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Network Working Group                                         C. PerkinsRequest for Comments: 2354                                     O. HodsonCategory: Informational                        University College London                                                               June 1998Options for Repair of Streaming MediaStatus of this Memo   This memo provides information for the Internet community.  This memo   does not specify an Internet standard of any kind.  Distribution of   this memo is unlimited.Copyright Notice   Copyright (C) The Internet Society (1998).  All Rights Reserved.Abstract   This document summarizes a range of possible techniques for the   repair of continuous media streams subject to packet loss.  The   techniques discussed include redundant transmission, retransmission,   interleaving and forward error correction.  The range of   applicability of these techniques is noted, together with the   protocol requirements and dependencies.1  Introduction   A number of applications have emerged which use RTP/UDP transport to   deliver continuous media streams.  Due to the unreliable nature of   UDP packet delivery, the quality of the received stream will be   adversely affected by packet loss.  A number of techniques exist by   which the effects of packet loss may be repaired.  These techniques   have a wide range of applicability and require varying degrees of   protocol support.  In this document, a number of such techniques are   discussed, and recommendations for their applicability made.   It should be noted that this document is introductory in nature, and   does not attempt to be comprehensive.  In particular, we restrict our   discussion to repair techniques which require the involvement of the   sender of a media stream, and do not discuss possibilities for   receiver based repair.   For a more detailed survey, the reader is referred to [5].Perkins & Hodson             Informational                      [Page 1]

RFC 2354         Options for Repair of Streaming Media         June 19982  Terminology and Protocol Framework   A unit is defined to be a timed interval of media data, typically   derived from the workings of the media coder.  A packet comprises one   or more units, encapsulated for transmission over the network.  For   example, many audio coders operate on 20ms units, which are typically   combined to produce 40ms or 80ms packets for transmission.  The   framework of RTP [18] is assumed.  This implies that packets have a   sequence number and timestamp.  The sequence number denotes the order   in which packets are transmitted, and is used to detect losses.  The   timestamp is used to determine the playout order of units.  Most loss   recovery schemes rely on units being sent out of order, so an   application must use the RTP timestamp to schedule playout.   The use of RTP allows for several different media coders, with a   payload type field being used to distinguish between these at the   receiver.  Some loss repair schemes send multiple copies of units, at   different times and possibly with different encodings, to increase   the probability that a receiver has something to decode.  A receiver   is assumed to have a `quality' ranking of the differing encodings,   and so is capable of choosing the `best' unit for playout, given   multiple options.   A session is defined as interactive if the end-to-end delay is less   then 250ms, including media coding and decoding, network transit and   host buffering.3  Network Loss Characteristics   If it is desired to repair a media stream subject to packet loss, it   is useful to have some knowledge of the loss characteristics which   are likely to be encountered.  A number of studies have been   conducted on the loss characteristics of the Mbone [2,8,21] and   although the results vary somewhat, the broad conclusion is clear:   in a large conference it is inevitable that some receivers will   experience packet loss.  Packet traces taken by Handley [8] show a   session in which most receivers experience loss in the range 2-5%,   with a somewhat smaller number seeing significantly higher loss   rates.  Other studies have presented broadly similar results.   It has also been shown that the vast majority of losses are of single   packets.  Burst losses of two or more packets are around an order of   magnitude less frequent than single packet loss, although they do   occur more often than would be expected from a purely random process.   Longer burst losses (of the order of tens of packets) occur   infrequently.  These results are consistent with a network where   small amounts of transient congestion cause the majority of packet   loss.  In a few cases, a network link is found to be severelyPerkins & Hodson             Informational                      [Page 2]

RFC 2354         Options for Repair of Streaming Media         June 1998   overloaded, and large amount of loss results.   The primary focus of a repair scheme must, therefore, be to correct   single packet loss, since this is by far the most frequent   occurrence.  It is desirable that losses of a relatively small number   of consecutive packets may also be repaired, since such losses   represent a small but noticeable fraction of observed losses.  The   correction of large bursts of loss is of considerably less   importance.4  Loss Mitigation Schemes   In the following sections, four loss mitigation schemes are   discussed.  These schemes have been discussed in the literature a   number of times, and found to be of use in a number of scenarios.   Each technique is briefly described, and its advantages and   disadvantages noted.4.1 Retransmission   Retransmission of lost packets is an obvious means by which loss may   be repaired.  It is clearly of value in non-interactive applications,   with relaxed delay bounds, but the delay imposed means that it does   not typically perform well for interactive use.   In addition to the possibly high latency, there is a potentially   large bandwidth overhead to the use of retransmission.  Not only are   units of data sent multiple times, but additional control traffic   must flow to request the retransmission.  It has been shown that, in   a large Mbone session, most packets are lost by at least one receiver   [8].  In this case the overhead of requesting retransmission for most   packets may be such that the use of forward error correction is more   acceptable.  This leads to a natural synergy between the two   mechanisms, with a forward error correction scheme being used to   repair all single packet losses, and those receivers experiencing   burst losses, and willing to accept the additional latency, using   retransmission based repair as an additional recovery mechanism.   Similar mechanisms have been used in a number of reliable multicast   schemes, and have received some discussion in the literature [9,13].   In order to reduce the overhead of retransmission, the retransmitted   units may be piggy-backed onto the ongoing transmission, using a   payload format such as that described in [15].  This also allows for   the retransmission to be recoded in a different format, to further   reduce the bandwidth overhead.  As an alternative, FEC information   may be sent in response to retransmission requests [13], allowing a   single retransmission to potentially repair several losses.  The   choice of a retransmission request algorithm which is both timely andPerkins & Hodson             Informational                      [Page 3]

RFC 2354         Options for Repair of Streaming Media         June 1998   network friendly is an area of current study.  An obvious starting   point is the SRM protocol [7], and experiments have been conducted   using this, and with a low-delay variant, STORM [20].  This work   shows the trade-off between latency and quality for retransmission   based repair schemes, and illustrates that retransmission is an   effective approach to repair for applications which can tolerate the   latency.   There is no standard protocol framework for requesting retransmission   of streaming media.  An experimental RTP profile extension for SRM-   style retransmission requests has described in [14].4.2 Forward Error Correction   Forward error correction (FEC) is the means by which repair data is   added to a media stream, such that packet loss can be repaired by the   receiver of that stream with no further reference to the sender.   There are two classes of repair data which may be added to a stream:   those which are independent of the contents of the stream, and those   which use knowledge of the stream to improve the repair process.4.2.1 Media-Independent FEC   A number of media-independent FEC schemes have been proposed for use   with streamed media.  These techniques add redundant data, which is   transmitted in separate packets, to a media stream.  Traditionally,   FEC techniques are described as loss detecting and/or loss   correcting.  In the case of streamed media, loss detection is   provided by the sequence numbers in RTP packets.   The redundant FEC data is typically calculated using the mathematics   of finite fields [1].  The simplest of finite field is GF(2) where   addition is just the eXclusive-OR operation.  Basic FEC schemes   transmit k data packets with n-k parity packets allowing the   reconstruction of the original data from any k of the n transmitted   packets.  Budge et al [4] proposed applying the XOR operation across   different combinations of the media data with the redundant data   transmitted separately as parity packets.  These vary the pattern of   packets over which the parity is calculated, and hence have different   bandwidth, latency and loss repair characteristics.   Parity-based FEC based techniques have a significant advantage in   that they are media independent, and provide exact repair for lost   packets.  In addition, the processing requirements are relatively   light, especially when compared with some media-specific FEC   (redundancy) schemes which use very low bandwidth, but high   complexity encodings.  The disadvantage of parity based FEC is that   the codings have higher latency in comparison with the media-specificPerkins & Hodson             Informational                      [Page 4]

RFC 2354         Options for Repair of Streaming Media         June 1998   schemes discussed in following section.   A number of FEC schemes exist which are based on higher-order finite   fields, for example Reed-Solomon (RS) codes, which are more   sophisticated and computationally demanding.  These are usually   structured so that they have good burst loss protection, although   this typically comes at the expense of increased latency.  Dependent   on the observed loss patterns, such codes may give improved   performance, compared to parity-based FEC.   An RTP payload format for generic FEC, suitable for both parity based   and Reed-Solomon encoded FEC is defined in [17].4.2.2 Media-Specific FEC   The basis of media-specific FEC is to employ knowledge of a media   compression scheme to achieve more efficient repair of a stream than   can otherwise be achieved.  To repair a stream subject to packet   loss, it is necessary to add redundancy to that stream:  some   information is added which is not required in the absence of packet   loss, but which can be used to recover from that loss.   The nature of a media stream affects the means by which the   redundancy is added.  If units of media data are packets, or if   multiple units are included in a packet, it is logical to use the   unit as the level of redundancy, and to send duplicate units.  By   recoding the redundant copy of a unit, significant bandwidth savings   may be made, at the expense of additional computational complexity   and approximate repair.  This approach has been advocated for use   with streaming audio [2,10] and has been shown to perform well.  An   RTP payload format for this form of redundancy has been defined [15].   If media units span multiple packets, for instance video, it is   sensible to include redundancy directly within the output of a codec.   For example the proposed RTP payload for H.263+ [3] includes multiple   copies of key portions of the stream, separated to avoid the problems   of packet loss.  The advantages of this second approach is   efficiency: the codec designer knows exactly which portions of the   stream are most important to protect, and low complexity since each   unit is coded once only.   An alternative approach is to apply media-independent FEC techniques   to the most significant bits of a codecs output, rather than applying   it over the entire packet.  Several codec descriptions include bit   sensitivities that make this feasible.  This approach has low   computational cost and can be tailored to represent an arbitrary   fraction of the transmitted data.Perkins & Hodson             Informational                      [Page 5]

RFC 2354         Options for Repair of Streaming Media         June 1998   The use of media-specific FEC has the advantage of low-latency, with   only a single-packet delay being added.  This makes it suitable for   interactive applications, where large end-to-end delays cannot be   tolerated.  In a uni-directional non-interactive environment it is   possible to delay sending the redundant data, achieving improved   performance in the presence of burst losses [11], at the expense of   additional latency.4.3 Interleaving   When the unit size is smaller than the packet size, and end-to-end   delay is unimportant, interleaving [16] is a useful technique for   reducing the effects of loss.  Units are resequenced before   transmission, so that originally adjacent units are separated by a   guaranteed distance in the transmitted stream, and returned to their   original order at the receiver.  Interleaving disperses the effect of   packet losses.  If, for example, units are 5ms in length and packets   20ms (ie:  4 units per packet), then the first packet could contain   units 1, 5, 9, 13; the second packet would contain units 2, 6, 10,   14; and so on.  It can be seen that the loss of a single packet from   an interleaved stream results in multiple small gaps in the   reconstructed stream, as opposed to the single large gap which would   occur in a non-interleaved stream.  In many cases it is easier to   reconstruct a stream with such loss patterns, although this is   clearly media and codec dependent.  Note that the size of the gaps is   dependent on the degree of interleaving used, and can be made   arbitrarily small at the expense of additional latency.   The obvious disadvantage of interleaving is that it increases   latency.  This limits the use of this technique for interactive   applications, although it performs well for non-interactive use.  The   major advantage of interleaving is that it does not increase the   bandwidth requirements of a stream.   A potential RTP payload format for interleaved data is a simple   extension of the redundant audio payload [15].  That payload requires   that the redundant copy of a unit is sent after the primary.  If this   restriction is removed, it is possible to transmit an arbitrary   interleaving of units with this payload format.5  Recommendations   If the desired scenario is a non-interactive uni-directional   transmission, in the style of a radio or television broadcast,   latency is of considerably less importance than reception quality.   In this case, the use of interleaving, retransmission based repair or   FEC is appropriate.  If approximate repair is acceptable,   interleaving is clearly to be preferred, since it does not increasePerkins & Hodson             Informational                      [Page 6]

RFC 2354         Options for Repair of Streaming Media         June 1998   the bandwidth of a stream.  Media independent FEC is typically the   next best option, since a single FEC packet has the potential to   repair multiple lost packets, providing efficient transmission.   In an interactive session, the delay imposed by the use of   interleaving and retransmission is not acceptable, and a low-latency   FEC scheme is the only means of repair suitable.  The choice between   media independent and media specific forward error correction is less   clear-cut:  media-specific FEC can be made more efficient, but   requires modification to the output of the codec.  When defining the   packet format for a new codec, this is clearly an appropriate   technique, and should be encouraged.   If an existing codec is to be used, a media independent forward error   correction scheme is usually easier to implement, and can perform   well.  A media stream protected in this way may be augmented with   retransmission based repair with minimal overhead, providing improved   quality for those receivers willing to tolerate additional delay, and   allowing interactivity for those receivers which desire it.  Whilst   the addition of FEC data to an media stream is an effective means by   which that stream may be protected against packet loss, application   designers should be aware that the addition of large amounts of   repair data when loss is detected will increase network congestion,   and hence packet loss, leading to a worsening of the problem which   the use of error correction coding was intended to solve.   At the time of writing, there is no standard solution to the problem   of congestion control for streamed media which can be used to solve   this problem.  There have, however, been a number of contributions   which show the likely form the solution will take [12,19].  This   work typically used some form of layered encoding of data over   multiple channels, with receivers joining and leaving layers in   response to packet-loss (which indicates congestion).  The aim of   such schemes is to emulate the congestion control behavior of a TCP   stream, and hence compete fairly with non-real time traffic.  This is   necessary for stable network behavior in the presence of much   streamed media.   Since streaming media applications are in use now, without congestion   control, it is important to give some advice to authors of those   tools as to the behavior which is acceptable, until congestion   control mechanisms can be deployed.  The remainder of this section   uses the throughput of a TCP connection over a link with a given loss   rate as an example to indicate behavior which may be classified as   reasonable.   As a number of authors have noted (eg:  [6]), the loss rate and   throughput of a TCP connection are approximately related as follows:Perkins & Hodson             Informational                      [Page 7]

RFC 2354         Options for Repair of Streaming Media         June 1998    T = (s * c) / (RTT * sqrt(p))   where T is the observed throughput in octets per second, s is the   packet size in octets, RTT is the round-trip time for the session in   seconds, p is the packet loss rate and c is a constant in the range   0.9...1.5 (a value of 1.22 has been suggested [6]).  Using this   relation, one may determine the packet loss rate which would result   in a given throughput for a particular session, if a TCP connection   was used.   Whilst this relation between packet loss rate and throughput is   specific to the TCP congestion control algorithm, it also provides an   estimate of the acceptable loss rate for a streaming media   application using the same network path, which wishes to coexist   fairly with TCP traffic.  Clearly this is not sufficient for fair   sharing of a link with TCP traffic, since it does not capture the   dynamic behavior of the connection, merely the average behavior, but   it does provide one definition of "reasonable" behavior in the   absence of real congestion control.   For example, an RTP audio session with DVI encoding and 40ms data   packets will have 40 bytes RTP/UDP/IP header, 4 bytes codec state and   160 bytes of media data, giving a packet size, s, of 204 bytes.  It   will send 25 packets per second, giving T = 4800.  It is possible to   estimate the round-trip time from RTCP reception report statistics   (say 200 milliseconds for the purpose of example).  Substituting   these values into the above equation, we estimate a "reasonable"   packet loss rate, p, of 6.7%.  This would represent an upper bound on   the packet loss rate which this application should be designed to   tolerate.   It should be noted that a round trip time estimate based on RTCP   reception report statistics is, at best, approximate; and that a   round trip time for a multicast group can only be an `average'   measure.  This implies that the TCP equivalent throughput/loss rate   determined by this relation will be an approximation of the upper   bound to the rate a TCP connection would actually achieve.6  Security Considerations   Some of the repair techniques discussed in this document result in   the transmission of additional traffic to correct for the effects of   packet loss.  Application designers should be aware that the   transmission of large amounts of repair traffic will increase network   congestion, and hence packet loss, leading to a worsening of the   problem which the use of error correction was intended to solve.  At   its worst, this can lead to excessive network congestion and may   constitute a denial of service attack.Section 5 discusses this inPerkins & Hodson             Informational                      [Page 8]

RFC 2354         Options for Repair of Streaming Media         June 1998   more detail, and provides guidelines for prevention of this problem.7  Summary   Streaming media applications using the Internet will be subject to   packet loss due to the unreliable nature of UDP packet delivery.   This document has summarized the typical loss patterns seen on the   public Mbone at the time of writing, and a range of techniques for   recovery from such losses.  We have further discussed the need for   congestion control, and provided some guidelines as to reasonable   behavior for streaming applications in the interim until congestion   control can be deployed.8  Acknowledgments   The authors wish to thank Phil Karn and Lorenzo Vicisano for their   helpful comments.9  Authors' Addresses   Colin Perkins   Department of Computer Science   University College London   Gower Street   London WC1E 6BT   United Kingdom   EMail: c.perkins@cs.ucl.ac.uk   Orion Hodson   Department of Computer Science   University College London   Gower Street   London WC1E 6BT   United Kingdom   EMail: o.hodson@cs.ucl.ac.ukReferences   [1] R.E. Blahut. Theory and Practice ofError Control Codes.       Addison Wesley, 1983.   [2] J.-C. Bolot and A. Vega-Garcia. The case for FEC based       error control for packet audio in the Internet. To appear       in ACM Multimedia Systems.Perkins & Hodson             Informational                      [Page 9]

RFC 2354         Options for Repair of Streaming Media         June 1998   [3] C. Bormann, L. Cline, G. Deisher, T. Gardos, C. Maciocco,       D. Newell, J. Ott, S. Wenger, and C.  Zhu. RTP payload       format for the 1998 version of ITU-T reccomendation  H.263       video (H.263+).  Work in Progress.   [4] D. Budge, R. McKenzie, W. Mills, W. Diss,  and P. Long.       Media-independent error correction using RTP, Work in Progress.   [5] G. Carle and E. W. Biersack. Survey of error recovery       techniques for IP-based audio-visual multicast applications.       IEEE Network, 11(6):24--36, November/December 1997.   [6] S. Floyd and K. Fall. Promoting the use  of end-to-end       congestion control in the internet. Submitted to IEEE/ACM       Transactions on Networking, February 1998.   [7] S. Floyd, V. Jacobson, S. McCanne, C.-G. Liu, and L. Zhang.       A reliable multicast framework for light-weight sessions and       applications level framing. IEEE/ACM Transactions on Networking,       1995.   [8] M. Handley.   An examination of Mbone performance.  USC/ISI       Research Report:  ISI/RR-97-450, April 1997.   [9] M. Handley and J. Crowcroft.   Network text editor (NTE): A       scalable shared text editor for the Mbone.   In Proceedings       ACM SIGCOMM'97, Cannes, France, September 1997.  [10] V. Hardman, M. A. Sasse, M. Handley, and  A. Watson.       Reliable audio for use over the Internet.    In Proceedings       of INET'95, 1995.  [11] I. Kouvelas, O. Hodson, V. Hardman, and J.  Crowcroft.       Redundancy control in real-time Internet audio conferencing.       In Proceedings of AVSPN'97, Aberdeen, Scotland, September 1997.  [12] S. McCanne, V. Jacobson, and M. Vetterli.   Receiver-driven       layered multicast.  In Proceedings ACM SIGCOMM'96, Stanford,       CA., August 1996.  [13] J. Nonnenmacher, E. Biersack, and D. Towsley.   Parity-based       loss recovery for reliable multicast transmission. In       Proceedings ACM SIGCOMM'97, Cannes, France, September 1997.  [14] P. Parnes.   RTP extension for scalable reliable multicast,       Work in Progress.Perkins & Hodson             Informational                     [Page 10]

RFC 2354         Options for Repair of Streaming Media         June 1998  [15] 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.  [16] J.L. Ramsey. Realization of optimum interleavers. IEEE Transactions       on Information Theory, IT-16:338--345, May 1970.  [17] J. Rosenberg and H. Schulzrinne. An A/V profile extension for       generic forward error correction in RTP.  Work in Progress.  [18] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,       "RTP: A transport protocol for real-time applications",RFC 1889, January 1996.  [19] L. Vicisano, L. Rizzo, and Crowcroft J.  TCP-like congestion       control for layered multicast data transfer.  In Proceedings       IEEE INFOCOM'98, 1998.  [20] R. X. Xu, A. C. Myers, H. Zhang,  and R. Yavatkar.       Resilient multicast support for continuous media applications.       In Proceedingsof the 7th International Workshop on Network and       Operating Systems Support for Digital Audio and Video       (NOSSDAV'97), Washington University in St. Louis, Missouri,       May 1997.  [21] M. Yajnik, J. Kurose, and D. Towsley. Packet loss correlation       in the Mbone multicast network. In Proceedings IEEE Global       Internet Conference, November 1996.Perkins & Hodson             Informational                     [Page 11]

RFC 2354         Options for Repair of Streaming Media         June 1998Full 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.Perkins & Hodson             Informational                     [Page 12]