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Network Working Group                                            M. HandleyRequest for Comments: 2736                                            ACIRIBCP: 36                                                          C. PerkinsCategory: Best Current Practice                                         UCL                                                              December 1999Guidelines for Writers of RTP Payload Format SpecificationsStatus of this Memo   This document specifies an Internet Best Current Practices for the   Internet Community, and requests discussion and suggestions for   improvements.  Distribution of this memo is unlimited.Copyright Notice   Copyright (C) The Internet Society (1999).  All Rights Reserved.Abstract   This document provides general guidelines aimed at assisting the   authors of RTP Payload Format specifications in deciding on good   formats.  These guidelines attempt to capture some of the experience   gained with RTP as it evolved during its development.1.  Introduction   This document provides general guidelines aimed at assisting the   authors of RTP [9] Payload Format specifications in deciding on good   formats.  These guidelines attempt to capture some of the experience   gained with RTP as it evolved during its development.   The principles outlined in this document are applicable to almost all   data types, but are framed in examples of audio and video codecs for   clarity.2.  Background   RTP was designed around the concept of Application Level Framing   (ALF), first described by Clark and Tennenhouse [2]. The key argument   underlying ALF is that there are many different ways an application   might be able to cope with misordered or lost packets.  These range   from ignoring the loss, to re-sending the missing data (either from a   buffer or by regenerating it), and to sending new data which   supersedes the missing data.  The application only has this choice if   the transport protocol is dealing with data in "Application Data   Units" (ADUs). An ADU contains data that can be processed out-of-Handley & Perkins        Best Current Practice                  [Page 1]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999   order with respect to other ADUs.  Thus the ADU is the minimum unit   of error recovery.   The key property of a transport protocol for ADUs is that each ADU   contains sufficient information to be processed by the receiver   immediately.  An example is a video stream, wherein the compressed   video data in an ADU must be capable of being decompressed regardless   of whether previous ADUs have been received.  Additionally the ADU   must contain "header" information detailing its position in the video   image and the frame from which it came.   Although an ADU need not be a packet, there are many applications for   which a packet is a natural ADU.  Such ALF applications have the   great advantage that all packets that are received can be processed   by the application immediately.   RTP was designed around an ALF philosophy.  In the context of a   stream of RTP data, an RTP packet header provides sufficient   information to be able to identify and decode the packet irrespective   of whether it was received in order, or whether preceding packets   have been lost. However, these arguments only hold good if the RTP   payload formats are also designed using an ALF philosophy.   Note that this also implies smart, network aware, end-points. An   application using RTP should be aware of the limitations of the   underlying network, and should adapt its transmission to match those   limitations.  Our experience is that a smart end-point implementation   can achieve significantly better performance on real IP-based   networks than a naive implementation.3.  Channel Characteristics   We identify the following channel characteristics that influence the   best-effort transport of RTP over UDP/IP in the Internet:   o  Packets may be lost   o  Packets may be duplicated   o  Packets may be reordered in transit   o  Packets will be fragmented if they exceed the MTU of the      underlying network   The loss characteristics of a link may vary widely over short time   intervals.Handley & Perkins        Best Current Practice                  [Page 2]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999   Although fragmentation is not a disastrous phenomenon if it is a rare   occurrence, relying on IP fragmentation is a bad design strategy as   it significantly increases the effective loss rate of a network and   decreases goodput.  This is because if one fragment is lost, the   remaining fragments (which have used up bottleneck bandwidth) will   then need to be discarded by the receiver.  It also puts additional   load on the routers performing fragmentation and on the end-systems   re-assembling the fragments.   In addition, it is noted that the transit time between two hosts on   the Internet will not be constant.  This is due to two effects -   jitter caused by being queued behind cross-traffic, and routing   changes.  The former is possible to characterise and compensate for   by using a playout buffer, but the latter is impossible to predict   and difficult to accommodate gracefully.4.  Guidelines   We identify the following requirements of RTP payload format   specifications:   +  A payload format should be devised so that the stream being      transported is still useful even in the presence of a moderate      amount of packet loss.   +  Ideally all the contents of every packet should be possible to be      decoded and played out irrespective of whether preceding packets      have been lost or arrive late.   The first of these requirements is based on the nature of the   Internet.  Although it may be possible to engineer parts of the   Internet to produce low loss rates through careful provisioning or   the use of non-best-effort services, as a rule payload formats should   not be designed for these special purpose environments.  Payload   formats should be designed to be used in the public Internet with   best effort service, and thus should expect to see moderate loss   rates.  For example, a 5% loss rate is not uncommon.  We note that   TCP steady state models [3][4][6] indicate that a 5% loss rate with a   1KByte packet size and 200ms round-trip time will result in TCP   achieving a throughput of around 180Kbit/s.  Higher loss rates,   smaller packet sizes, or a larger RTT are required to constrain TCP   to lower data rates.  For the most part, it is such TCP traffic that   is producing the background loss that many RTP flows must co-exist   with.  Without explicit congestion notification (ECN) [8], loss must   be considered an intrinsic property of best-effort parts of the   Internet.Handley & Perkins        Best Current Practice                  [Page 3]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999   When payload formats do not assume packet loss will occur, they   should state this explicitly up front, and they will be considered   special purpose payload formats, unsuitable for use on the public   Internet without special support from the network infrastructure.   The second of these requirements is more explicit about how RTP   should cope with loss.  If an RTP payload format is properly   designed, every packet that is actually received should be useful.   Typically this implies the following guidelines are adhered to:   +  Packet boundaries should coincide with codec frame boundaries.      Thus a packet should normally consist of one or more complete      codec frames.   +  A codec's minimum unit of data should never be packetised so that      it crosses a packet boundary unless it is larger than the MTU.   +  If a codec's frame size is larger than the MTU, the payload format      must not rely on IP fragmentation.  Instead it must define its own      fragmentation mechanism.  Such mechanisms may involve codec-      specific information that allows decoding of fragments.      Alternatively they might allow codec-independent packet-level      forward error correction [5] to be applied that cannot be used      with IP-level fragmentation.   In the abstract, a codec frame (i.e., the ADU or the minimum size   unit that has semantic meaning when handed to the codec) can be of   arbitrary size.  For PCM audio, it is one byte.  For GSM audio, a   frame corresponds to 20ms of audio.  For H.261 video, it is a Group   of Blocks (GOB), or one twelfth of a CIF video frame.   For PCM, it does not matter how audio is packetised, as the ADU size   is one byte.  For GSM audio, arbitrary packetisation would split a   20ms frame over two packets, which would mean that if one packet were   lost, partial frames in packets before and after the loss are   meaningless.  This means that not only were the bits in the missing   packet lost, but also that additional bits in neighboring packets   that used bottleneck bandwidth were effectively also lost because the   receiver must throw them away.  Instead, we would packetise GSM by   including several complete GSM frames in a packet; typically four GSM   frames are included in current implementations.  Thus every packet   received can be decoded because even in the presence of loss, no   incomplete frames are received.   The H.261 specification allows GOBs to be up to 3KBytes long,   although most of the time they are smaller than this.  It might be   thought that we should insert a group of blocks into a packet when it   fits, and arbitrarily split the GOB over two or more packets when aHandley & Perkins        Best Current Practice                  [Page 4]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999   GOB is large.  In the first version of the H.261 payload format, this   is what was done.  However, this still means that there are   circumstances where H.261 packets arrive at the receiver and must be   discarded because other packets were lost - a loss multiplier effect   that we wish to avoid.  In fact there are smaller units than GOBs in   the H.261 bit-stream called macroblocks, but they are not   identifiable without parsing from the start of the GOB.  However, if   we provide a little additional information at the start of each   packet, we can reinstate information that would normally be found by   parsing from the start of the GOB, and we can packetise H.261 by   splitting the data stream on macroblock boundaries.  This is a less   obvious packetisation for H.261 than the GOB packetisation, but it   does mean that a slightly smarter depacketiser at the receiver can   reconstruct a valid H.261 bitstream from a stream of RTP packets that   has experienced loss, and not have to discard any of the data that   arrived.   An additional guideline concerns codecs that require the decoder   state machine to keep step with the encoder state machine.  Many   audio codecs such as LPC or GSM are of this form.  Typically they are   loss tolerant, in that after a loss, the predictor coefficients   decay, so that after a certain amount of time, the predictor error   induced by the loss will disappear.  Most codecs designed for   telephony services are of this form because they were designed to   cope with bit errors without the decoder predictor state permanently   remaining incorrect.  Just packetising these formats so that packets   consist of integer multiples of codec frames may not be optimal, as   although the packet received immediately after a packet loss can be   decoded, the start of the audio stream produced will be incorrect   (and hence distort the signal) because the decoder predictor is now   out of step with the encoder.  In principle, all of the decoder's   internal state could be added using a header attached to the start of   every packet, but for lower bit-rate encodings, this state is so   substantial that the bit rate is no longer low.  However, a   compromise can usually be found, where a greatly reduced form of   decoder state is sent in every packet, which does not recreate the   encoders predictor precisely, but does reduce the magnitude and   duration of the distortion produced when the previous packet is lost.   Such compressed state is, by definition, very dependent on the codec   in question.  Thus we recommend:   +  Payload formats for encodings where the decoder contains internal      data-driven state that attempts to track encoder state should      normally consider including a small additional header that conveys      the most critical elements of this state to reduce distortion      after packet loss.Handley & Perkins        Best Current Practice                  [Page 5]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999   A similar issue arises with codec parameters, and whether or not they   should be included in the payload format. An example is with a codec   that has a choice of huffman tables for compression.  The codec may   use either huffman table 1 or table 2 for encoding and the receiver   needs to know this information for correct decoding. There are a   number of ways in which this kind of information can be conveyed:   o  Out of band signalling, prior to media transmission.   o  Out of band signalling, but the parameter can be changed mid-      session.  This requires synchronization of the change in the media      stream.   o  The change is signaled through a change in the RTP payload type      field. This requires mapping the parameter space into particular      payload type values and signalling this mapping out-of-band prior      to media transmission.   o  Including the parameter in the payload format. This allows for      adapting the parameter in a robust manner, but makes the payload      format less efficient.   Which mechanism to use depends on the utility of changing the   parameter in mid-session to support application layer adaptation.   However, using out-of-band signalling to change a parameter in mid-   session is generally to be discouraged due to the problem of   synchronizing the parameter change with the media stream.4.1.  RTP Header Extensions   Many RTP payload formats require some additional header information   to be carried in addition to that included in the fixed RTP packet   header.  The recommended way of conveying this information is in the   payload section of the packet. The RTP header extension should not be   used to convey payload specific information ([9], section 5.3) since   this is inefficient in its use of bandwidth; requires the definition   of a new RTP profile or profile extension; and makes it difficult to   employ FEC schemes such as, for example, [7].  Use of an RTP header   extension is only appropriate for cases where the extension in   question applies across a wide range of payload types.4.2.  Header Compression   Designers of payload formats should also be aware of the needs of RTP   header compression [1]. In particular, the compression algorithm   functions best when the RTP timestamp increments by a constant value   between consecutive packets. Payload formats which rely on sending   packets out of order, such that the timestamp increment is notHandley & Perkins        Best Current Practice                  [Page 6]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999   constant, are likely to compress less well than those which send   packets in order. This has most often been an issue when designing   payload formats for FEC information, although some video codecs also   rely on out-of-order transmission of packets at the expense of   reduced compression. Although in some cases such out-of-order   transmission may be the best solution, payload format designers are   encourage to look for alternative solutions where possible.5.  Summary   Designing packet formats for RTP is not a trivial task.  Typically a   detailed knowledge of the codec involved is required to be able to   design a format that is resilient to loss, does not introduce loss   magnification effects due to inappropriate packetisation, and does   not introduce unnecessary distortion after a packet loss.  We believe   that considerable effort should be put into designing packet formats   that are well tailored to the codec in question.  Typically this   requires a very small amount of processing at the sender and   receiver, but the result can be greatly improved quality when   operating in typical Internet environments.   Designers of new codecs for use with RTP should consider making the   output of the codec "naturally packetizable". This implies that the   codec should be designed to produce a packet stream, rather than a   bit-stream; and that that packet stream contains the minimal amount   of redundancy necessary to ensure that each packet is independently   decodable with minimal loss of decoder predictor tracking. It is   recognised that sacrificing some small amount of bandwidth to ensure   greater robustness to packet loss is often a worthwhile tradeoff.   It is hoped that, in the long run, new codecs should be produced   which can be directly packetised, without the trouble of designing a   codec-specific payload format.   It is possible to design generic packetisation formats that do not   pay attention to the issues described in this document, but such   formats are only suitable for special purpose networks where packet   loss can be avoided by careful engineering at the network layer, and   are not suited to current best-effort networks.6.  Security Considerations   The guidelines in this document result in RTP payload formats that   are robust in the presence of real world network conditions.   Designing payload formats for special purpose networks that assume   negligable loss rates will normally result in slightly better   compression, but produce formats that are more fragile, thus   rendering them easier targets for denial-of-service attacks.Handley & Perkins        Best Current Practice                  [Page 7]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999   Designers of payload formats should pay close attention to possible   security issues that might arise from poor implementations of their   formats, and should be careful to specify the correct behaviour when   anomalous conditions arise.  Examples include how to process illegal   field values, and conditions when there are mismatches between length   fields and actual data.  Whilst the correct action will normally be   to discard the packet, possible such conditions should be brought to   the attention of the implementor to ensure that they are trapped   properly.   The RTP specification covers encryption of the payload.  This issue   should not normally be dealt with by payload formats themselves.   However, certain payload formats spread information about a   particular application data unit over a number of packets, or rely on   packets which relate to a number of application data units. Care must   be taken when changing the encryption of such streams, since such   payload formats may constrain the places in a stream where it is   possible to change the encryption key without exposing sensitive   data.   Designers of payload formats which include FEC should be aware that   the automatic addition of FEC in response to packet loss may increase   network congestion, leading to a worsening of the problem which the   use of FEC was intended to solve. Since this may, at its worst,   constitute a denial of service attack, designers of such payload   formats should take care that appropriate safeguards are in place to   prevent abuse.Authors' Addresses   Mark Handley   AT&T Center for Internet Research at ICSI,   International Computer Science Institute,   1947 Center Street, Suite 600,   Berkeley, CA 94704, USA   EMail: mjh@aciri.org   Colin Perkins   Dept of Computer Science,   University College London,   Gower Street,   London WC1E 6BT, UK.   EMail: C.Perkins@cs.ucl.ac.ukHandley & Perkins        Best Current Practice                  [Page 8]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999Acknowledgments   This document is based on experience gained over several years by   many people, including Van Jacobson, Steve McCanne, Steve Casner,   Henning Schulzrinne, Thierry Turletti, Jonathan Rosenberg and   Christian Huitema amongst others.References   [1]  Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP Headers for        Low-Speed Serial Links",RFC 2508, February 1999.   [2]  D. Clark and  D. Tennenhouse, "Architectural Considerations for        a New Generation of Network Protocols" Proc ACM Sigcomm 90.   [3]  J. Mahdavi and S. Floyd. "TCP-friendly unicast rate-based flow        control". Note sent to end2end-interest mailing list, Jan 1997.   [4]  M. Mathis, J. Semske, J. Mahdavi, and T. Ott. "The macro-scopic        behavior of the TCP congestion avoidance algorithm". Computer        Communication Review, 27(3), July 1997.   [5]  J. Nonnenmacher, E. Biersack, Don Towsley, "Parity-Based Loss        Recovery for Reliable Multicast Transmission", Proc ACM Sigcomm   [6]  J. Padhye, V. Firoiu, D. Towsley, J.  Kurose, "Modeling TCP        Throughput: A Simple Model and its Empirical Validation", Proc.        ACM Sigcomm 1998.   [7]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,        Bolot, J.C., Vega-Garcia, A. and S. Fosse-Parisis, "RTP Payload        for Redundant Audio Data",RFC 2198, September 1997.   [8]  Ramakrishnan, K. and  S. Floyd, "A Proposal to add Explicit        Congestion Notification (ECN) to IP",RFC 2481, January 1999.   [9]  Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,        "Real-Time Transport Protocol",RFC 1889, January 1996.Handley & Perkins        Best Current Practice                  [Page 9]

RFC 2736     Guidelines for Writers of RTP Payload Formats December 1999Full Copyright Statement   Copyright (C) The Internet Society (1999).  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.Acknowledgement   Funding for the RFC Editor function is currently provided by the   Internet Society.Handley & Perkins        Best Current Practice                 [Page 10]