Network Working Group|
Request for Comments: 2343
This memo defines an Experimental Protocol for the Internet community. This memo does not specify an Internet standard of any kind. Discussion and suggestions for improvement are requested. Distribution of this memo is unlimited.
Copyright © The Internet Society (1998). All Rights Reserved.
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.
This document describes a bundled packetization scheme for MPEG-2 encoded audio and video streams using the Real-time Transport Protocol (RTP), version 2 .
The MPEG-2 International standard consists of three layers: audio, video and systems . 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 2250  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:
And, the advantages over packetization of the transport layer streams are:
Video encapsulation follows rules similar to the ones described in  for encapsulation of MPEG elementary streams. Specifically,
In addition to these, it is required that:
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) , this payload type depends on the lower protocol layers for fragmentation although, this may cause problems with packet classification for integrated services (e.g. with RSVP).
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.
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.
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 1).
MBZ: Must be zero. Reserved for future use.
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 , 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.
RTP packets using the payload format defined in this specification are subject to the security considerations discussed in the RTP specification . 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.
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 are made available to the receiver through some other channel.
If data for more than one picture is lost and headers are 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 heavy packet losses, if the correct headers for the missing Pictures are available, they 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.
As described in , 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.
 Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", RFC 1889, January 1996.
 ISO/IEC International Standard 13818; "Generic coding of moving pictures and associated audio information," November 1994.
 Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar, "RTP Payload Format for MPEG1/MPEG2 Video", RFC 2250, January 1998.
 Mogul, J., and S. Deering, "Path MTU Discovery", RFC 1191, November 1990.
Glenn L. Cash
100 Schultz Drive
Red Bank, NJ 07701
Barry G. Haskell
100 Schultz Drive
Red Bank, NJ 07701
Copyright © The Internet Society (1998). All Rights Reserved.
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