RFC2354 - Options for Repair of Streaming Media

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].
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
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
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
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.
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
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:
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
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.
[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.
[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", RFC2198, 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",
RFC1889, 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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