Network Working Group C. Perkins
Request for Comments: 2198 I. Kouvelas
Category: Standards Track O. Hodson
V. Hardman
University College London
M. Handley
ISI
J.C. Bolot
A. Vega-Garcia
S. Fosse-Parisis
INRIA Sophia Antipolis
September 1997
RTP Payload for Redundant Audio Data
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Abstract
This document describes a payload format for use with the real-time
transport protocol (RTP), version 2, for encoding redundant audio
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.
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. This 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 consecutively lost packets 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
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 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. That packet then becomes a
container, encapsulating multiple payloads into a single RTP packet.
Such a scheme is flexible, since any amount of redundancy may be
encapsulated within a single packet. There is, however, a small
overhead since each encapsulated payload 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 key Words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC2119 [7].
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. If multiple redundant encodings are
carried in a packet, they should correspond to different time
intervals: there is no reason to include multiple copies of data for
a single time interval within a packet.
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The bits in the header are specified as follows:
F: 1 bit First 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 bits RTP payload type for this block.
timestamp offset: 14 bits Unsigned 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 bits Length in bytes of the corresponding data
block excluding header.
It is noted that the use of an unsigned timestamp offset 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 (ie: audio samples,
with the same clock rate as the primary). The implication of this is
that the redundant encoding MUST be sampled at the same rate as the
primary.
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.
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.
The choice of encodings used should reflect the bandwidth
requirements of those encodings. It is expected that the redundant
encoding shall use significantly less bandwidth that the primary
encoding: the exception being the case where the primary is very
low-bandwidth and has high processing requirement, in which case a
copy of the primary MAY be used as the redundancy. The redundant
encoding MUST NOT be higher bandwidth than the primary.
The use of multiple levels of redundancy is rarely necessary.
However, in those cases which require it, the bandwidth required by
each level of redundancy is expected to be significantly less than
that of the previous level.
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 (using the RTP audio/video profile [3]) is:
m=audio 12345 RTP/AVP 121 0 5
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 monaural. 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. Note that the additional
payload formats defined in the "m=" field may themselves be dynamic
payload types, and if so a number of additional "a=" attributes may
be required to describe these dynamic payload types.
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 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.
Whilst the addition of low-bandwidth redundancy to an audio 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 redundancy will increase network congestion, and
hence packet loss, leading to a worsening of the problem which the
use of redundancy was intended to solve. At its worst, this can lead
to excessive network congestion and may constitute a denial of
service attack.
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), as defined in the RTP audio/video profile [3] is
illustrated:
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=2PX 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) +
/ /
+ +
+ +
+ +---------------+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
8 Authors' Addresses
Colin Perkins/Isidor Kouvelas/Orion Hodson/Vicky Hardman
Department of Computer Science
University College London
London WC1E 6BT
United Kingdom
EMail: {c.perkinsi.kouvelaso.hodsonv.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: {bolotavegasfosse}@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] Schulzrinne, H., Casner, S., Frederick R., and V. Jacobson, "RTP:
A Transport Protocol for Real-Time Applications", RFC1889, January
1996.
[3] Schulzrinne, H., "RTP Profile for Audio and Video Conferences
with Minimal Control", RFC1890, 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; ACM Multimedia Systems,
1997
[6] Handley, M., and V. Jacobson, "SDP: Session Description Protocol
(draft 03.2)", Work in Progress.
[7] Bradner, S., "Key words for use in RFCs to indicate requirement
levels", RFC2119, March 1997.
