[Live-devel] RTSPServer - RTSPClient latency

[Simon Schampijer]

Hello,

We are working on the distributed virtual concert.
We have developped a RTSPServer and a RTSPClient
using the live.com library.
The RTSPserver and RTSPCclient are both clients
of the jack audio connection kit.

jack : http://jackit.sourceforge.net/ 
       http://www.djcj.org/LAU/jack/


We got the following data flow :

The RTSPServer :

Audiosoftware(a readable jack client) -> write to the jack ringbuffer ->
we read in the dogetnextframe() method of JackSource derived from
FramedSource from the ringbuffer and put it into RTP packets
(payload type = 96)

The RTSPClient :
(inspired by the openrtsp example)

After the setup of our RTSPClient we create the media session from the 
sdp description we got from the RTSPServer, setup the subsession and
start playing the subsession. This time we write the audio data in the
jack ringbuffer in the continuePlaying() method of JackSink derived from
the MediaSink class. We read the audio data from the ringbuffer and
write it to our pcm device.

Problem :

Running the RTSPClient and RTSPServer on one machine adds latency    
of ~200ms. Our application needs realtime performance.  
Should we change the Recordering Threshhold time or the
socketInputBufferSize in the RTSPClient?

What is the buffering strategy used by live.com library - is there an
extra packet buffering in reception or emission and where could we find
them. 

Thanks in advance for any suggestion

Simon Schampijer