Finite streaming questions

[Dave Higton]

My company is developing a specialised RTSP/RTP streaming server. I'm using VLCMP to test it during development.

Currently I'm streaming a 10 second long "file" via RTSP/RTP. Our response to the PLAY request includes "Range: npt=0.00-10.000". Three things surprise me:

1) VLCMP buffers all 10 seconds before it starts to play any audio. Is this normal? Can I persuade it to buffer less?

2) After the audio has finished, VLCMP issues many warnings that tell me it was expecting more audio. Finally, something like 10 seconds after the end of the replay, it issues a TEARDOWN request. Is it possible to tell VLCMP that the 10 seconds is all there is?

3) Bearing in mind that it is a finite file, shouldn't the track slider move?

[Rémi Denis-Courmont]

The RTSP client stack is provided by the live555 library, not directly by VLC. So if there is any interoperability problem, I would advise you contact the author of live555 directly.

I can only hazard that live555 expects the duration of the stream in the SDP payload from DESCRIBE (t=...). Normally, VLC only buffers about one second, there may be some lip sync problem with RTCP or something. Again, this is pure speculation; it's up to live555 whose internals are not known to VLC developers.