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Internet Engineering Task Force                                      AVT WG
INTERNET-DRAFT                                       M. Handley, C. Perkins
draft-ietf-avt-rtp-format-guidelines-04.txt                      ACIRI, UCL
                                                              15th Oct 1999
                                                          Expires: Apr 1999

      Guidelines for Writers of RTP Payload Format Specifications

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.

Status of this Memo

This document is an Internet-Draft and is in full conformance with all
provisions of Section 10 of RFC2026.  Internet-Drafts are working docu-
ments of the Internet Engineering Task Force (IETF), its areas, and its
working groups.  Note that other groups may also distribute working doc-
uments as Internet-Drafts.

Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time.  It is inappropriate to use Internet- Drafts as reference material
or to cite them other than as ``work in progress.''

The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt

The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.

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.

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INTERNET-DRAFT                                             15th Oct 1999

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 regen-
erating it), and to sending new data which supersedes the missing data.
The application only has this choice if the transport protocol is deal-
ing 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 con-
tains sufficient information to be processed by the receiver immedi-
ately.  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.  How-
ever, 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 appli-
cation 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 sig-
nificantly 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

Handley/Perkins                                                 [Page 2]


INTERNET-DRAFT                                             15th Oct 1999

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 remain-
ing 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 accom-
modate gracefully.

4.  Guidelines

We identify the following requirements of RTP payload format specifica-
tions:

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

Handley/Perkins                                                 [Page 3]


INTERNET-DRAFT                                             15th Oct 1999

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 con-
strain TCP to lower data rates.  For the most part, it is such TCP traf-
fic 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 pur-
pose 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
    crossed 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 correc-
    tion[5] to be applied that cannot be used with IP-level fragmenta-
    tion.

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 corre-
sponds 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

Handley/Perkins                                                 [Page 4]


INTERNET-DRAFT                                             15th Oct 1999

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 arbi-
trarily 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 pack-
ets 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 mac-
roblocks, 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 nor-
mally be found by parsing from the start of the GOB, and we can packe-
tise 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 predic-
tor 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 compro-
mise 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

Handley/Perkins                                                 [Page 5]


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    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-ses-
    sion.  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 this problems 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 sec-
tion of the packet. The RTP header extension should not be used to con-
vey payload specific information ([9],section 5.3) since this is ineffi-
cient in its use of bandwidth; requires the definition of a new RTP pro-
file 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.

Handley/Perkins                                                 [Page 6]


INTERNET-DRAFT                                             15th Oct 1999

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 func-
tions 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 informa-
tion, 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 mag-
nification 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 envi-
ronments.

Designers of new codecs for use with RTP should consider making the out-
put 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 sacri-
ficing 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.

Handley/Perkins                                                 [Page 7]


INTERNET-DRAFT                                             15th Oct 1999

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 pay-
load 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 for-
mats, and should be careful to specify the correct behaviour when anoma-
lous 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.  How-
ever, 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 net-
work 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
mjh@aciri.org

Colin Perkins
Dept of Computer Science,
University College London,
Gower Street,
London WC1E 6BT, UK.

Handley/Perkins                                                 [Page 8]


INTERNET-DRAFT                                             15th Oct 1999

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]  S. Casner, V. Jacobson, ``Compressing IP/UDP/RTP Headers for Low-
     Speed Serial Links'', RFC 2508.

[2]  D. Clark, 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 Com-
     munication Review, 27(3), July 1997.

[5]  J. Nonnenmacher, E. Biersack, Don Towsley, ``Parity-Based Loss
     Recovery for Reliable Multicast Transmission'', Proc ACM Sigcomm
     '97, Cannes, France, 1997.

[6]  J. Padhye, V. Firoiu, D. Towsley, J.  Kurose, ``Modeling TCP
     Throughput: A Simple Model and its Empirical Validation'', Proc.
     ACM Sigcomm 1998.

[7]  C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C.
     Bolot, A. Vega-Garcia, S. Fosse-Parisis, ``RTP Payload for Redun-
     dant Audio Data'', RFC 2198.

[8]  K. K. Ramakrishnan, Sally Floyd, ``A Proposal to add Explicit Con-
     gestion Notification (ECN) to IP'' RFC 2481, Jan 1999.

[9]  H.Schulzrinne, S.Casner, R.Frederick, V. Jacobson, "Real-Time
     Transport Protocol", RFC1889, Jan 1996.

Handley/Perkins                                                 [Page 9]


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