path: root/doc/rfc/rfc2354.txt

Network Working Group                                         C. Perkins
Request for Comments: 2354                                     O. Hodson
Category: Informational                        University College London
                                                               June 1998


                 Options for Repair of Streaming Media

Status 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].






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2  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 severely



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   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 and



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   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-specific



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   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.




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   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 increase



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   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:



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    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 in



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   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.uk

References

   [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.




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   [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.





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  [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.























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Full Copyright Statement

   Copyright (C) The Internet Society (1998).  All Rights Reserved.

   This document and translations of it may be copied and furnished to
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   The limited permissions granted above are perpetual and will not be
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