openRTSP - open, stream, receive, and (optionally) record media
streams that are specified by a RTSP URL
playSIP - SIP session recorder
vobStreamer [options...]
playISP [options...]
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.)
- -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)