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RFC 3984

RTP Payload Format for H.264 Video

Pages: 83
Obsoleted by:  6184
Part 3 of 3 – Pages 62 to 83
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9. Security Considerations

RTP packets using the payload format defined in this specification are subject to the security considerations discussed in the RTP specification [4], 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 applied end-to-end, any encryption needs to be performed after compression.
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   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.  And
   in the case of confidentiality protection it will even be prevented
   from performing discarding of packets in a media aware way.  To allow
   any MANE to perform its operations, it will be required to be a
   trusted entity which is included in the security context

10. Congestion Control

Congestion control for RTP SHALL be used in accordance with RFC 3550 [4], and with any applicable RTP profile; e.g., RFC 3551 [16]. An additional requirement if best-effort service is being used is: 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 bit rate 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 bit rates, 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 re-start 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 Consideration

IANA has registered one new MIME type; see section 8.1.
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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 ITU-T Recommendation H.241 Annex A

H.323-based video telephony systems that use H.264 as an optional video compression scheme are required to support H.241 Annex A [15] 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 H.241 Annex A, 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 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
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   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 2429 packetization [10]. It has been implemented, and good results were reported [12]. 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 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 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, 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 [12]. 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; i.e., 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 2733 [18], for example, specifies which media packets are associated with the FEC packet.
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   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 an 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 2733 [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 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; i.e., n-k parity packets are generated for k information packets. When a code is used with parameters (n,k) within the RFC 2733 framework, the following properties are well known: a) If applied over one RTP packet, RFC 2733 provides only packet repetition. b) RFC 2733 is most bit rate 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 is, 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 is, 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
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      for an extended Golay code with k=12 and n=24, the residual error
      rate is about 0.01%.

   For applying RFC 2733 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
      length might also be connected, which decreases bit rate
      efficiency according to b), above.  However, with some care and
      for slices of 1kb 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
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   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 bit rate efficient use of
   RFC 2733.

   If the error probability depends on the length of the transmitted
   packet (e.g., in case of mobile transmission [14]), 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.

   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 bit
   rate, including the overhead.

12.6. Low Bit-Rate 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 bit rates 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 bit-rate video, the use of large packets means that sometimes up to few pictures should be encapsulated in one packet.
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   However, loss of a packet including many coded pictures would have
   drastic consequences for visual quality, as there is practically no
   other way to conceal a loss of an entire picture than to repeat 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 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
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   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 to this RTP payload specification.

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.
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   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 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.

   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.
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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 the 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 bit rate control purposes, for example. In some cases, the pre-encoding buffer may not exist; instead, the sampled pictures are 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
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   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:

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

      Figure 17.  Re-ordered 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
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   When pictures are transmitted in decoding order, they are received as

                ... 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 MIME type parameter is set to
   0, as the transmission (or reception) order is identical to the
   decoding order.

   The decoder has to buffer for one picture interval initially 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
   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.
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   Let us then assume that IDR pictures are transmitted two frame
   intervals earlier than their decoding position; i.e., 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 MIME 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

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

                                ... 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
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   loss resiliency of IDR pictures is improved in systems supporting
   retransmission compared to the case in which pictures were
   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 packets in transmission order. In low bit-rate video communication, it is relatively common that an entire coded picture can 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.

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).
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   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 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. Acknowledgements

The authors thank Roni Even, Dave Lindbergh, Philippe Gentric, Gonzalo Camarillo, Gary Sullivan, Joerg Ott, and Colin Perkins for careful review.

15. References

15.1. Normative References

[1] ITU-T Recommendation H.264, "Advanced video coding for generic audiovisual services", May 2003. [2] ISO/IEC International Standard 14496-10:2003. [3] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [4] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. [5] Handley, M. and V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998. [6] Josefsson, S., "The Base16, Base32, and Base64 Data Encodings", RFC 3548, July 2003.
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   [7]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
        Session Description Protocol (SDP)", RFC 3264, June 2002.

15.2. Informative References

[8] "Draft ITU-T Recommendation and Final Draft International Standard of Joint Video Specification (ITU-T Rec. H.264 | ISO/IEC 14496-10 AVC)", available from arch/jvt-site/2003_03_Pattaya/, May 2003. [9] Luthra, A., Sullivan, G.J., and T. Wiegand (eds.), Special Issue on H.264/AVC. IEEE Transactions on Circuits and Systems on Video Technology, July 2003. [10] Bormann, C., Cline, L., Deisher, G., Gardos, T., Maciocco, C., Newell, D., Ott, J., Sullivan, G., Wenger, S., and C. Zhu, "RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)", RFC 2429, October 1998. [11] ISO/IEC IS 14496-2. [12] Wenger, S., "H.26L over IP", IEEE Transaction on Circuits and Systems for Video technology, Vol. 13, No. 7, July 2003. [13] Wenger, S., "H.26L over IP: The IP Network Adaptation Layer", Proceedings Packet Video Workshop 02, April 2002. [14] Stockhammer, T., Hannuksela, M.M., and S. Wenger, "H.26L/JVT Coding Network Abstraction Layer and IP-based Transport" in Proc. ICIP 2002, Rochester, NY, September 2002. [15] ITU-T Recommendation H.241, "Extended video procedures and control signals for H.300 series terminals", 2004. [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] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for Generic Forward Error Correction", RFC 2733, December 1999. [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.
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   [20] Varsa, V. and M. Karczewicz, "Slice interleaving in compressed
        video packetization", Packet Video Workshop 2000.

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

   [22] Hannuksela, M.M., "Enhanced concept of GOP", JVT-B042, available,
        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] ISO/IEC 14496-15: "Information technology - Coding of audio-
        visual objects - Part 15: Advanced Video Coding (AVC) file

   [30] Castagno, R. and D. Singer, "MIME Type Registrations for 3rd
        Generation Partnership Project (3GPP) Multimedia files", RFC
        3839, July 2004.
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Authors' Addresses

Stephan Wenger TU Berlin / Teles AG Franklinstr. 28-29 D-10587 Berlin Germany Phone: +49-172-300-0813 EMail: Miska M. Hannuksela Nokia Corporation P.O. Box 100 33721 Tampere Finland Phone: +358-7180-73151 EMail: Thomas Stockhammer Nomor Research D-83346 Bergen Germany Phone: +49-8662-419407 EMail: Magnus Westerlund Multimedia Technologies Ericsson Research EAB/TVA/A Ericsson AB Torshamsgatan 23 SE-164 80 Stockholm Sweden Phone: +46-8-7190000 EMail:
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   David Singer
   QuickTime Engineering
   1 Infinite Loop MS 302-3MT
   CA 95014

   Phone +1 408 974-3162
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