Internet DRAFT - draft-ietf-avt-info-repair


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INTERNET-DRAFT                                7 January 1998

                                               Colin Perkins
                                                Orion Hodson
                                   University College London

            Options for Repair of Streaming Media

                    Status of this memo

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    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

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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.

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 [15] 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 recovery schemes send some units multiple times, using different
encoding schemes.  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.

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 [8,9] 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
[5] 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

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cases, a network link is found to be severely overloaded, and large
amount of loss results.

The primary focus of a packet loss 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 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.

4.1.1 Media-Independent FEC

A number of media-independent FEC schemes have been proposed for use with
streamed media.  These techniques add redundant data to a media stream
which is transmitted in separate packets.  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

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 [3] 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.

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Luby et al [8] have discussed applying parity in layers.  The first layer
in their scheme is the parity bits generated from the media.  The second
layer is calculated as the parity bits of the first layer and so on.  This
obviously improves the repair properties, but consumes additional

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 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.  An RTP payload format for
parity-based FEC is defined in [14].  The format is generic, and can
specify many different parity encodings.

A number of FEC schemes exist which are based on higher-order finite
fields.  An example of such are Reed-Solomon (RS) codes which are more
sophisticated and computationally demanding.  These are usually structured
so that they have good burst loss protection.  There has been much work
conducted in this area, and it is believed that a number of streaming
applications use RS codes.

4.1.2 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 [5,6] and has been shown to
perform well.  An RTP payload format for this form of redundancy has been
defined [12].

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+ [2] 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

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most important to protect, and low complexity since each unit is coded once

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 [7], at the expense of additional latency.

4.2 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 [5].  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 [10, 6].

In order to reduce the overhead of retransmission, the retransmitted units
may be piggy-backed onto the ongoing transmission.  This also allows for
the retransmission to be recoded in a different format, to further reduce
the bandwidth overhead.

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 [4], and experiments have been conducted
using this, and with a low-delay variant, STORM [17].  This work
shows the trade-off between latency and quality for retransmission

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based repair schemes, and illustrates that retransmission is an effective
approach to repair for applications which can tolerate the latency.

An RTP profile extension for SRM-style retransmission requests is
described in [11].

4.3 Interleaving

When the unit size is smaller than the packet size, and end-to-end delay is
unimportant, interleaving [13] 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.  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 [12].  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 for example, latency is of
considerably less importance than reception quality.  In this case, the use
of interleaving and/or retransmission based repair is appropriate, with
interleaving being preferred due to its bandwidth efficiency (provided that
approximate repair is acceptable).

In an interactive session (typically defined as a session where the
end-to-end delay is less then 250ms, this includes media coding/decoding,
network transit and host buffering), the delay imposed by the use of
interleaving and retransmission is not acceptable, and a low-latency

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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 packetisation
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.
Whilst the addition of error correction 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 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 [9, 16].  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 behaviour of a TCP stream, and
hence compete fairly with non-real-time traffic.  This is necessary
for stable network behaviour in the presence of much streamed media.

6  Acknowledgements

The authors wish to thank Phil Karn for his helpful comments.

7  Author's Address

Colin Perkins/Orion Hodson
Department of Computer Science
University College London
Gower Street
London WC1E 6BT
United Kingdom

Email:  <c.perkins|o.hodson>

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