📄 rfc2198.txt
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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
Perkins, et. al. Standards Track [Page 1]
RFC 2198 RTP Payload for Redundant Audio Data September 1997
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:
Perkins, et. al. Standards Track [Page 2]
RFC 2198 RTP Payload for Redundant Audio Data September 1997
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.
Perkins, et. al. Standards Track [Page 3]
RFC 2198 RTP Payload for Redundant Audio Data September 1997
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Perkins, et. al. Standards Track [Page 4]
RFC 2198 RTP Payload for Redundant Audio Data September 1997
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
Perkins, et. al. Standards Track [Page 5]
RFC 2198 RTP Payload for Redundant Audio Data September 1997
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
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