NAME¶

playSIP - SIP session recorder

SYNOPSIS¶

playISP [options...]

DESCRIPTION¶

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¶

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