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Versions: 00                                                            
INTERNET-DRAFT                               3 August 1998


                                             Colin Perkins
                                           Isidor Kouvelas
                                              Orion Hodson
                                             Vicky Hardman
                                 University College London

                                              Mark Handley
                                                       ISI

                                    Jean-Chrysostome Bolot
                                        Andres Vega-Garcia
                                       Sacha Fosse-Parisis
                                    INRIA Sophia Antipolis



            RTP Payload for Redundant Audio Data
           draft-ietf-avt-redundancy-revised-00.txt


Status of this Memo


This document is an Internet-Draft.  Internet-Drafts are working documents
of the Internet Engineering Task Force (IETF), its areas, and its working
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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 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.

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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.  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 [4-6] 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,

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    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

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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 [8].

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 in redundant stream 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.


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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|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 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 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


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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.

At the start of talkspurts, there is not enough information to send             |
redundant information. In this case the largest offset in the talkspurt         |
SHOULD be advertised to enable receiving applications to provision              |
sufficient buffer space to ensure all of the redundant data received            |
can be used.  The advertisement is communicated by a redundant header           |
showing zero block length and the maximum timestamp offset required.            |

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

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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) [7].  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.

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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:

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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/Orion Hodson/Vicky Hardman
Department of Computer Science
University College London
London WC1E 6BT
United Kingdom
Email:  {c.perkins|i.kouvelas|o.hodson|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|sfosse}@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; ACM Multimedia Systems, 1997

[6] I. Kouvelas, O.Hodson, V.Hardman and J. Crowcroft; Redundancy Control
    in Real-Time Internet Audio Conferencing Proceedings AVSPN 1997,
    Aberdeen, Scotland, September 1997

[7] M. Handley and V. Jacobson; SDP: Session Description Protocol;
    RFC2327, April 1998.


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[8] S. Bradner, "Key words for use in RFCs to indicate requirement
    levels"; RFC2119, March 1997.

















































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