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Versions: (draft-gentric-avt-profile) 00 01 02 03 04 05 06 RFC 1890

Internet Engineering Task Force          Audio-Video Transport Working Group
INTERNET-DRAFT                                                H. Schulzrinne
draft-ietf-avt-profile-03.txt                         AT&T Bell Laboratories
                                                            October 20, 1993
                                                          Expires:  12/31/93

    Sample Profile and Encodings for the Use of RTP for Audio and Video
                      Conferences with Minimal Control

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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Distribution of this document is unlimited.


     This note describes a profile for the use of the real-time
    transport protocol (RTP) and the associated control protocol, RTCP,
    within audio and video multiparticipant conferences with minimal
    control.  It provides interpretations of generic fields within the
    RTP specification suitable for audio and video conferences.   In
    particular, this document defines a set of default mappings from
    format index to encodings.
     The document also describes how audio and video data may be
    carried within RTP. It defines a set of standard encodings and
    their names when used within RTP. However, the definitions are
    independent of the particular transport mechanism used.    The
    descriptions provide pointers to reference implementations and
    the detailed standards.    This document is meant as an aid
    for implementors of audio, video and other real-time multimedia
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1 Introduction                                                            2

2 Demultiplexing                                                          3

3 Audio                                                                   3

  3.1 Encoding-independent recommendations . . . . . . . . . . . . . . . 3

  3.2 Recommended Audio Encodings. . . . . . . . . . . . . . . . . . . . 4

  3.3 The RTCP FMT Option for Audio. . . . . . . . . . . . . . . . . . . 6

  3.4 Port Assignment. . . . . . . . . . . . . . . . . . . . . . . . . . 7

4 Video                                                                   8

  4.1 The RTCP FMT Option for Video. . . . . . . . . . . . . . . . . . . 9

  4.2 Port Assignment. . . . . . . . . . . . . . . . . . . . . . . . . . 9

5 Miscellaneous                                                          10

6 Address of Author                                                      10

1 Introduction

This profile defines aspects of RTP left unspecified in the RTP protocol
definition (RFC TBD). This profile is intended for the use within audio and
video conferences with minimal session control.  In particular, no support
for the negotiation of parameters or membership control is provided.  Other
profiles may make different choices for the items specified here.   The
profile specifies the use of RTP over unicast and multicast UDP as well
as ST-II. For unicast UDP and ST-II, references to multicast addresses
are to be ignored.   The use of this profile is indicated by the use of
a media-specific well-known port number.   The profile may also be used
with other port numbers.   For example, the use of a particular session
announcement tool could imply use of this profile.

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2 Demultiplexing

For applications which choose to share a single network destination address
and port for both audio and video, the default channel identifier for audio
is 0 and for video is 1.  In that case, the port number for audio is used.
This combination should only be used when it is known that all receiving
applications can properly demultiplex audio and video.

3 Audio

3.1 Encoding-independent recommendations

The following recommendations are default operating parameters.    Ap-
plications  should  be  prepared  to  handle  other  values.     The  ranges
given are meant to give guidance to application writers, allowing a set
of applications conforming to these guidelines to interoperate without
additional negotiation.   These guidelines are not intended to restrict
operating  parameters  for  applications  that  can  negotiate  a  set  of
interoperable parameters, e.g., through a conference control protocol.

For packetized audio, the default packetization interval should have a
duration of 20 ms, unless otherwise noted in Table 1.  The packetization
interval determines the minimum end-to-end delay; longer packets introduce
less header overhead but higher delay and make packet loss more noticeable.
For non-interactive applications such as lectures or links with severe
bandwidth constraints, a higher packetization delay may be appropriate.  For
frame-based encodings (marked as F in the table 1 below) such as LPC, CELP
and GSM, the sender may choose to combine several frame intervals into a
single message.  The receiver can tell the number of frames contained in a
message since the frame duration is defined as part of the encoding.

If multiple channels are used, the left channel information always precedes
the right-channel information.  For more than two channels, the convention
followed by the AIFF-C audio interchange format should be followed.  (The
AIFF-C specification is available by anonymous ftp at ftp.sgi.com in the
file sgi/aiff-c.9.26.91.ps.)  For two-channel stereo, the sequence is left,
right; for three channels, left, right, center; for quadrophonic systems,
front left, front right, rear left, rear right; for four-channel systems,
left, center, right, and surround sound; for six-channel systems left, left
center, center, right, right center and surround sound.

The sampling frequency should be drawn from the set:  8, 11.025, 16, 22.05,
44.1 and 48 kHz.

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3.2 Recommended Audio Encodings

The table 1 shows the names, types (sample vs.  frame oriented), per-channel
bit rates and default sampling frequencies of recommended encodings.  The
list is partially drawn from the document "Recommended practices for
enhancing digital audio compatibility in multimedia systems", published by
the Interactive Multimedia Assocation, Version 3.00, Oct.  1992 (referenced
as [IMA]). The names are for identification only; they correspond to the
names used within the Real-Time Transport Protocol (RTP). Other applications
may choose different namings.  Note that the L16 encoding may be used with
different sampling rates.  The CCITT changed its name in 1993 to ITU-T; to
limit confusion, both old and new name are used.

  name nom.  sampling  rate type frame description
                   kHz  kb/s S/F  ms
  L16              48   768 S           16-bit linear, 2's complement
  L16            44.1 705.6 S
  L16           22.05 352.8 S
  L16          11.025 176.4 S
  G722             16    64 S           CCITT/ITU-T subband ADPCM
  PCMU              8    64 S           CCITT/ITU-T mu-law PCM
  PCMA              8    64 S           CCITT/ITU-T A-law PCM
  G721              8    32 S           CCITT/ITU-T ADPCM
  IDVI              8    32 S           Intel/DVI ADPCM [IMA]
  G723              8    24 S           CCITT/ITU-T ADPCM
  GSM               8    13 F    20    RTE/LTP GSM 06.10
  1016              8   4.8 F    30    CELP

                         Table 1:  Audio encodings

For multi-octet encodings, octets are transmitted in network byte order
(i.e., most significant octet first).

A detailed description of the encodings is given below.  The names shown
(L16, PCMU, etc.)  are limited to four characters and suitable to be used
for identification in protocols such as RTP (RFC TBD).

L16: denotes uncompressed audio data, using 16-bit signed representation
    with 65535 equally divided steps between minimum and maximum signal
    level, ranging from -32768 to 32767.  The value is represented in two's
    complement notation.

PCMU: specified in CCITT/ITU-T recommendation G.711.  Audio data is encoded
    as eight bits per sample, after companding.  Code to convert between
    linear and mu-law companded data is available in the IMA document.

PCMA: specified in CCITT/ITU-T recommendation G.711.  Audio data is encoded

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    as eight bits per sample, after companding.  Code to convert between
    linear and A-law companded data is available in the IMA document.

G721 through G729: specified in the corresponding CCITT/ITU-T recommenda-
    tions.   Reference implementations for G.721 and G.723 are available
    as part of the CCITT/ITU-T Software Tool Library (STL) from the
    ITU General Secretariat, Sales Service, Place du Nations, CH-1211
    Geneve  20,  Switzerland.     The  library  is  covered  by  a  license
    and  is  available  for  anonymous  ftp  on  gaia.cs.umass.edu,  file

GSM: (group speciale mobile) denotes the European GSM 06.10 provisional
    standard for full-rate speech transcoding, prI-ETS 300 036, which
    is based on RPE/LTP (residual pulse excitation/long term prediction)
    coding at a rate of 13 kb/s.  A reference implementation was written by
    Carsten Borman and Jutta Degener (TU Berlin, Germany) and is available
    for anonymous ftp from tub.cs.tu-berlin.de, directory tub/tubmik.

1016: uses code-excited linear prediction (CELP) and is specified in
    Federal Standard FED-STD 1016, published by the Office of Technology
    and Standards, Washington, DC 20305-2010.

    The U. S. DoD's Federal-Standard-1016 based 4800 bps code excited
    linear prediction voice coder version 3.2 (CELP 3.2) Fortran and
    C simulation source codes are available for worldwide distribution
    at no charge (on DOS diskettes, but configured to compile on Sun
    SPARC stations) from:  Bob Fenichel, National Communications System,
    Washington, D.C. 20305, phone +1-703-692-2124, fax +1-703-746-4960.

    Example input and processed speech files,  a technical information
    bulletin, and the official standard "Federal Standard 1016, Telecom-
    munications:   Analog to Digital Conversion of Radio Voice by 4,800
    bit/second Code Excited Linear Prediction (CELP)" are included at no
    charge.  According to Vincent Cate (Carnegie Mellon), the distribution
    is  also  available  for  anonymous  ftp  at  furmint.nectar.cs.cmu.edu
    ( in directory celp.audio.compression.

    The following articles describes the Federal-Standard-1016 4.8-kbps
    CELP coder:

    Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The
    Proposed Federal Standard 1016 4800 bps Voice Coder:  CELP," Speech
    Technology Magazine, April/May 1990, p.  58-64.

    Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The
    Federal Standard 1016 4800 bps CELP Voice Coder," Digital Signal
    Processing, Academic Press, 1991, Vol.  1, No.  3, p.  145-155.

    Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The
    DoD 4.8 kbps Standard (Proposed Federal Standard 1016)," in Advances
    in Speech Coding, ed.   Atal, Cuperman and Gersho, Kluwer Academic

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    Publishers, 1991, Chapter 12, p.  121-133.

    Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The
    Proposed Federal Standard 1016 4800 bps Voice Coder:  CELP," Speech
    Technology Magazine, April/May 1990, p.  58-64.

    Copies of the FS-1016 document are available for $2.50 each from:

    GSA Rm 6654
    7th & D St SW
    Washington, D.C. 20407

DVI: is specified in the "Recommended Practices for Enhancing Digital Audio
    Compatibility in Multimedia Systems", published by the Interactive
    Multimedia Association (IMA), Annapolis, MD. The document also contains
    reference implementations for mu-law to 16-bit, ADPCM and sample rate

For sample-based encodings, a receiver should accept packets representing
between 0 and 200 ms of audio data.(1)   Receivers should be prepared to
accept multi-channel audio, but may choose to only play a single channel.

All block-oriented audio codecs should be able to encode and decode several
consecutive blocks within a single packet.    Since the frame size for
the block-oriented codecs is given, there is no need to use a separate
designation for the same encoding, but with different number of blocks per

3.3 The RTCP FMT Option for Audio

Unless specified with the FMT option, the mapping between the format field
in an RTP packet and audio encodings, sampling rates and channel counts is
specified by Tables 2.

Format values of 31 and below cannot be redefined by FMT options.  In other
words, only values of 32 and above are valid in the format field within an
FMT option.   The receiver is expected to discard RTP packets containing
media data with unknown format field values.  Sites are expected to keep
the mapping between format and encoding constant, so that lost packets
containing FMT options do not lead the receiver to misinterpret media data.
Additional standard encodings may be registered with the Internet Assigned
 1. This restriction allows reasonable buffer sizing for the receiver.

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Numbers Authority (IANA). The format name is intended to describe the format
in an unambiguous way; it is interpreted as a sequence of four ASCII
characters, with uppercase and lowercase characters treated as distinct.
Format names beginning with the letter 'X' are reserved for experimental use
and not subject to registration.  These experimental encodings may be mapped
to format values 32 and above using the FMT option.  Additional standard
mappings to format values of 31 and below may also be registered with IANA.
Registered assignments are published periodically in the Assigned Numbers

Within the FMT option, the format name is followed by a field containing a
channel count and a sample rate field, measured in samples per second.(2)  A
channel count of zero is considered invalid.  A packetization interval of 20
ms or a multiple thereof is suggested as it leads to integral sample counts
for all common sampling rates.

 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|     FMT     |    length     |0|0|   format  |    reserved   |
|                        name of format                         |
|    channels   | sampling rate (Hz)                            |
...  encoding specific parameters                               ...

                 Figure 1:  FMT option for audio encodings

3.4 Port Assignment

ST-II SAP and UDP port 5005 is the default destination for multicast
real-time audio data carried by RTP for this profile.

A fixed port number is useful as it is less likely than a randomly chosen
port number to be already in use by another application at one or more of
the intended destination hosts.   Also, fixed port numbers allow traffic
statistics to be collected and may simplify firewall implementations.   A
single fixed port number requires that hosts allow several processes to use
a single UDP port with different multicast addresses.  (The particular port
number was chosen to lie in the range above 5000 to accomodate port number
 2. Fractional samples per second was considered excessive as the typical
crystal accuraccy of 100 ppm translates into about one Hz or more of
sampling rate inaccuracy.

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                  index encoding sampling rate channels
                         name             (kHz)
                      0 PCMU                 8        1
                      1 1016                 8        1
                      2 G721                 8        1
                      3 GSM                  8        1
                      4 G723                 8        1
                      5 IDVI                 8        1
                     10 L16               44.1        2

                     Table 2:  Standard audio encodings

allocation practice within the Unix operating system, where port numbers
below 1024 can only be used by privileged processes and port numbers between
1024 and 5000 are automatically assigned by the operating system.)

Unicast connections may use the this or a set of mutually agreed-upon port

4 Video

The following video encodings are currently defined, with their abbreviated
names used for identification:

CPV: This encoding, "Compressed Packet Video" is implemented by Concept,
    Bolter, and ViewPoint Systems video codecs.

JPEG: The encoding is specified in ISO Standards DIS 10918-1 and DIS
    10918-2.   The data is formatted according to the JFIF (JPEG File
    Interchange Format) defined by C-Cube Microsystems.

H261: The encoding is specified in CCITT/ITU-T standard H.261.    The
    packetization and RTP-specific properties are described in RFC TBD.

nv: The encoding is implemented in the program 'nv' developed at Xerox PARC
    by Ron Frederick.

CUSM: The encoding is implemented in the program CU-SeeMe developed at
    Cornell University by Dick Cogger, Scott Brim, Tim Dorcey and John

PicW: The encoding is implemented in the program PictureWindow developed at
    Bolt, Beranek and Newman (BBN).

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4.1 The RTCP FMT Option for Video

Unless specified with the RTCP FMT option, the mapping between the format
field in an RTP packet and the video encoding is specified by Tables 3.  The
second paragraph of Section 3.3 applies for video as well.

Within the video FMT option, a one-octet numeric version identifier further
describes the encoding.  Unless otherwise defined, the version identifier
has the value zero.

 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|     FMT     |    length     |0|0|   format  |    reserved   |
|                        name of format                         |
|    version    | encoding-specific parameters                  |
... encoding-specific parameters                                ...

                 Figure 2:  FMT option for video encodings

                                number name
                                26     JPEG
                                27     CUSM
                                28     nv
                                29     PicW
                                30     Bolt
                                31     H261

            Table 3:  Format values for standard video encodings

4.2 Port Assignment

ST-II SAP and UDP port 5006 is the default destination for multicast
real-time video data carried by RTP for this profile.   The remainder of
section 3.4 applies.

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5 Miscellaneous

RTCP messages should be sent periodically, with a period varying randomly
around a set mean to avoid synchronized bursts of RTCP packets.   (For
example, the time between messages could vary uniformly between one half and
1.5 times the mean.)  The average period between transmissions determines
the additional network load due to RTCP packets and also determines how
long it will take a new arrival to discover the identities of the other
conference participants.  The average period should be chosen such that no
more than a small fraction (say, 1%) of the media bandwidth is consumed by
RTCP messages from all sources, with a minimum period of a few seconds.
By scaling the message frequency with the (slowly increasing) number of
observed participants, a new conference participant will quickly inform all
other participants of its arrival and then slow its announcement rate.

6 Address of Author

Henning Schulzrinne
AT&T Bell Laboratories
MH 2A244
600 Mountain Avenue
Murray Hill, NJ 07974-0636
telephone:  +1 908 582 2262
facsimile:  +1 908 582 5809
electronic mail:  hgs@research.att.com

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