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Versions: 00 01 02 RFC 2343

  Internet Engineering Task Force                         M. Reha Civanlar
  INTERNET-DRAFT                                             Glenn L. Cash
  File: draft-civanlar-bmpeg-02.txt                       Barry G. Haskell

                                                        AT&T Labs-Research

                                                           November, 1997


                   RTP Payload Format for Bundled MPEG


                           Status of this Memo

  This document is an Internet-Draft.  Internet-Drafts are working
  documents of the Internet Engineering Task Force (IETF), its areas,
  and its working groups.  Note that other groups may also distribute
  working documents as Internet-Drafts.

  Internet-Drafts are draft documents valid for a maximum of six months
  and may be updated, replaced, or obsoleted by other documents at any
  time.  It is inappropriate to use Internet- Drafts as reference
  material or to cite them other than as ``work in progress.''

  To learn the current status of any Internet-Draft, please check the
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  ftp.isi.edu (US West Coast).

  Distribution of this memo is unlimited.


                                 Abstract

  This document describes a payload type for bundled, MPEG-2 encoded
  video and audio data that may be used with RTP, version 2. Bundling
  has some advantages for this payload type particularly when it is used
  for video-on-demand applications. This payload type may be used when
  its advantages are important enough to sacrifice the modularity of
  having separate audio and video streams.

  A technique to improve packet loss resilience based on ''out-of-band''
  transmission of MPEG-2 specific, vital information is described as an
  Appendix.

  A section on security considerations for this payload type is added.




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  1. Introduction

  This document describes a bundled packetization scheme for MPEG-2
  encoded audio and video streams using the Real-time Transport Protocol
  (RTP), version 2 [1].

  The MPEG-2 International standard consists of three layers: audio,
  video and systems [2]. The audio and the video layers define the
  syntax and semantics of the corresponding "elementary streams." The
  systems layer supports synchronization and interleaving of multiple
  compressed streams, buffer initialization and management, and time
  identification.  RFC 2038 [3] describes packetization techniques to
  transport individual audio and video elementary streams as well as the
  transport stream, which is defined at the system layer, using the RTP.

  The bundled packetization scheme is needed because it has several
  advantages over other schemes for some important applications
  including video-on-demand (VOD) where, audio and video are always used
  together.  Its advantages over independent packetization of audio and
  video are:

    1. Uses a single port per "program" (i.e. bundled A/V).
    This may increase the number of streams that can be served
    e.g., from a VOD server. Also, it eliminates the performance
    hit when two ports are used for the separate audio and video
    messages on the client side.

    2. Provides implicit synchronization of audio and video.
    This is particularly convenient when the A/V data is stored
    in an interleaved format at the server.

    3. Reduces the header overhead. Since using large packets
    increases the effects of losses and delay, audio only
    packets need to be smaller increasing the overhead. An
    A/V bundled format can provide about 1% overall overhead
    reduction. Considering the high bitrates used for MPEG-2
    encoded material, e.g. 4 Mbps, the number of bits saved,
    e.g. 40 Kbps, may provide noticeable audio or video
    quality improvement.

    4. May reduce overall receiver buffer size. Audio and
    video streams may experience different delays when
    transmitted separately. The receiver buffers need to be
    designed for the longest of these delays. For example,
    let's assume that using two buffers, each with a size B,
    is sufficient with probability P when each stream is
    transmitted individually. The probability that the same
    buffer size will be sufficient when both streams need to



draft-civanlar-bmpeg-02.txt                                     [Page 2]

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    be received is P times the conditional probability of B
    being sufficient for the second stream given that it was
    sufficient for the first one. This conditional probability
    is, generally, less than one requiring use of a larger
    buffer size to achieve the same probability level.

    5. May help with the control of the overall bandwidth used
    by an A/V program.

 And, the advantages over packetization of the transport layer streams
 are:

   1. Reduced overhead. It does not contain systems layer
   information which is redundant for the RTP (essentially
   they address similar issues).

   2. Easier error recovery. Because of the structured
   packetization consistent with the application layer
   framing (ALF) principle, loss concealment and error
   recovery can be made simpler and more effective.

2. Encapsulation of Bundled MPEG Video and Audio

Video encapsulation follows rules similar to the ones described in [3]
for encapsulation of MPEG elementary streams. Specifically,

  1. The MPEG Video_Sequence_Header, when present, will always
     be at the beginning of an RTP payload.
  2. An MPEG GOP_header, when present, will always be at the
     beginning of the RTP payload, or will follow a
     Video_Sequence_Header.
  3. An MPEG Picture_Header, when present, will always be at the
     beginning of a RTP payload, or will follow a GOP_header.

In addition to these, it is required that:

  4. Each packet must contain an integral number of video slices.

It is the application's responsibility to adjust the slice sizes and the
number of slices put in each RTP packet so that lower level
fragmentation does not occur. This approach simplifies the receivers
while somewhat increasing the complexity of the transmitter's
packetizer. Considering that a slice can be as small as a single
macroblock, it is possible to prevent fragmentation for most of the
cases.  If  a packet size exceeds the path maximum transmission unit
(path-MTU) [4], this payload type depends on the lower protocol layers
for fragmentation and this may cause problems with packet classification
for integrated services (e.g. with RSVP).



draft-civanlar-bmpeg-02.txt                                     [Page 3]

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The video data is followed by a sufficient number of integral audio
frames to cover the duration of the video segment included in a packet.
For example, if the first packet contains three 1/900 seconds long
slices of video, and Layer I audio coding is used at a 44.1kHz sampling
rate, only one audio frame covering 384/44100 seconds of audio need be
included in this packet. Since the length of this audio frame (8.71
msec.) is longer than that of the video segment contained in this packet
(3.33 msec), the next few packets may not contain any audio frames until
the packet in which the covered video time extends outside the length of
the previously transmitted audio frames. Alternatively, it is possible,
in this proposal, to repeat the latest audio frame in "no-audio" packets
for packet loss resilience. Again, it is the application's
responsibility to adjust the bundled packet size according to the
minimum MTU size to prevent fragmentation.

2.1. RTP Fixed Header for BMPEG Encapsulation

The following RTP header fields are used:

  Payload Type: A distinct payload type number, which may be dynamic,
  should be assigned to BMPEG.

  M Bit: Set for packets containing end of a picture.

  timestamp: 32-bit 90 kHz timestamp representing sampling time of the
  MPEG picture. May not be monotonically increasing if B pictures are
  present. Same for all packets belonging to the same picture. For
  packets that contain only a sequence, extension and/or GOP header, the
  timestamp is that of the subsequent picture.

2.2. BMPEG Specific Header:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| P |N|MBZ|    Audio Length   | |         Audio Offset          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                              MBZ

  P: Picture type (2 bits). I (0), P (1), B (2).

  N: Header data changed (1 bit). Set if any part of the video sequence,
  extension, GOP and picture header data is different than that of the
  previously sent headers. It gets reset when all the header data gets
  repeated (see Appendix 2).

  MBZ: Must be zero. Reserved for future use.




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  Audio Length: (10 bits) Length of the audio data in this packet in
  bytes. Start of the audio data is found by subtracting "Audio Length"
  from the total length of the received packet.

  Audio Offset: (16 bits) The offset between the start of the audio
  frame and the RTP timestamp for this packet in number of audio samples
  (for multi-channel sources, a set of samples covering all channels is
  counted as one sample for this purpose.)

  Audio offset is a signed integer in two's complement form. It allows a
  ~ +/- 750 msec offset at 44.1 KHz audio sampling rate. For a very low
  video frame rate (e.g., a frame per second), this offset may not be
  sufficient and this payload format may not be usable.

  If  B frames are present, audio frames are not re-ordered along with
  video.  Instead, they are packetized along with video frames in their
  transmission order  (e.g., an audio segment packetized with a video
  segment corresponding to a P picture may belong to a B picture, which
  will be transmitted later and should be rendered at the same time with
  this audio segment.) Even though the video segments are reordered, the
  audio offset for a particular audio segment is still relative to the
  RTP timestamp in the packet containing that audio segment.

  Since a special picture counter, such as  the "temporal reference
  (TR)" field of [3], is not included in this payload format, lost GOP
  headers may not be detected.  The only effect of this may be incorrect
  decoding of the B pictures immediately following the lost GOP header
  for some edited video material.

3. Security Considerations

RTP packets using the payload format defined in this specification are
subject to the security considerations discussed in the RTP
specification [1]. This implies that confidentiality of the media
streams is achieved by encryption. Because the data compression used
with this payload format is applied end-to-end, encryption may be
performed after compression so there is no conflict between the two
operations.

This payload type does not exhibit any significant non-uniformity in the
receiver side computational complexity for packet processing  to cause a
potential denial-of-service threat.

A security review of this payload format found no additional
considerations beyond those in the RTP specification.






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Appendix 1. Out-of-band Transmission of the "High Priority" Information

In MPEG encoded video, loss of the header information, which includes
sequence, GOP, and picture headers, and the corresponding extensions,
causes severe degradations in the decoded video. When possible,
dependable transmission of the header information to the receivers can
improve the loss resiliency of MPEG video significantly [5].  RFC 2038
describes a payload type where the header information can be repeated in
each RTP packet. Although this is a straightforward approach, it may
increase the overhead.

The "data partitioning" method in MPEG-2 defines the syntax and
semantics for partitioning an MPEG-2 encoded video bitstream into "high
priority" and "low priority" parts. If the "high priority" (HP) part is
selected to contain only the header information, it is less than two
percent of the video data and can be transmitted before the start of the
real-time transmission using a reliable protocol. In order to
synchronize the HP data with the corresponding real-time stream, the
initial value of the timestamp for the real-time stream may be inserted
at the beginning of the HP data.

Alternatively, the HP data may be transmitted along with the A/V data
using layered multimedia transmission techniques for RTP [6].

Appendix 2. Error Recovery

Packet losses can be detected from a combination of the sequence number
and the timestamp fields of the RTP fixed header. The extent of the loss
can be determined from the timestamp, the slice number and the
horizontal location of the first slice in the packet. The slice number
and the horizontal location can be determined from the slice header and
the first macroblock address increment, which are located at fixed bit
positions.

If lost data consists of slices all from the same picture, new data
following the loss may simply be given to the video decoder which will
normally repeat missing pixels from a previous picture. The next audio
frame must be played at the appropriate time determined by the timestamp
and the audio offset contained in the received packet. Appropriate audio
frames (e.g., representing background noise) may need to be fed to the
audio decoder in place of the lost audio frames to keep the lip-synch
and/or to conceal the effects of the losses.

If the received new data after a loss is from the next picture (i.e. no
complete picture loss) and the N bit is not set, previously received
headers for the particular picture type (determined from the P bits) can
be given to the video decoder followed by the new data. If N is set,
data deletion until a new picture start code is advisable unless headers



draft-civanlar-bmpeg-02.txt                                     [Page 6]

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are available from previously received HP data.

If data for more than one picture is lost and HP data is not available,
unless N is zero and at least one packet has been received for every
intervening picture of the same type and that the N bit was 0 for each
of those pictures, resynchronization to a new video sequence header is
advisable.

In all cases of large packet losses, if the HP data is available,
appropriate portions of it can be given to the video decoder and the
received data can be used irrespective of the N bit value or the number
of lost pictures.

Appendix 3. Resynchronization

As described in [3], use of frequent video sequence headers makes it
possible to join in a program at arbitrary times. Also, it reduces the
resynchronization time after severe losses.

References:

[1] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson,
    "RTP: A Transport Protocol for Real-Time Applications,"
    RFC 1889, January 1996.

[2] ISO/IEC International Standard 13818; "Generic coding of moving
    pictures and associated audio information," November 1994.

[3] D.Hoffman, G. Fernando, V. Goyal, M. R. Civanlar, "RTP Payload Format
    for MPEG1/MPEG2 Video," draft-ietf-avt-mpeg-new-00.txt, April 1997.

[4] J. Mogul, S. Deering, "Path MTU Discovery," RFC 1191, November 1990.

[5] M. R. Civanlar, G. L. Cash, "A practical system for MPEG-2 based
    video-on-demand over ATM packet networks and the WWW," Signal
    Processing: Image Communication, no. 8, pp. 221-227, Elsevier, 1996.

[6] M. F. Speer, S. McCanne, "RTP Usage with Layered Multimedia
    Streams," Internet Draft, draft-speer-avt-layered-video-02.txt,
    December 1996.

Author's  Address:

   M. Reha Civanlar
   Glenn L. Cash
   Barry G. Haskell

   AT&T Labs-Research



draft-civanlar-bmpeg-02.txt                                     [Page 7]

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   100 Schultz Drive
   Red Bank, NJ 07701
   USA

   e-mail: civanlar|glenn|bgh@research.att.com














































draft-civanlar-bmpeg-02.txt                                     [Page 8]


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