RFC2736 - Guidelines for Writers of RTP Payload Format Specifications

Network Working Group M. Handley

Request for Comments: 2736 ACIRI

BCP: 36 C. Perkins

Category: Best Current Practice UCL

December 1999

Guidelines for Writers of RTP Payload Format Specifications

Status of this Memo

This document specifies an Internet Best Current Practices for the

Internet Community, and requests discussion and suggestions for

improvements. Distribution of this memo is unlimited.

Abstract

This document provides general guidelines aimed at assisting the

authors of RTP Payload Format specifications in deciding on good

formats. These guidelines attempt to capture some of the eXPerience

gained with RTP as it evolved during its development.

1. Introduction

This document provides general guidelines aimed at assisting the

authors of RTP [9] Payload Format specifications in deciding on good

formats. These guidelines attempt to capture some of the experience

gained with RTP as it evolved during its development.

The principles outlined in this document are applicable to almost all

data types, but are framed in examples of audio and video codecs for

clarity.

2. Background

RTP was designed around the concept of Application Level Framing

(ALF), first described by Clark and Tennenhouse [2]. The key argument

underlying ALF is that there are many different ways an application

might be able to cope with misordered or lost packets. These range

from ignoring the loss, to re-sending the missing data (either from a

buffer or by regenerating it), and to sending new data which

supersedes the missing data. The application only has this choice if

the transport protocol is dealing with data in "Application Data

Units" (ADUs). An ADU contains data that can be processed out-of-

order with respect to other ADUs. Thus the ADU is the minimum unit

of error recovery.

The key property of a transport protocol for ADUs is that each ADU

contains sufficient information to be processed by the receiver

immediately. An example is a video stream, wherein the compressed

video data in an ADU must be capable of being decompressed regardless

of whether previous ADUs have been received. Additionally the ADU

must contain "header" information detailing its position in the video

image and the frame from which it came.

Although an ADU need not be a packet, there are many applications for

which a packet is a natural ADU. Such ALF applications have the

great advantage that all packets that are received can be processed

by the application immediately.

RTP was designed around an ALF philosophy. In the context of a

stream of RTP data, an RTP packet header provides sufficient

information to be able to identify and decode the packet irrespective

of whether it was received in order, or whether preceding packets

have been lost. However, these arguments only hold good if the RTP

payload formats are also designed using an ALF philosophy.

Note that this also implies smart, network aware, end-points. An

application using RTP should be aware of the limitations of the

underlying network, and should adapt its transmission to match those

limitations. Our experience is that a smart end-point implementation

can achieve significantly better performance on real IP-based

networks than a naive implementation.

3. Channel Characteristics

We identify the following channel characteristics that influence the

best-effort transport of RTP over UDP/IP in the Internet:

o Packets may be lost

o Packets may be duplicated

o Packets may be reordered in transit

o Packets will be fragmented if they exceed the MTU of the

underlying network

The loss characteristics of a link may vary widely over short time

intervals.

Although fragmentation is not a disastrous phenomenon if it is a rare

occurrence, relying on IP fragmentation is a bad design strategy as

it significantly increases the effective loss rate of a network and

decreases goodput. This is because if one fragment is lost, the

remaining fragments (which have used up bottleneck bandwidth) will

then need to be discarded by the receiver. It also puts additional

load on the routers performing fragmentation and on the end-systems

re-assembling the fragments.

In addition, it is noted that the transit time between two hosts on

the Internet will not be constant. This is due to two effects -

jitter caused by being queued behind cross-traffic, and routing

changes. The former is possible to characterise and compensate for

by using a playout buffer, but the latter is impossible to predict

and difficult to accommodate gracefully.

4. Guidelines

We identify the following requirements of RTP payload format

specifications:

+ A payload format should be devised so that the stream being

transported is still useful even in the presence of a moderate

amount of packet loss.

+ Ideally all the contents of every packet should be possible to be

decoded and played out irrespective of whether preceding packets

have been lost or arrive late.

The first of these requirements is based on the nature of the

Internet. Although it may be possible to engineer parts of the

Internet to produce low loss rates through careful provisioning or

the use of non-best-effort services, as a rule payload formats should

not be designed for these special purpose environments. Payload

formats should be designed to be used in the public Internet with

best effort service, and thus should expect to see moderate loss

rates. For example, a 5% loss rate is not uncommon. We note that

TCP steady state models [3][4][6] indicate that a 5% loss rate with a

1KByte packet size and 200ms round-trip time will result in TCP

achieving a throughput of around 180Kbit/s. Higher loss rates,

smaller packet sizes, or a larger RTT are required to constrain TCP

to lower data rates. For the most part, it is such TCP traffic that

is producing the background loss that many RTP flows must co-exist

with. Without explicit congestion notification (ECN) [8], loss must

be considered an intrinsic property of best-effort parts of the

Internet.

When payload formats do not assume packet loss will occur, they

should state this explicitly up front, and they will be considered

special purpose payload formats, unsuitable for use on the public

Internet without special support from the network infrastructure.

The second of these requirements is more explicit about how RTP

should cope with loss. If an RTP payload format is properly

designed, every packet that is actually received should be useful.

Typically this implies the following guidelines are adhered to:

+ Packet boundaries should coincide with codec frame boundaries.

Thus a packet should normally consist of one or more complete

codec frames.

+ A codec's minimum unit of data should never be packetised so that

it crosses a packet boundary unless it is larger than the MTU.

+ If a codec's frame size is larger than the MTU, the payload format

must not rely on IP fragmentation. Instead it must define its own

fragmentation mechanism. Such mechanisms may involve codec-

specific information that allows decoding of fragments.

Alternatively they might allow codec-independent packet-level

forward error correction [5] to be applied that cannot be used

with IP-level fragmentation.

In the abstract, a codec frame (i.e., the ADU or the minimum size

unit that has semantic meaning when handed to the codec) can be of

arbitrary size. For PCM audio, it is one byte. For GSM audio, a

frame corresponds to 20ms of audio. For H.261 video, it is a Group

of Blocks (GOB), or one twelfth of a CIF video frame.

For PCM, it does not matter how audio is packetised, as the ADU size

is one byte. For GSM audio, arbitrary packetisation would split a

20ms frame over two packets, which would mean that if one packet were

lost, partial frames in packets before and after the loss are

meaningless. This means that not only were the bits in the missing

packet lost, but also that additional bits in neighboring packets

that used bottleneck bandwidth were effectively also lost because the

receiver must throw them away. Instead, we would packetise GSM by

including several complete GSM frames in a packet; typically four GSM

frames are included in current implementations. Thus every packet

received can be decoded because even in the presence of loss, no

incomplete frames are received.

The H.261 specification allows GOBs to be up to 3KBytes long,

although most of the time they are smaller than this. It might be

thought that we should insert a group of blocks into a packet when it

fits, and arbitrarily split the GOB over two or more packets when a

GOB is large. In the first version of the H.261 payload format, this

is what was done. However, this still means that there are

circumstances where H.261 packets arrive at the receiver and must be

discarded because other packets were lost - a loss multiplier effect

that we wish to avoid. In fact there are smaller units than GOBs in

the H.261 bit-stream called macroblocks, but they are not

identifiable without parsing from the start of the GOB. However, if

we provide a little additional information at the start of each

packet, we can reinstate information that would normally be found by

parsing from the start of the GOB, and we can packetise H.261 by

splitting the data stream on macroblock boundaries. This is a less

obvious packetisation for H.261 than the GOB packetisation, but it

does mean that a slightly smarter depacketiser at the receiver can

reconstruct a valid H.261 bitstream from a stream of RTP packets that

has experienced loss, and not have to discard any of the data that

arrived.

An additional guideline concerns codecs that require the decoder

state machine to keep step with the encoder state machine. Many

audio codecs such as LPC or GSM are of this form. Typically they are

loss tolerant, in that after a loss, the predictor coefficients

decay, so that after a certain amount of time, the predictor error

induced by the loss will disappear. Most codecs designed for

telephony services are of this form because they were designed to

cope with bit errors without the decoder predictor state permanently

remaining incorrect. Just packetising these formats so that packets

consist of integer multiples of codec frames may not be optimal, as

although the packet received immediately after a packet loss can be

decoded, the start of the audio stream produced will be incorrect

(and hence distort the signal) because the decoder predictor is now

out of step with the encoder. In principle, all of the decoder's

internal state could be added using a header attached to the start of

every packet, but for lower bit-rate encodings, this state is so

substantial that the bit rate is no longer low. However, a

compromise can usually be found, where a greatly reduced form of

decoder state is sent in every packet, which does not recreate the

encoders predictor precisely, but does reduce the magnitude and

duration of the distortion produced when the previous packet is lost.

Such compressed state is, by definition, very dependent on the codec

in question. Thus we recommend:

+ Payload formats for encodings where the decoder contains internal

data-driven state that attempts to track encoder state should

normally consider including a small additional header that conveys

the most critical elements of this state to reduce distortion

after packet loss.

A similar issue arises with codec parameters, and whether or not they

should be included in the payload format. An example is with a codec

that has a choice of huffman tables for compression. The codec may

use either huffman table 1 or table 2 for encoding and the receiver

needs to know this information for correct decoding. There are a

number of ways in which this kind of information can be conveyed:

o Out of band signalling, prior to media transmission.

o Out of band signalling, but the parameter can be changed mid-

session. This requires synchronization of the change in the media

stream.

o The change is signaled through a change in the RTP payload type

field. This requires mapping the parameter space into particular

payload type values and signalling this mapping out-of-band prior

to media transmission.

o Including the parameter in the payload format. This allows for

adapting the parameter in a robust manner, but makes the payload

format less efficient.

Which mechanism to use depends on the utility of changing the

parameter in mid-session to support application layer adaptation.

However, using out-of-band signalling to change a parameter in mid-

session is generally to be discouraged due to the problem of

synchronizing the parameter change with the media stream.

4.1. RTP Header Extensions

Many RTP payload formats require some additional header information

to be carried in addition to that included in the fixed RTP packet

header. The recommended way of conveying this information is in the

payload section of the packet. The RTP header extension should not be

used to convey payload specific information ([9], section 5.3) since

this is inefficient in its use of bandwidth; requires the definition

of a new RTP profile or profile extension; and makes it difficult to

employ FEC schemes such as, for example, [7]. Use of an RTP header

extension is only appropriate for cases where the extension in

question applies across a wide range of payload types.

4.2. Header Compression

Designers of payload formats should also be aware of the needs of RTP

header compression [1]. In particular, the compression algorithm

functions best when the RTP timestamp increments by a constant value

between consecutive packets. Payload formats which rely on sending

packets out of order, such that the timestamp increment is not

constant, are likely to compress less well than those which send

packets in order. This has most often been an issue when designing

payload formats for FEC information, although some video codecs also

rely on out-of-order transmission of packets at the expense of

reduced compression. Although in some cases such out-of-order

transmission may be the best solution, payload format designers are

encourage to look for alternative solutions where possible.

5. Summary

Designing packet formats for RTP is not a trivial task. Typically a

detailed knowledge of the codec involved is required to be able to

design a format that is resilient to loss, does not introduce loss

magnification effects due to inappropriate packetisation, and does

not introduce unnecessary distortion after a packet loss. We believe

that considerable effort should be put into designing packet formats

that are well tailored to the codec in question. Typically this

requires a very small amount of processing at the sender and

receiver, but the result can be greatly improved quality when

operating in typical Internet environments.

Designers of new codecs for use with RTP should consider making the

output of the codec "naturally packetizable". This implies that the

codec should be designed to produce a packet stream, rather than a

bit-stream; and that that packet stream contains the minimal amount

of redundancy necessary to ensure that each packet is independently

decodable with minimal loss of decoder predictor tracking. It is

recognised that sacrificing some small amount of bandwidth to ensure

greater robustness to packet loss is often a worthwhile tradeoff.

It is hoped that, in the long run, new codecs should be produced

which can be directly packetised, without the trouble of designing a

codec-specific payload format.

It is possible to design generic packetisation formats that do not

pay attention to the issues described in this document, but such

formats are only suitable for special purpose networks where packet

loss can be avoided by careful engineering at the network layer, and

are not suited to current best-effort networks.

6. Security Considerations

The guidelines in this document result in RTP payload formats that

are robust in the presence of real world network conditions.

Designing payload formats for special purpose networks that assume

negligable loss rates will normally result in slightly better

compression, but produce formats that are more fragile, thus

rendering them easier targets for denial-of-service attacks.

Designers of payload formats should pay close attention to possible

security issues that might arise from poor implementations of their

formats, and should be careful to specify the correct behaviour when

anomalous conditions arise. Examples include how to process illegal

field values, and conditions when there are mismatches between length

fields and actual data. Whilst the correct action will normally be

to discard the packet, possible such conditions should be brought to

the attention of the implementor to ensure that they are trapped

properly.

The RTP specification covers encryption of the payload. This issue

should not normally be dealt with by payload formats themselves.

However, certain payload formats spread information about a

particular application data unit over a number of packets, or rely on

packets which relate to a number of application data units. Care must

be taken when changing the encryption of such streams, since such

payload formats may constrain the places in a stream where it is

possible to change the encryption key without exposing sensitive

data.

Designers of payload formats which include FEC should be aware that

the automatic addition of FEC in response to packet loss may increase

network congestion, leading to a worsening of the problem which the

use of FEC was intended to solve. Since this may, at its worst,

constitute a denial of service attack, designers of such payload

formats should take care that appropriate safeguards are in place to

prevent abuse.

Authors' Addresses

Mark Handley

AT&T Center for Internet Research at ICSI,

International Computer Science Institute,

1947 Center Street, Suite 600,

Berkeley, CA 94704, USA

EMail: mjh@aciri.org

Colin Perkins

Dept of Computer Science,

University College London,

Gower Street,

London WC1E 6BT, UK.

EMail: C.Perkins@cs.ucl.ac.uk

Acknowledgments

This document is based on experience gained over several years by

many people, including Van Jacobson, Steve McCanne, Steve Casner,

Henning Schulzrinne, Thierry Turletti, Jonathan Rosenberg and

Christian Huitema amongst others.

References

[1] Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP Headers for

Low-Speed Serial Links", RFC2508, February 1999.

[2] D. Clark and D. Tennenhouse, "Architectural Considerations for

a New Generation of Network Protocols" Proc ACM Sigcomm 90.

[3] J. Mahdavi and S. Floyd. "TCP-friendly unicast rate-based flow

control". Note sent to end2end-interest mailing list, Jan 1997.

[4] M. Mathis, J. Semske, J. Mahdavi, and T. Ott. "The macro-scopic

behavior of the TCP congestion avoidance algorithm". Computer

Communication Review, 27(3), July 1997.

[5] J. Nonnenmacher, E. Biersack, Don Towsley, "Parity-Based Loss

Recovery for Reliable Multicast Transmission", Proc ACM Sigcomm

[6] J. Padhye, V. Firoiu, D. Towsley, J. Kurose, "Modeling TCP

Throughput: A Simple Model and its Empirical Validation", Proc.

ACM Sigcomm 1998.

[7] 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.

[8] Ramakrishnan, K. and S. Floyd, "A Proposal to add Explicit

Congestion Notification (ECN) to IP", RFC2481, January 1999.

[9] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,

"Real-Time Transport Protocol", RFC1889, January 1996.

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