.323-based video telephony systems that use H.264 as an optional
   video compression scheme are required to support Annex A of H.241 [3]
   as a packetization scheme.  The packetization mechanism defined in
   this Annex is technically identical with a small subset of this
   specification.

   When a system operates according to Annex A of H.241, parameter set
   NAL units are sent in-band.  Only single NAL unit packets are used.
   Many such systems are not sending IDR pictures regularly, but only

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   when required by user interaction or by control protocol means, e.g.,
   when switching between video channels in a Multipoint Control Unit or
   for error recovery requested by feedback.

12.2.  Video Telephony, No Slice Data Partitioning, No NAL Unit
       Aggregation

   The RTP part of this scheme is implemented and tested (though not the
   control-protocol part; see below).

   In most real-world video telephony applications, picture parameters
   such as picture size or optional modes never change during the
   lifetime of a connection.  Therefore, all necessary parameter sets
   (usually only one) are sent as a side effect of the capability
   exchange/announcement process, e.g., according to the SDP syntax
   specified in Section 8.2 of this document.  As all necessary
   parameter set information is established before the RTP session
   starts, there is no need for sending any parameter set NAL units.
   Slice data partitioning is not used either.  Thus, the RTP packet
   stream basically consists of NAL units that carry single coded
   slices.

   The encoder chooses the size of coded slice NAL units so that they
   offer the best performance.  Often, this is done by adapting the
   coded slice size to the MTU size of the IP network.  For small
   picture sizes, this may result in a one-picture-per-one-packet
   strategy.  Intra refresh algorithms clean up the loss of packets and
   the resulting drift-related artifacts.

12.3.  Video Telephony, Interleaved Packetization Using NAL Unit
       Aggregation

   This scheme allows better error concealment and is used in
   H.263-based designs using RFC 4629 packetization [11].  It has been
   implemented, and good results were reported [13].

   The VCL encoder codes the source picture so that all macroblocks
   (MBs) of one MB line are assigned to one slice.  All slices with even
   MB row addresses are combined into one STAP, and all slices with odd
   MB row addresses are combined into another.  Those STAPs are
   transmitted as RTP packets.  The establishment of the parameter sets
   is performed as discussed above.

   Note that the use of STAPs is essential here, as the high number of
   individual slices (18 for a Common Intermediate Format (CIF) picture)
   would lead to unacceptably high IP/UDP/RTP header overhead (unless
   the source coding tool FMO is used, which is not assumed in this
   scenario).  Furthermore, some wireless video transmission systems,

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   such as H.324M and the IP-based video telephony specified in 3GPP,
   are likely to use relatively small transport packet size.  For
   example, a typical MTU size of H.223 AL3 SDU is around 100 bytes
   [17].  Coding individual slices according to this packetization
   scheme provides further advantage in communication between wired and
   wireless networks, as individual slices are likely to be smaller than
   the preferred maximum packet size of wireless systems.  Consequently,
   a gateway can convert the STAPs used in a wired network into several
   RTP packets with only one NAL unit, which are preferred in a wireless
   network, and vice versa.

12.4.  Video Telephony with Data Partitioning

   This scheme has been implemented and has been shown to offer good
   performance, especially at higher packet loss rates [13].

   Data partitioning is known to be useful only when some form of
   unequal error protection is available.  Normally, in single-session
   RTP environments, even error characteristics are assumed; that is,
   the packet loss probability of all packets of the session is the same
   statistically.  However, there are means to reduce the packet loss
   probability of individual packets in an RTP session.  A FEC packet
   according to RFC 5109 [18], for example, specifies which media
   packets are associated with the FEC packet.

   In all cases, the incurred overhead is substantial but is in the same
   order of magnitude as the number of bits that have otherwise been
   spent for intra information.  However, this mechanism does not add
   any delay to the system.

   Again, the complete parameter set establishment is performed through
   control protocol means.

12.5.  Video Telephony or Streaming with FUs and Forward Error
       Correction

   This scheme has been implemented and has been shown to provide good
   performance, especially at higher packet loss rates [19].

   The most efficient means to combat packet losses for scenarios where
   retransmissions are not applicable is forward error correction (FEC).
   Although application layer, end-to-end use of FEC is often less
   efficient than a FEC-based protection of individual links (especially
   when links of different characteristics are in the transmission
   path), application layer, end-to-end FEC is unavoidable in some
   scenarios.  RFC 5109 [18] provides means to use generic, application
   layer, end-to-end FEC in packet loss environments.  A binary forward
   error correcting code is generated by applying the XOR operation to

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   the bits at the same bit position in different packets.  The binary
   code can be specified by the parameters (n,k), in which k is the
   number of information packets used in the connection and n is the
   total number of packets generated for k information packets; that is,
   n-k parity packets are generated for k information packets.

   When a code is used with parameters (n,k) within the RFC 5109
   framework, the following properties are well known:

   a) If applied over one RTP packet, RFC 5109 provides only packet
      repetition.

   b) RFC 5109 is most bitrate efficient if XOR-connected packets have
      equal length.

   c) At the same packet loss probability p and for a fixed k, the
      greater the value of n, the smaller the residual error probability
      becomes.  For example, for a packet loss probability of 10%, k=1,
      and n=2, the residual error probability is about 1%, whereas for
      n=3, the residual error probability is about 0.1%.

   d) At the same packet loss probability p and for a fixed code rate
      k/n, the greater the value of n, the smaller the residual error
      probability becomes.  For example, at a packet loss probability of
      p=10%, k=1, and n=2, the residual error rate is about 1%, whereas
      for an extended Golay code with k=12 and n=24, the residual error
      rate is about 0.01%.

   For applying RFC 5109 in combination with H.264 baseline-coded video
   without using FUs, several options might be considered:

   1) The video encoder produces NAL units for which each video frame is
      coded in a single slice.  Applying FEC, one could use a simple
      code, e.g., (n=2, k=1).  That is, each NAL unit would basically
      just be repeated.  The disadvantage is obviously the bad code
      performance according to d), above, and the low flexibility, as
      only (n, k=1) codes can be used.

   2) The video encoder produces NAL units for which each video frame is
      encoded in one or more consecutive slices.  Applying FEC, one
      could use a better code, e.g., (n=24, k=12), over a sequence of
      NAL units.  Depending on the number of RTP packets per frame, a
      loss may introduce a significant delay, which is reduced when more
      RTP packets are used per frame.  Packets of completely different
      lengths might also be connected, which decreases bitrate

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      efficiency according to b), above.  However, with some care and
      for slices of 1 kb or larger, similar length (100-200 bytes
      difference) may be produced, which will not lower the bit
      efficiency catastrophically.

   3) The video encoder produces NAL units, for which a certain frame
      contains k slices of possibly almost equal length.  Then, applying
      FEC, a better code, e.g., (n=24, k=12), can be used over the
      sequence of NAL units for each frame.  The delay compared to that
      of 2), above, may be reduced, but several disadvantages are
      obvious.  First, the coding efficiency of the encoded video is
      lowered significantly, as slice-structured coding reduces intra-
      frame prediction and additional slice overhead is necessary.
      Second, pre-encoded content or, when operating over a gateway, the
      video is usually not appropriately coded with k slices such that
      FEC can be applied.  Finally, the encoding of video producing k
      slices of equal length is not straightforward and might require
      more than one encoding pass.

   Many of the mentioned disadvantages can be avoided by applying FUs in
   combination with FEC.  Each NAL unit can be split into any number of
   FUs of basically equal length; therefore, FEC, with a reasonable k
   and n, can be applied, even if the encoder made no effort to produce
   slices of equal length.  For example, a coded slice NAL unit
   containing an entire frame can be split to k FUs, and a parity check
   code (n=k+1, k) can be applied.  However, this has the disadvantage
   that unless all created fragments can be recovered, the whole slice
   will be lost.  Thus, a larger section is lost than would be if the
   frame had been split into several slices.

   The presented technique makes it possible to achieve good
   transmission error tolerance, even if no additional source coding
   layer redundancy (such as periodic intra frames) is present.
   Consequently, the same coded video sequence can be used to achieve
   the maximum compression efficiency and quality over error-free
   transmission and for transmission over error-prone networks.
   Furthermore, the technique allows the application of FEC to pre-
   encoded sequences without adding delay.  In this case, pre-encoded
   sequences that are not encoded for error-prone networks can still be
   transmitted almost reliably without adding extensive delays.  In
   addition, FUs of equal length result in a bitrate efficient use of
   RFC 5109.

   If the error probability depends on the length of the transmitted
   packet (e.g., in case of mobile transmission [15]), the benefits of
   applying FUs with FEC are even more obvious.  Basically, the
   flexibility of the size of FUs allows appropriate FEC to be applied
   for each NAL unit and unequal error protection of NAL units.

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   When FUs and FEC are used, the incurred overhead is substantial but
   is in the same order of magnitude as the number of bits that have to
   be spent for intra-coded macroblocks if no FEC is applied.  In [19],
   it was shown that the overall performance of the FEC-based approach
   enhanced quality when using the same error rate and same overall
   bitrate, including the overhead.

12.6.  Low Bitrate Streaming

   This scheme has been implemented with H.263 and non-standard RTP
   packetization and has given good results [20].  There is no technical
   reason why similarly good results could not be achievable with H.264.

   In today's Internet streaming, some of the offered bitrates are
   relatively low in order to allow terminals with dial-up modems to
   access the content.  In wired IP networks, relatively large packets,
   say 500 - 1500 bytes, are preferred to smaller and more frequently
   occurring packets in order to reduce network congestion.  Moreover,
   use of large packets decreases the amount of RTP/UDP/IP header
   overhead.  For low bitrate video, the use of large packets means that
   sometimes up to few pictures should be encapsulated in one packet.

   However, the loss of a packet including many coded pictures would
   have drastic consequences for visual quality, as there is practically
   no way to conceal the loss of an entire picture other than repeating
   the previous one.  One way to construct relatively large packets and
   maintain possibilities for successful loss concealment is to
   construct MTAPs that contain interleaved slices from several
   pictures.  An MTAP should not contain spatially adjacent slices from
   the same picture or spatially overlapping slices from any picture.
   If a packet is lost, it is likely that a lost slice is surrounded by
   spatially adjacent slices of the same picture and spatially
   corresponding slices of the temporally previous and succeeding
   pictures.  Consequently, concealment of the lost slice is likely to
   be relatively successful.

12.7.  Robust Packet Scheduling in Video Streaming

   Robust packet scheduling has been implemented with MPEG-4 Part 2 and
   simulated in a wireless streaming environment [21].  There is no
   technical reason why similar or better results could not be
   achievable with H.264.

   Streaming clients typically have a receiver buffer that is capable of
   storing a relatively large amount of data.  Initially, when a
   streaming session is established, a client does not start playing the
   stream back immediately.  Rather, it typically buffers the incoming
   data for a few seconds.  This buffering helps maintain continuous

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   playback, as, in case of occasional increased transmission delays or
   network throughput drops, the client can decode and play buffered
   data.  Otherwise, without initial buffering, the client has to freeze
   the display, stop decoding, and wait for incoming data.  The
   buffering is also necessary for either automatic or selective
   retransmission in any protocol level.  If any part of a picture is
   lost, a retransmission mechanism may be used to resend the lost data.
   If the retransmitted data is received before its scheduled decoding
   or playback time, the loss is recovered perfectly.  Coded pictures
   can be ranked according to their importance in the subjective quality
   of the decoded sequence.  For example, non-reference pictures, such
   as conventional B pictures, are subjectively least important, as
   their absence does not affect decoding of any other pictures.  In
   addition to non-reference pictures, the ITU-T H.264 | ISO/IEC
   14496-10 standard includes a temporal scalability method called sub-
   sequences [22].  Subjective ranking can also be made on coded slice
   data partition or slice group basis.  Coded slices and coded slice
   data partitions that are subjectively the most important can be sent
   earlier than their decoding order indicates, whereas coded slices and
   coded slice data partitions that are subjectively the least important
   can be sent later than their natural coding order indicates.
   Consequently, any retransmitted parts of the most important slices
   and coded slice data partitions are more likely to be received before
   their scheduled decoding or playback time compared to the least
   important slices and slice data partitions.

13.  Informative Appendix: Rationale for Decoding Order Number

13.1.  Introduction

   The Decoding Order Number (DON) concept was introduced mainly to
   enable efficient multi-picture slice interleaving (see Section 12.6)
   and robust packet scheduling (see Section 12.7).  In both of these
   applications, NAL units are transmitted out of decoding order.  DON
   indicates the decoding order of NAL units and should be used in the
   receiver to recover the decoding order.  Example use cases for
   efficient multi-picture slice interleaving and for robust packet
   scheduling are given in Sections 13.2 and 13.3, respectively.
   Section 13.4 describes the benefits of the DON concept in error
   resiliency achieved by redundant coded pictures.  Section 13.5
   summarizes considered alternatives to DON and justifies why DON was
   chosen for this RTP payload specification.

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13.2.  Example of Multi-Picture Slice Interleaving

   An example of multi-picture slice interleaving follows.  A subset of
   a coded video sequence is depicted below in output order.  R denotes
   a reference picture, N denotes a non-reference picture, and the
   number indicates a relative output time.

      ... R1 N2 R3 N4 R5 ...

   The decoding order of these pictures from left to right is as
   follows:

      ... R1 R3 N2 R5 N4 ...

   The NAL units of pictures R1, R3, N2, R5, and N4 are marked with a
   DON equal to 1, 2, 3, 4, and 5, respectively.

   Each reference picture consists of three slice groups that are
   scattered as follows (a number denotes the slice group number for
   each macroblock in a Quarter Common Intermediate Format (QCIF)
   frame):

      0 1 2 0 1 2 0 1 2 0 1
      2 0 1 2 0 1 2 0 1 2 0
      1 2 0 1 2 0 1 2 0 1 2
      0 1 2 0 1 2 0 1 2 0 1
      2 0 1 2 0 1 2 0 1 2 0
      1 2 0 1 2 0 1 2 0 1 2
      0 1 2 0 1 2 0 1 2 0 1
      2 0 1 2 0 1 2 0 1 2 0
      1 2 0 1 2 0 1 2 0 1 2

   For the sake of simplicity, we assume that all the macroblocks of a
   slice group are included in one slice.  Three MTAPs are constructed
   from three consecutive reference pictures so that each MTAP contains
   three aggregation units, each of which contains all the macroblocks
   from one slice group.  The first MTAP contains slice group 0 of
   picture R1, slice group 1 of picture R3, and slice group 2 of picture
   R5.  The second MTAP contains slice group 1 of picture R1, slice
   group 2 of picture R3, and slice group 0 of picture R5.  The third
   MTAP contains slice group 2 of picture R1, slice group 0 of picture
   R3, and slice group 1 of picture R5.  Each non-reference picture is
   encapsulated into an STAP-B.

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   Consequently, the transmission order of NAL units is the following:

      R1, slice group 0, DON 1, carried in MTAP,RTP SN: N
      R3, slice group 1, DON 2, carried in MTAP,RTP SN: N
      R5, slice group 2, DON 4, carried in MTAP,RTP SN: N
      R1, slice group 1, DON 1, carried in MTAP,RTP SN: N+1
      R3, slice group 2, DON 2, carried in MTAP,RTP SN: N+1
      R5, slice group 0, DON 4, carried in MTAP,RTP SN: N+1
      R1, slice group 2, DON 1, carried in MTAP,RTP SN: N+2
      R3, slice group 1, DON 2, carried in MTAP,RTP SN: N+2
      R5, slice group 0, DON 4, carried in MTAP,RTP SN: N+2
      N2, DON 3, carried in STAP-B, RTP SN: N+3
      N4, DON 5, carried in STAP-B, RTP SN: N+4

   The receiver is able to organize the NAL units back in decoding order
   based on the value of DON associated with each NAL unit.

   If one of the MTAPs is lost, the spatially adjacent and temporally
   co-located macroblocks are received and can be used to conceal the
   loss efficiently.  If one of the STAPs is lost, the effect of the
   loss does not propagate temporally.

13.3.  Example of Robust Packet Scheduling

   An example of robust packet scheduling follows.  The communication
   system used in the example consists of the following components in
   the order that the video is processed from source to sink:

   o camera and capturing
   o pre-encoding buffer
   o encoder
   o encoded picture buffer
   o transmitter
   o transmission channel
   o receiver
   o receiver buffer
   o decoder
   o decoded picture buffer
   o display

   The video communication system used in this example operates as
   follows.  Note that processing of the video stream happens gradually
   and at the same time in all components of the system.  The source
   video sequence is shot and captured to a pre-encoding buffer.  The
   pre-encoding buffer can be used to order pictures from sampling order
   to encoding order or to analyze multiple uncompressed frames for
   bitrate control purposes, for example.  In some cases, the pre-
   encoding buffer may not exist; instead, the sampled pictures are

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   encoded right away.  The encoder encodes pictures from the pre-
   encoding buffer and stores the output (i.e., coded pictures) to the
   encoded picture buffer.  The transmitter encapsulates the coded
   pictures from the encoded picture buffer to transmission packets and
   sends them to a receiver through a transmission channel.  The
   receiver stores the received packets to the receiver buffer.  The
   receiver buffering process typically includes buffering for
   transmission delay jitter.  The receiver buffer can also be used to
   recover correct decoding order of coded data.  The decoder reads
   coded data from the receiver buffer and produces decoded pictures as
   output into the decoded picture buffer.  The decoded picture buffer
   is used to recover the output (or display) order of pictures.
   Finally, pictures are displayed.

   In the following example figures, I denotes an IDR picture, R denotes
   a reference picture, N denotes a non-reference picture, and the
   number after I, R, or N indicates the sampling time relative to the
   previous IDR picture in decoding order.  Values below the sequence of
   pictures indicate scaled system clock timestamps.  The system clock
   is initialized arbitrarily in this example, and time runs from left
   to right.  Each I, R, and N picture is mapped into the same timeline
   compared to the previous processing step, if any, assuming that
   encoding, transmission, and decoding take no time.  Thus, events
   happening at the same time are located in the same column throughout
   all example figures.

   A subset of a sequence of coded pictures is depicted below in
   sampling order.

       ...  N58 N59 I00 N01 N02 R03 N04 N05 R06 ... N58 N59 I00 N01 ...
       ... --|---|---|---|---|---|---|---|---|- ... -|---|---|---|- ...
       ...  58  59  60  61  62  63  64  65  66  ... 128 129 130 131 ...

       Figure 16.  Sequence of pictures in sampling order

   The sampled pictures are buffered in the pre-encoding buffer to
   arrange them in encoding order.  In this example, we assume that the
   non-reference pictures are predicted from both the previous and the
   next reference picture in output order, except for the non-reference
   pictures immediately preceding an IDR picture, which are predicted
   only from the previous reference picture in output order.  Thus, the
   pre-encoding buffer has to contain at least two pictures, and the
   buffering causes a delay of two picture intervals.  The output of the
   pre-encoding buffering process and the encoding (and decoding) order
   of the pictures are as follows:

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       ... N58 N59 I00 R03 N01 N02 R06 N04 N05 ...
       ... -|---|---|---|---|---|---|---|---|- ...
       ... 60  61  62  63  64  65  66  67  68  ...

       Figure 17.  Reordered pictures in the pre-encoding buffer

   The encoder or the transmitter can set the value of DON for each
   picture to a value of DON for the previous picture in decoding order
   plus one.

   For the sake of simplicity, let us assume that:

   o  the frame rate of the sequence is constant,
   o  each picture consists of only one slice,
   o  each slice is encapsulated in a single NAL unit packet,
   o  there is no transmission delay, and
   o  pictures are transmitted at constant intervals (that is, 1 /
      (frame rate)).

   When pictures are transmitted in decoding order, they are received as
   follows:

       ... N58 N59 I00 R03 N01 N02 R06 N04 N05 ...
       ... -|---|---|---|---|---|---|---|---|- ...
       ... 60  61  62  63  64  65  66  67  68  ...

       Figure 18.  Received pictures in decoding order

   The OPTIONAL sprop-interleaving-depth media type parameter is set to
   0, as the transmission (or reception) order is identical to the
   decoding order.

   Initially, the decoder has to buffer for one picture interval in its
   decoded picture buffer to organize pictures from decoding order to
   output order, as depicted below:

       ... N58 N59 I00 N01 N02 R03 N04 N05 R06 ...
       ... -|---|---|---|---|---|---|---|---|- ...
       ... 61  62  63  64  65  66  67  68  69  ...

       Figure 19.  Output order

   The amount of required initial buffering in the decoded picture
   buffer can be signaled in the buffering period SEI message or with
   the num_reorder_frames syntax element of H.264 video usability
   information.  num_reorder_frames indicates the maximum number of
   frames, complementary field pairs, or non-paired fields that precede
   any frame, complementary field pair, or non-paired field in the

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   sequence in decoding order and that follow it in output order.  For
   the sake of simplicity, we assume that num_reorder_frames is used to
   indicate the initial buffer in the decoded picture buffer.  In this
   example, num_reorder_frames is equal to 1.

   It can be observed that if the IDR picture I00 is lost during
   transmission and a retransmission request is issued when the value of
   the system clock is 62, there is one picture interval of time (until
   the system clock reaches timestamp 63) to receive the retransmitted
   IDR picture I00.

   Let us then assume that IDR pictures are transmitted two frame
   intervals earlier than their decoding position; that is, the pictures
   are transmitted as follows:

       ...  I00 N58 N59 R03 N01 N02 R06 N04 N05 ...
       ... --|---|---|---|---|---|---|---|---|- ...
       ...  62  63  64  65  66  67  68  69  70  ...

       Figure 20.  Interleaving: Early IDR pictures in sending order

   The OPTIONAL sprop-interleaving-depth media type parameter is set
   equal to 1 according to its definition.  (The value of sprop-
   interleaving-depth in this example can be derived as follows: picture
   I00 is the only picture preceding picture N58 or N59 in transmission
   order and following it in decoding order.  Except for pictures I00,
   N58, and N59, the transmission order is the same as the decoding
   order of pictures.  As a coded picture is encapsulated into exactly
   one NAL unit, the value of sprop-interleaving-depth is equal to the
   maximum number of pictures preceding any picture in transmission
   order and following the picture in decoding order).

   The receiver buffering process contains two pictures at a time
   according to the value of the sprop-interleaving-depth parameter and
   orders pictures from the reception order to the correct decoding
   order based on the value of DON associated with each picture.  The
   output of the receiver buffering process is as follows:

       ... N58 N59 I00 R03 N01 N02 R06 N04 N05 ...
       ... -|---|---|---|---|---|---|---|---|- ...
       ... 63  64  65  66  67  68  69  70  71  ...

       Figure 21.  Interleaving: Receiver buffer

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   Again, an initial buffering delay of one picture interval is needed
   to organize pictures from decoding order to output order, as depicted
   below:

        ... N58 N59 I00 N01 N02 R03 N04 N05 ...
        ... -|---|---|---|---|---|---|---|- ...
        ... 64  65  66  67  68  69  70  71  ...

        Figure 22.  Interleaving: Receiver buffer after reordering

   Note that the maximum delay that IDR pictures can undergo during
   transmission, including possible application, transport, or link
   layer retransmission, is equal to three picture intervals.  Thus, the
   loss resiliency of IDR pictures is improved in systems supporting
   retransmission compared to the case in which pictures are transmitted
   in their decoding order.

13.4.  Robust Transmission Scheduling of Redundant Coded Slices

   A redundant coded picture is a coded representation of a picture or a
   part of a picture that is not used in the decoding process if the
   corresponding primary coded picture is correctly decoded.  There
   should be no noticeable difference between any area of the decoded
   primary picture and a corresponding area that would result from
   application of the H.264 decoding process for any redundant picture
   in the same access unit.  A redundant coded slice is a coded slice
   that is a part of a redundant coded picture.

   Redundant coded pictures can be used to provide unequal error
   protection in error-prone video transmission.  If a primary coded
   representation of a picture is decoded incorrectly, a corresponding
   redundant coded picture can be decoded.  Examples of applications and
   coding techniques using the redundant codec picture feature include
   the video redundancy coding [23] and the protection of "key pictures"
   in multicast streaming [24].

   One property of many error-prone video communications systems is that
   transmission errors are often bursty.  Therefore, they may affect
   more than one consecutive transmission packet in transmission order.
   In low bitrate video communication, it is relatively common for an
   entire coded picture to be encapsulated into one transmission packet.
   Consequently, a primary coded picture and the corresponding redundant
   coded pictures may be transmitted in consecutive packets in
   transmission order.  To make the transmission scheme more tolerant of
   bursty transmission errors, it is beneficial to transmit the primary
   coded picture and redundant coded picture separated by more than a
   single packet.  The DON concept enables this.

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13.5.  Remarks on Other Design Possibilities

   The slice header syntax structure of the H.264 coding standard
   contains the frame_num syntax element that can indicate the decoding
   order of coded frames.  However, the usage of the frame_num syntax
   element is not feasible or desirable to recover the decoding order,
   due to the following reasons:

   o  The receiver is required to parse at least one slice header per
      coded picture (before passing the coded data to the decoder).

   o  Coded slices from multiple coded video sequences cannot be
      interleaved, as the frame number syntax element is reset to 0 in
      each IDR picture.

   o  The coded fields of a complementary field pair share the same
      value of the frame_num syntax element.  Thus, the decoding order
      of the coded fields of a complementary field pair cannot be
      recovered based on the frame_num syntax element or any other
      syntax element of the H.264 coding syntax.

   The RTP payload format for transport of MPEG-4 elementary streams
   [25] enables interleaving of access units and transmission of
   multiple access units in the same RTP packet.  An access unit is
   specified in the H.264 coding standard to comprise all NAL units
   associated with a primary coded picture according to Subclause
   7.4.1.2 of [1].  Consequently, slices of different pictures cannot be
   interleaved, and the multi-picture slice interleaving technique (see
   Section 12.6) for improved error resilience cannot be used.

14.  Changes from RFC 3984

   Following is the list of technical changes (including bug fixes) from
   RFC 3984.  Besides this list of technical changes, numerous editorial
   changes have been made, but not documented in this section.  Note
   that Section 8.2.2 is where much of the important changes in this
   memo occurs and deserves particular attention.

   1)  In Sections 5.4, 5.5, 6.2, 6.3, and 6.4, removed that the
       packetization mode in use may be signaled by external means.

   2)  In Section 7.2.2, changed the sentence

       There are N VCL NAL units in the de-interleaving buffer.

       to

       There are N or more VCL NAL units in the de-interleaving buffer.

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   3)  In Section 8.1, the semantics of sprop-init-buf-time (paragraph
       2), changed the sentence

       The parameter is the maximum value of (transmission time of a NAL
       unit - decoding time of the NAL unit), assuming reliable and
       instantaneous transmission, the same timeline for transmission
       and decoding, and that decoding starts when the first packet
       arrives.

       to

       The parameter is the maximum value of (decoding time of the NAL
       unit - transmission time of a NAL unit), assuming reliable and
       instantaneous transmission, the same timeline for transmission
       and decoding, and that decoding starts when the first packet
       arrives.

   4)  Added media type parameters max-smbps, sprop-level-parameter-
       sets, use-level-src-parameter-sets, in-band-parameter-sets, sar-
       understood, and sar-supported.

   5)  In Section 8.1, removed the specification of parameter-add.
       Other descriptions of parameter-add (in Sections 8.2 and 8.4)
       were also removed.

   6)  In Section 8.1, added a constraint to sprop-parameter-sets such
       that it can only contain parameter sets for the same profile and
       level as indicated by profile-level-id.

   7)  In Section 8.2.1, added that sprop-parameter-sets and sprop-
       level-parameter-sets may be either included in the "a=fmtp" line
       of SDP or conveyed using the "fmtp" source attribute as specified
       in Section 6.3 of [9].

   8)  In Section 8.2.2, removed sprop-deint-buf-req from being part of
       the media format configuration in usage with the SDP Offer/Answer
       model.

   9)  In Section 8.2.2, made it clear that level is downgradable in the
       SDP Offer/Answer model, i.e., the use of the level part of
       profile-level-id does not need to be symmetric (the level
       included in the answer can be lower than or equal to the level
       included in the offer).

   10) In Section 8.2.2, removed that the capability parameters may be
       used to declare encoding capabilities.

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   11) In Section 8.2.2, added rules on how to use sprop-parameter-sets
       and sprop-level-parameter-sets for out-of-band transport of
       parameter sets, with or without level downgrading.

   12) In Section 8.2.2, clarified the rules of using the media type
       parameters with SDP Offer/Answer for multicast.

   13) In Section 8.2.2, completed and corrected the list of how
       different media type parameters shall be interpreted in the
       different combinations of offer or answer and direction
       attribute.

   14) In Section 8.4, changed the text such that both out-of-band and
       in-band transport of parameter sets are allowed, and neither is
       recommended or required.

   15) Added Section 8.5 (informative) providing example methods for
       decoder refresh to handle parameter set losses.

   16) Added media type parameters max-recv-level and level-asymmetry-
       allowed and adjusted associated text and examples for level
       upgrade and asymmetry.

15.  Backward Compatibility to RFC 3984

   The current document is a revision of RFC 3984 and obsoletes it.  The
   technical changes relative to RFC 3984 are listed in Section 14.
   This section addresses the backward compatibility issues.

   It should be noted that for the majority of cases, there will be no
   compatibility issues for legacy implementations per RFC 3984 and new
   implementations per this document to interwork.  Compatibility issues
   may only occur when both of the following conditions are true: 1)
   legacy implementations and new implementations are interworking, and
   2) parameter sets are transported out-of-band.  When such
   compatibility issues occur, it is easy to debug and find the reason
   for the incompatibility using the following analyses.

   Items 1, 2, 3, 7, 9, 10, 12, and 13 are bug-fix types of changes and
   do not incur any backward compatibility issues.

   Item 4 (addition of six new media type parameters) does not incur any
   backward compatibility issues for SDP Offer/Answer-based
   applications, as legacy RFC 3984 receivers ignore these parameters,
   and it is fine for legacy RFC 3984 senders not to use these
   parameters as they are optional.  However, there is a backward
   compatibility issue for declarative-usage-based applications (only
   for the parameter sprop-level-parameter-sets as the other five

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   parameters are not usable in declarative usage).  For example,
   declarative-usage-based applications using RTSP and SAP have a
   backward compatibility issue because the SDP receiver per RFC 3984
   cannot accept a session for which the SDP includes an unrecognized
   parameter.  Therefore, the RTSP or SAP server may have to prepare two
   sets of streams, one for legacy RFC 3984 receivers and one for
   receivers according to this memo.

   Items 5, 6, and 11 are related to out-of-band transport of parameter
   sets.  There are following backward compatibility issues.

   1)  When a legacy sender per RFC 3984 includes parameter sets for a
       level different than the default level indicated by profile-
       level-id to sprop-parameter-sets, the parameter value of sprop-
       parameter-sets is invalid to the receiver per this memo;
       therefore, the session may be rejected.

   2)  In SDP Offer/Answer between a legacy offerer per RFC 3984 and an
       answerer per this memo, when the answerer includes in the answer
       parameter sets that are not a superset of the parameter sets
       included in the offer, the parameter value of sprop-parameter-
       sets is invalid to the offerer, and the session may not be
       initiated properly (related to change item 11).

   3)  When one endpoint A per this memo includes in-band-parameter-sets
       equal to 1, the other side B per RFC 3984 does not understand
       that it must transmit parameter sets in-band, and B may still
       exclude parameter sets in the in-band stream it is sending.
       Consequently, endpoint A cannot decode the stream it receives.

   Item 7 (allowance of conveying sprop-parameter-sets and sprop-level-
   parameter-sets using the "fmtp" source attribute as specified in
   Section 6.3 of [9]) is similar to item 4.  It does not incur any
   backward compatibility issues for SDP Offer/Answer-based
   applications, as legacy RFC 3984 receivers ignore the "fmtp" source
   attribute, and it is fine for legacy RFC 3984 senders not to use the
   "fmtp" source attribute as it is optional.  However, there is a
   backward compatibility issue for SDP declarative-usage-based
   applications, e.g., those using RTSP and SAP, because the SDP
   receiver per RFC 3984 cannot accept a session for which the SDP
   includes an unrecognized parameter (i.e., the "fmtp" source
   attribute).  Therefore, the RTSP or SAP server may have to prepare
   two sets of streams, one for legacy RFC 3984 receivers and one for
   receivers according to this memo.

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   Item 14 does not incur any backward compatibility issues, as out-of-
   band transport of parameter sets is still allowed.

   Item 15 does not incur any backward compatibility issues, as the
   added Section 8.5 is informative.

   Item 16 does not create any backward compatibility issues as the
   handling of the default level is the same if either end is RFC 3984
   compliant, and, furthermore, RFC-3984-compliant ends would simply
   ignore the new media type parameters, if present.

16.  Acknowledgements

   Stephan Wenger, Miska Hannuksela, Thomas Stockhammer, Magnus
   Westerlund, and David Singer are thanked as the authors of RFC 3984.
   Dave Lindbergh, Philippe Gentric, Gonzalo Camarillo, Gary Sullivan,
   Joerg Ott, and Colin Perkins are thanked for careful review during
   the development of RFC 3984. Stephen Botzko, Magnus Westerlund, Alex
   Eleftheriadis, Thomas Schierl, Tom Taylor, Ali Begen, Aaron Wells,
   Stuart Taylor, Robert Sparks, Dan Romascanu, and Niclas Comstedt are
   thanked for their valuable comments and input during the development
   of this memo.

17.  References

17.1.  Normative References

   [1]   ITU-T Recommendation H.264, "Advanced video coding for generic
         audiovisual services", March 2010.

   [2]   ISO/IEC International Standard 14496-10:2008.

   [3]   ITU-T Recommendation H.241, "Extended video procedures and
         control signals for H.300-series terminals", May 2006.

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

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

   [6]   Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
         Description Protocol", RFC 4566, July 2006.

   [7]   Josefsson, S., "The Base16, Base32, and Base64 Data Encodings",
         RFC 4648, October 2006.

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   [8]   Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
         Session Description Protocol (SDP)", RFC 3264, June 2002.

   [9]   Lennox, J., Ott, J., and T. Schierl, "Source-Specific Media
         Attributes in the Session Description Protocol (SDP)", RFC
         5576, June 2009.

17.2.  Informative References

   [10]  Luthra, A., Sullivan, G.J., and T. Wiegand (eds.),
         "Introduction to the special issue on the H.264/AVC video
         coding standard", IEEE Transactions on Circuits and Systems for
         Video Technology, Vol. 13, No. 7, July 2003.

   [11]  Ott, J., Bormann, C., Sullivan, G., Wenger, S., and R. Even,
         Ed., "RTP Payload Format for ITU-T Rec. H.263 Video", RFC 4629,
         January 2007.

   [12]  ISO/IEC International Standard 14496-2:2004.

   [13]  Wenger, S., "H.264/AVC over IP", IEEE Transaction on Circuits
         and Systems for Video Technology, Vol. 13, No. 7, July 2003.

   [14]  Wenger, S., "H.26L over IP: The IP-Network Adaptation Layer",
         Proceedings Packet Video Workshop, April 2002.

   [15]  Stockhammer, T., Hannuksela, M.M., and S. Wenger, "H.26L/JVT
         Coding Network Abstraction Layer and IP-Based Transport", IEEE
         International Conference on Image Processing (ICIP 2002),
         Rochester, NY, September 2002.

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

   [17]  ITU-T Recommendation H.223, "Multiplexing protocol for low bit
         rate multimedia communication", July 2001.

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

   [19]  Stockhammer, T., Wiegand, T., Oelbaum, T., and F. Obermeier,
         "Video Coding and Transport Layer Techniques for H.264/AVC-
         Based Transmission over Packet-Lossy Networks", IEEE
         International Conference on Image Processing (ICIP 2003),
         Barcelona, Spain, September 2003.

   [20]  Varsa, V. and M. Karczewicz, "Slice interleaving in compressed
         video packetization", Packet Video Workshop 2000.

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   [21]  Kang, S.H. and A. Zakhor, "Packet scheduling algorithm for
         wireless video streaming", Packet Video Workshop 2002.

   [22]  Hannuksela, M.M., "Enhanced Concept of GOP", JVT-B042,
         available http://ftp3.itu.int/av-arch/video-site/0201_Gen/JVT-
         B042.doc, January 2002.

   [23]  Wenger, S., "Video Redundancy Coding in H.263+", 1997
         International Workshop on Audio-Visual Services over Packet
         Networks, September 1997.

   [24]  Wang, Y.-K., Hannuksela, M.M., and M. Gabbouj, "Error Resilient
         Video Coding Using Unequally Protected Key Pictures", in Proc.
         International Workshop VLBV03, September 2003.

   [25]  van der Meer, J., Mackie, D., Swaminathan, V., Singer, D., and
         P. Gentric, "RTP Payload Format for Transport of MPEG-4
         Elementary Streams", RFC 3640, November 2003.

   [26]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
         Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
         3711, March 2004.

   [27]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
         Protocol (RTSP)", RFC 2326, April 1998.

   [28]  Handley, M., Perkins, C., and E. Whelan, "Session Announcement
         Protocol", RFC 2974, October 2000.

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

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

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Authors' Addresses

   Ye-Kui Wang
   Huawei Technologies
   400 Crossing Blvd, 2nd Floor
   Bridgewater, NJ 08807
   USA

   Phone: +1-908-541-3518
   EMail: yekui.wang@huawei.com


   Roni Even
   Huawei Technologies
   14 David Hamelech
   Tel Aviv 64953
   Israel

   Phone: +972-545481099
   EMail: even.roni@huawei.com


   Tom Kristensen
   TANDBERG
   Philip Pedersens vei 22
   N-1366 Lysaker
   Norway

   Phone: +47 67125125
   EMail: tom.kristensen@tandberg.com, tomkri@ifi.uio.no


   Randell Jesup
   WorldGate Communications
   3800 Horizon Blvd, Suite #103
   Trevose, PA 19053-4947
   USA

   Phone: +1-215-354-5166
   EMail: rjesup@wgate.com, randell_ietf@jesup.org