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OPENRTSP

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/

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