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.

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.

Full Copyright Statement

This document and translations of it may be copied and furnished to

others, and derivative works that comment on or otherwise explain it

or assist in its implementation may be prepared, copied, published

and distributed, in whole or in part, without restriction of any

kind, provided that the above copyright notice and this paragraph are

included on all such copies and derivative works. However, this

document itself may not be modified in any way, such as by removing

the copyright notice or references to the Internet Society or other

Internet organizations, except as needed for the purpose of

developing Internet standards in which case the procedures for

copyrights defined in the Internet Standards process must be

followed, or as required to translate it into languages other than

English.

The limited permissions granted above are perpetual and will not be

revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an

"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING

TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING

BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION

HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF

MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.