NAME

       openRTSP - open, stream, receive, and (optionally) record media streams that are specified by a RTSP URL

       playSIP - SIP session recorder

SYNOPSIS

       vobStreamer [options...]

       playISP [options...]

DESCRIPTION

       The  program  will  open  the  given  URL  (using  RTSP's "DESCRIBE" command), retrieve the session's SDP
       description, and then, for each audio/video subsession whose RTP payload format it  understands,  "SETUP"
       and "PLAY" the subsession.

       The received data for each subsession is written into a separate output file, named according to its MIME
       type. For example, if the session contains a MPEG-1 or 2 audio subsession (RTP payload type 14)  -  e.g.,
       MP3  -  and  a  MPEG-1  or  2 video subsession (RTP payload type 32), then each subsession's data will be
       extracted from the incoming RTP packets and  written  to  files  named  "audio-MPA-1"  and  "video-MPV-2"
       (respectively).  (You  will  probably  then  need  to  rename these files - by giving them an appropriate
       filename extension (e.g., ".mp3" and ".mpg") - in order to be able to play them using common media player
       tools.)

OPTIONS

       -4     output  a  '.mp4'-format  file  (to 'stdout', unless the "-P <interval-in-seconds>" option is also
              given)

       -a     play only the audio stream (to 'stdout', unless the  "-P  <interval-in-seconds>"  option  is  also
              given)

       -A <codec-number>
              specify  the  static  RTP  payload  format  number  of  the audio codec to request from the server
              ("playSIP" only)

       -b <buffer-size>
              change the output file buffer size

       -B <buffer-size>
              change the input network socket buffer size

       -c     play continuously

       -C     Explicitly ask for a multicast stream even if the server's "DESCRIBE" response doesn't  specift  a
              multicast address. (Note that not all servers will support this.) ("openRTSP" only)

       -d <duration>
              specify an explicit duration

       -D <maximum-inter-packet-gap>
              specify a maximum period of inactivity to wait before exiting

       -E <absolute-seek-end-time>
              request  that  the server end streaming at the specified absolute time (format: "YYYYMMDDTHHMMSSZ"
              or "YYYYMMDDTHHMMSS.<frac>Z") (used only with -U<initial-absolute-seek-time>)

       -f <frame-rate>
              specify the video frame rate (used only with "-q", "-4", or "-i")

       -F <fileName-prefix>
              specify a prefix for each output file name

       -g <user-agent-name>
              specify a user agent name to use in outgoing requests

       -h <height>
              specify the video image height (used only with "-q", "-4", or "-i")

       -H     output a QuickTime 'hint track' for each audio/video track (used only with "-q" or "-4")

       -i     output a '.avi'-format file (to 'stdout', unless the "-P  <interval-in-seconds>"  option  is  also
              given)

       -I <interface-name-or-address>
              specify a particular network interface on which to receive data

       -k <username> <password>
              specify  a  user  name and password that's required to authenticate an incoming "REGISTER" command
              (used with "-R" only)

       -K     Periodically send a RTSP "OPTIONS" command, to keep the connection alive.  (This  is  useful  with
              buggy servers that don't listen to our periodic RTCP "RR" packets instead.)

       -l     try to compensate for packet losses (used only with "-q", "-4", or "-i")

       -m     output each incoming frame into a separate file

       -M <MIME-subtype>
              specify  the  MIME subtype of a dynamic RTP payload format for the audio codec to request from the
              server ("playSIP" only)

       -n     be notified when RTP data packets start arriving

       -o     request the server's command options, without sending "DESCRIBE" ("openRTSP" only)

       -O     don't request the server's command options; just send "DESCRIBE" ("openRTSP" only)

       -p <starting-port-number>
              specify the client port number(s)

       -P <interval-in-seconds>
              write new output files every <interval-in-seconds> seconds

       -q     output a QuickTime '.mov'-format file (to 'stdout', unless the "-P  <interval-in-seconds>"  option
              is also given)

       -Q     output 'QOS' statistics about the data stream (when the program exits)

       -r     play the RTP streams, but don't receive them ourself

       -R [<port-number>]
              Waits for an incoming "REGISTER" command, specifying a "rtsp://" URL to play.  This option is used
              instead of a "rtsp://" URL on the command line. ("openRTSP" only)

       -s <initial-seek-time>
              request that the server seek to the specified time (in seconds) before streaming

       -S <byte-offset>
              assume a simple RTP payload format (skipping over a special header of the specified size)

       -t     stream RTP/RTCP data over TCP, rather than (the usual) UDP. ("openRTSP" only)

       -T <http-port-number>
              like "-t", except using RTSP-over-HTTP tunneling. ("openRTSP" only)

       -u <username> <password>
              specify a user name and password for digest authentication

       -U <initial-absolute-seek-time>
              request that the server seek  to  the  specified  absolute  time  (format:  "YYYYMMDDTHHMMSSZ"  or
              "YYYYMMDDTHHMMSS.<frac>Z") before streaming

       -v     play  only  the  video  stream  (to 'stdout', unless the "-P <interval-in-seconds>" option is also
              given)

       -V     print less verbose diagnostic output

       -w <width>
              specify the video image width (used only with "-q", "-4", or "-i")

       -y     try to synchronize the audio and video tracks (used only with "-q" or "-4")

       -z <scale>
              request that the server scale the stream (fast-forward, slow, or reverse play)

SEE ALSO

       openRTSP(1), playSIP(1)

       http://www.live555.com/openRTSP/, http://www.live555.com/playSIP/