--- name: "openRTSP" section: "1" platform: "linux" package: "livemedia-utils" version: "2018.11.26-1.1" release: "Debian 10 (Buster)" platforms: ["linux"] url: "https://new.carta.tech/openRTSP" --- # openRTSP(1) open, stream, receive, and (optionally) record media streams that are specified by a RTSP URL playSIP - SIP session recorder ## 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 \" option is also given) - **-a** play only the audio stream (to 'stdout', unless the "-P \" option is also given) - **-A** \ specify the static RTP payload format number of the audio codec to request from the server ("playSIP" only) - **-b** \ change the output file buffer size - **-B** \ change the input network socket buffer size - **-c** play continuously - **-CFP** **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 \** **specify an explicit duration** - **-D \** **specify a maximum period of inactivity to wait before exiting** - **-E \** **request that the server end streaming at the specified absolute time (format:** **"YYYYMMDDTHHMMSSZ" or "YYYYMMDDTHHMMSS.\Z") (used only with** **-U\)** - **-f \** **specify the video frame rate (used only with "-q", "-4", or "-i")** - **-F \** **specify a prefix for each output file name** - **-g \** **specify a user agent name to use in outgoing requests** - **-h \** **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 \"** **option is also given)** - **-I \** **specify a particular network interface on which to receive data** - **-k \ \** **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 \** **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 \** **specify the client port number(s)** - **-P \** **write new output files every \ seconds** - **-q** **output a QuickTime '.mov'-format file (to 'stdout', unless the "-P** **\" 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 \[\\]** **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 \** **request that the server seek to the specified time (in seconds) before streaming** - **-S \** **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 \** **like "-t", except using RTSP-over-HTTP tunneling. ("openRTSP" only)** - **-u \ \** **specify a user name and password for digest authentication** - **-U \** **request that the server seek to the specified absolute time (format:** **"YYYYMMDDTHHMMSSZ" or "YYYYMMDDTHHMMSS.\Z") before streaming** - **-v** **play only the video stream (to 'stdout', unless the "-P \"** **option is also given)** - **-V** **print less verbose diagnostic output** - **-w \** **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 \** **request that the server scale the stream (fast-forward, slow, or reverse play)** ## SEE ALSO *[openRTSP(1)](/openRTSP/1.md)***,** *[playSIP(1)](/playSIP/1.md)* *http://www.live555.com/openRTSP/***,** *http://www.live555.com/playSIP/*