Network Working Group M. Civanlar Request for Comments: 2343 G. Cash Category: Experimental B. Haskell AT&T Labs-Research May 1998 RTP Payload Format for Bundled MPEG Status of this Memo 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 Notice Copyright (C) The Internet Society (1998). All Rights Reserved. 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. 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 . 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: 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 streams 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 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  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) , 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. 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 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. 3. Security Considerations 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.
Appendix 1. 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 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. Appendix 2. Resynchronization 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.
References  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. Authors' Addresses M. Reha Civanlar AT&T Labs-Research 100 Schultz Drive Red Bank, NJ 07701 USA EMail: firstname.lastname@example.org Glenn L. Cash AT&T Labs-Research 100 Schultz Drive Red Bank, NJ 07701 USA EMail: email@example.com Barry G. Haskell AT&T Labs-Research 100 Schultz Drive Red Bank, NJ 07701 USA EMail: firstname.lastname@example.org
Full Copyright Statement Copyright (C) The Internet Society (1998). All Rights Reserved. This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English. The limited permissions granted above are perpetual and will not be revoked by the Internet Society or its successors or assigns. This document and the information contained herein is provided on an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.