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rfc:rfc2354

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].

Perkins & Hodson Informational [Page 1] RFC 2354 Options for Repair of Streaming Media June 1998

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

Perkins & 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 and

Perkins & 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-specific

Perkins & 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 increase

Perkins & 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 in

Perkins & 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.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.

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 1998

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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 or assist in its implementation may be prepared, copied, published
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Perkins & Hodson Informational [Page 12]

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