Provided by: livemedia-utils_2018.11.26-1.1_amd64 bug

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/