Provided by: livemedia-utils_2018.02.18-1_amd64 
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/
OPENRTSP December 2016 OPENRTSP(1)