Internet Engineering Task Force          Audio-Video Transport Working Group
INTERNET-DRAFT                                                H. Schulzrinne
                                                      AT&T Bell Laboratories
                                                               July 19, 1993
                                                          Expires:  10/01/93


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



Status of this Memo


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                                  Abstract


     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
    content index to encodings.



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


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video conferences with minimal session control.   In particular, no  support
for the negotiation of parameters or  admission 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.



2 Multiplexing and Demultiplexing


Packets sharing the same multicast group address, the same destination  port
number and the  same flow value  belong to the  same conference.   Within  a
conference, a packet is mapped to a site (state) through its synchronization
source identifier and network source port.  Unless otherwise defined, flow 0
is used for audio and flow 1 for video.


3 FMT


The content field within the FMT  option describes the media encoding  used.
The four  octets  contain one  of  the  encodings defined  by  the  Internet
Assigned Numbers  Authority (IANA)  or  an encoding  agreed upon  by  mutual
consent of all conference participants.   The names and the defined  codings
are defined in RFC TBD [media] and encoded in US-ASCII. Case is significant.
If the name is shorter than four  characters, it is padded with one or  more
space characters (ASCII 32  decimal).   Experimental encodings should  start
with the letter 'X'.


  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|  content  | clock quality |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | name of encoding                                              |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |   channels    | sampling rate (Hz)                            |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
...  encoding specific parameters                               ...
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


                          Figure 1:  FMT for Audio


For audio encodings, the  index into the table  of encodings is followed  by

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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|     FMT     |    length     |0|0|  content  | clock quality |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |                  name of video encoding                       |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |   version     | encoding-specific parameters                  |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
... encoding-specific parameters                                ...
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



                          Figure 2:  FMT for Video

a field containing  a channel  count and a  sample rate  field, measured  in
samples per second.(1)   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.

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



4 Standard Encodings


Unless specified  with  the FMT  option,  the  mapping between  the  content
field in an RTP packet and  encodings, sampling rates and channel counts  is
specified by Tables 1 and 2.  Values of 31 and below cannot be  redefined by
FMT options.  In other words, only  values of 32 and above are valid in  the
content field within an FMT option.  The receiver is expected to discard RTP
packets containing media data with unknown content field values.  Sites  are
expected to keep the mapping between content and encoding constant, so  that
lost packets containing FMT options do not lead the receiver to misinterpret
media data.    Additional standard  encodings are  defined by  the  Internet
Assigned Numbers Authority.
------------------------------
 1. 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  DVI                   8        1
                      6  L16                  16        1
                 _____7__L16__________________48________2__


                     Table 1:  Standard Audio Encodings














                               _number__name_
                                31      H261
                                30      Bolt
                                29      dvc
                                28      nv
                                27      CUSM
                                26      JPEG


                     Table 2:  Standard Video Encodings








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5 Port Assignments and Miscellaneous


ST-II SAPs and UDP  ports 4443 and  4444 are to be  used as the  destination
for multicast real-time audio and  video data carried by RTP,  respectively.
Unicast connections may use the this  or a set of mutually agreed-upon  port
numbers.

RTCP messages should be sent periodically, with a randomly varying period to
avoid synchronization.  A period of  between 3 and 10 seconds is  suggested,
increased to limit the  total network bandwidth and  host interrupt load  to
some small percentage of that attributable to media data.



6 Address of Author


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




























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