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INTERNET-DRAFT 26 March 1997
Expire in six months
Colin Perkins
Isidor Kouvelas
Vicky Hardman
UniversityCollege London
Mark Handley
ISI
Jean-Chrysostome Bolot
Andres Vega-Garcia
Sacha Fosse-Parisis
INRIA Sophia Antipolis
RTP Payload for Redundant Audio Data
draft-perkins-rtp-redundancy-03.txt
Status of this Memo
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Abstract
This document describes a payload format for use with the
real-time transport protocol (RTP), version 2, for encoding
redundant data. The primary motivation for the scheme described
herein is the development of audio conferencing tools for
use with lossy packet networks such as the Internet Mbone,
although this scheme is not limited to such applications.
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1 Introduction
If multimedia conferencing is to become widely used by the Internet
Mbone community, users must perceive the quality to be sufficiently
good for most applications. We have identified a number of problems
which impair the quality of conferences, the most significant of
which is packet loss over the Internet Mbone. Packet loss is a persistent
problem, particularly given the increasing popularity, and therefore
increasing load, of the Internet. The disruption of speech intelligibility
even at low loss rates which is currently experienced may convince
a whole generation of users that multimedia conferencing over the
Internet is not viable. The addition of redundancy to the data stream
is offered as a solution [1]. If a packet is lost then the missing
information may be reconstructed at the receiver from the redundant
data that arrives in the following packet(s), provided that the average
number of consequutively lost packet is small. Recent work [4,5]
shows that packet loss patterns in the Internet are such that this
scheme typically functions well.
This document describes an RTP payload format for the transmission
of audio data encoded in such a redundant fashion. Section 2 presents
the requirements and motivation leading to the definition of this
payload format, and does not form part of the payload format definition.
Sections 3 onwards define the RTP payload format for redundant audio
data.
2 Requirements/Motivation
The requirements for a redundant encoding scheme under RTP are as
follows:
o Packets have to carry a primary encoding and one or more redundant
encodings.
o As a multitude of encodings may be used for redundant information,
each block of redundant encoding has to have an encoding type
identifier.
o As the use of variable size encodings is desirable, each encoded
block in the packet has to have a length indicator.
o The RTP header provides a timestamp field that corresponds to
the time of creation of the encoded data. When redundant encodings
are used this timestamp field can refer to the time of creation
of the primary encoding data. Redundant blocks of data will
correspond to different time intervals than the primary data,
and hence each block of redundant encoding will require its own
timestamp. To reduce the number of bytes needed to carry the
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timestamp, it can be encoded as the difference of the timestamp
for the redundant encoding and the timestamp of the primary.
There are two essential means by which redundant audio may be added
to the standard RTP specification: a header extension may hold the
redundancy, or one, or more, additional payload types may be defined.
Including all the redundancy information for a packet in a header
extension would make it easy for applications that do not implement
redundancy to discard it and just process the primary encoding data.
There are, however, a number of disadvantages with this scheme:
o There is a large overhead from the number of bytes needed for
the extension header (4) and the possible padding that is needed
at the end of the extension to round up to a four byte boundary
(up to 3 bytes). For many applications this overhead is unacceptable.
o Use of the header extension limits applications to a single redundant
encoding, unless further structure is introduced into the extension.
This would result in further overhead.
For these reasons, the use of RTP header extension to hold redundant
audio encodings is disregarded.
The RTP profile for audio and video conferences [3] lists a set of
payload types and provides for a dynamic range of 32 encodings that
may be defined through a conference control protocol. This leads
to two possible schemes for assigning additional RTP payload types
for redundant audio applications:
1.A dynamic encoding scheme may be defined, for each combination
of primary/redundant payload types, using the RTP dynamic payload
type range.
2.A single fixed payload type may be defined to represent a packet
with redundancy. This may then be assigned to either a static
RTP payload type, or the payload type for this may be assigned
dynamically.
It is possible to define a set of payload types that signify a particular
combination of primary and secondary encodings for each of the 32
dynamic payload types provided. This would be a slightly restrictive
yet feasible solution for packets with a single block of redundancy
as the number of possible combinations is not too large. However
the need for multiple blocks of redundancy greatly increases the
number of encoding combinations and makes this solution not viable.
A modified version of the above solution could be to decide prior
to the beginning of a conference on a set a 32 encoding combinations
that will be used for the duration of the conference. All tools
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in the conference can be initialized with this working set of encoding
combinations. Communication of the working set could be made through
the use of an external, out of band, mechanism. Setup is complicated
as great care needs to be taken in starting tools with identical
parameters. This scheme is more efficient as only one byte is used
to identify combinations of encodings.
It is felt that the complication inherent in distributing the mapping
of payload types onto combinations of redundant data preclude the
use of this mechanism.
A more flexible solution is to have a single payload type which signifies
a packet with redundancy and have each of the encoding blocks in
the packet contain it's own payload type field: such a packet acts
as a container, encapsulating multiple packets into one.
Such a scheme is flexible, since any number of redundant encodings
may be enclosed within a single packet. There is, however, a small
overhead since each encapsulated packet must be preceded by a header
indicating the type of data enclosed. This is the preferred solution,
since it is both flexible, extensible, and has a relatively low overhead.
The remainder of this document describes this solution.
3 Payload Format Specification
The assignment of an RTP payload type for this new packet format
is outside the scope of this document, and will not be specified
here. It is expected that the RTP profile for a particular class
of applications will assign a payload type for this encoding, or
if that is not done then a payload type in the dynamic range shall
be chosen.
An RTP packet containing redundant data shall have a standard RTP
header, with payload type indicating redundancy. The other fields
of the RTP header relate to the primary data block of the redundant
data.
Following the RTP header are a number of additional headers, defined
in the figure below, which specify the contents of each of the encodings
carried by the packet. Following these additional headers are a
number of data blocks, which contain the standard RTP payload data
for these encodings. It is noted that all the headers are aligned
to a 32 bit boundary, but that the payload data will typically not
be aligned.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| block PT | timestamp offset | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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The bits in the header are specified as follows:
F: 1 bitFirst bit in header indicates whether another header block
follows. If 1 further header blocks follow, if 0 this is the
last header block.
block PT: 7 bitsRTP payload type for this block.
timestamp offset: 14 bitsUnsigned offset of timestamp of this block
relative to timestamp given in RTP header. The use of an unsigned
offset implies that redundant data must be sent after the primary
data, and is hence a time to be subtracted from the current
timestamp to determine the timestamp of the data for which this
block is the redundancy.
block length: 10 bitsLength in bytes of the corresponding data
block excluding header.
It is noted that the use of an unsigned timestamp offest limits the
use of redundant data slightly: it is not possible to send redundancy
before the primary encoding. This may affect schemes where a low
bandwidth coding suitable for redundancy is produced early in the
encoding process, and hence could feasibly be transmitted early. However,
the addition of a sign bit would unacceptably reduce the range of
the timestamp offset, and increasing the size of the field above
14 bits limits the block length field. It seems that limiting redundancy
to be transmitted after the primary will cause fewer problems than
limiting the size of the other fields.
The timestamp offset for a redundant block is measured in the same
units as the timestamp of the primary encoding. This does not necessarily
imply that the redundant block is sampled at the same rate as the
primary, and if the sampling rates are different, conversion must
be performed, with the offset being rounded to the nearest sampling
instant of the redundant audio clock.
It is further noted that the block length and timestamp offset are
10 bits, and 14 bits respectively; rather than the more obvious 8
and 16 bits. Whilst such an encoding complicates parsing the header
information slightly, and adds some additional processing overhead,
there are a number of problems involved with the more obvious choice:
An 8 bit block length field is sufficient for most, but not all,
possible encodings: for example 80ms PCM and DVI audio packets comprise
more than 256 bytes, and cannot be encoded with a single byte length
field. It is possible to impose additional structure on the block
length field (for example the high bit set could imply the lower
7 bits code a length in words, rather than bytes), however such schemes
are complex. The use of a 10 bit block length field retains simplicity
and provides an enlarged range, at the expense of a reduced range
of timestamp values. A 14 bit timestamp value does, however, allow
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for 4.5 complete packets delay with 48KHz audio, more at lower sampling
rates, and it is felt that this is sufficient.
The primary encoding block header is placed last in the packet. It
is therefore possible to omit the timestamp and block-length fields
from the header of this block, since they may be determined from
the RTP header and overall packet length. The header for the primary
(final) block comprises only a zero F bit, and the block payload
type information, a total of 8 bits. This is illustrated in the
figure below:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0| Block PT |
+-+-+-+-+-+-+-+-+
The final header is followed, immediately, by the data blocks, stored
in the same order as the headers. There is no padding or other delimiter
between the data blocks, and they are typically not 32 bit aligned.
Again, this choice was made to reduce bandwidth overheads, at the
expense of additional decoding time.
4 Limitations
The RTP marker bit is not preserved for redundant data blocks. Hence
if the primary (containing this marker) is lost, the marker is lost.
It is believed that this will not cause undue problems: even if
the marker bit was transmitted with the redundant information, there
would still be the possibility of its loss, so applications would
still have to be written with this in mind.
In addition, CSRC information is not preserved for redundant data.
The CSRC data in the RTP header of a redundant audio packet relates
to the primary only. Since CSRC data in an audio stream is expected
to change relatively infrequently, it is recommended that applications
which require this information assume that the CSRC data in the RTP
header may be applied to the reconstructed redundant data.
5 Relation to SDP
When a redundant payload is used, it may need to be bound to an
RTP dynamic payload type. This may be achieved through any out-of-band
mechanism, but one common way is to communicate this binding using
the Session Description Protocol (SDP) [6]. SDP has a mechanism
for binding a dynamic payload types to particular codec, sample rate,
and number of channels using the ``rtpmap'' attribute. An example
of its use is:
m=audio 12345 RTP/AVP 121 0 5
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a=rtpmap:121 red/8000/1
This specifies that an audio stream using RTP is using payload types
121 (a dynamic payload type), 0 (PCM u-law) and 5 (DVI). The ``rtpmap''
attribute is used to bind payload type 121 to codec ``red'' indicating
this codec is actually a redundancy frame, 8KHz, and monoaural. When
used with SDP, the term ``red'' is used to indicate the redundancy
format discussed in this document.
In this case the additional formats of PCM and DVI are specified.
The receiver must therefore be prepared to use these formats. Such
a specification means the sender will send redundancy by default,
but also may send PCM or DVI. However, with a redundant payload we
additionally take this to mean that no codec other than PCM or DVI
will be used in the redundant encodings.
To receive a redundant stream, this is all that is required. However
to send a redundant stream, the sender needs to know which codecs
are recommended for the primary and secondary (and tertiary, etc)
encodings. This information is specific to the redundancy format,
and is specified using an additional attribute ``fmtp'' which conveys
format-specific information. A session directory does not parse the
values specified in an fmtp attribute but merely hands it to the
media tool unchanged. For redundancy, we define the format parameters
to be a slash ``/'' separated list of RTP payload types.
Thus a complete example is:
m=audio 12345 RTP/AVP 121 0 5
a=rtpmap:121 red/8000/1
a=fmtp:121 0/5
This specifies that the default format for senders is redundancy
with PCM as the primary encoding and DVI as the secondary encoding.
Encodings cannot be specified in the fmtp attribute unless they are
also specified as valid encodings on the media (``m='') line.
6 Security Considerations
RTP packets containing redundant information are subject to the security
considerations discussed in the RTP specification [2], and any appropriate
RTP profile (for example [3]). This implies that confidentiality
of the media streams is achieved by encryption.
Encryption of a redundant data-stream may occur in two ways:
1.The entire stream is to be secured, and all participants are
expected to have keys to decode the entire stream. In this
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case, nothing special need be done, and encryption is performed
in the usual manner.
2.A portion of the stream is to be encrypted with a different
key to the remainder. In this case a redundant copy of the
last packet of that portion cannot be sent, since there is no
following packet which is encrypted with the correct key in which
to send it. Similar limitations may occur when enabling/disabling
encryption.
The choice between these two is a matter for the encoder only. Decoders
can decrypt either form without modification.
7 Example Packet
An RTP audio data packet containing a DVI4 (8KHz) primary, and a
single block of redundancy encoded using 8KHz LPC (both 20ms packets)
is illustrated:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| PT | sequence number of primary |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| block PT=7 | timestamp offset | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| block PT=5 | |
+-+-+-+-+-+-+-+-+ +
| |
+ LPC encoded redundant data (PT=7) +
| (14 bytes) |
+ +---------------+
| | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| |
+ +
| |
+ +
| |
+ +
| DVI4 encoded primary data (PT=5) |
+ (84 bytes, not to scale) +
/ /
+ +
| |
+ +
| |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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8 Author's Addresses
Colin Perkins/Isidor Kouvelas/Vicky Hardman
Department of Computer Science
University College London
London WC1E 6BT
United Kingdom
Email: {c.perkins|i.kouvelas|v.hardman}@cs.ucl.ac.uk
Mark Handley
USC Information Sciences Institute
c/o MIT Laboratory for Computer Science
545 Technology Square
Cambridge, MA 02139, USA
Email: mjh@isi.edu
Jean-Chrysostome Bolot/Andres Vega-Garcia/Sacha Fosse-Parisis
INRIA Sophia Antipolis
2004 Route des Lucioles, BP 93
06902 Sophia Antipolis
France
Email: {bolot|avega}@sophia.inria.fr
9 References
[1] V.J. Hardman, M.A. Sasse, M. Handley and A. Watson; Reliable
Audio for Use over the Internet; Proceedings INET'95, Honalulu, Oahu,
Hawaii, September 1995. http://www.isoc.org/in95prc/
[2] H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson; RTP:
A Transport Protocol for Real-Time Applications; RFC 1889, January
1996
[3] H. Schulzrinne; RTP Profile for Audio and Video Conferences with
Minimal Control; RFC 1890, January 1996
[4] M. Yajnik, J. Kurose and D. Towsley; Packet loss correlation
in the MBone multicast network; IEEE Globecom Internet workshop, London,
November 1996
[5] J.-C. Bolot and A. Vega-Garcia; The case for FEC-based error
control for packet audio in the Internet; Multimedia Systems, 1997
[6] M. Handley and V. Jacobson; SDP: Session Description Protocol
(draft 03.2) draft-ietf-mmusic-sdp-03.txt, November 1996
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