[Live-devel] RTSPServer - RTSPClient latency

[Ross Finlayson]

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

No - that parameter has an effect only when arriving network packets are 
out of order.  Normally, that doesn't happen.

>  or the
>socketInputBufferSize in the RTSPClient?

No - that parameter affects the size of the OS's input socket buffer, but 
shouldn't affect its latency (if it's empty).

>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.

No, there's no 'extra' buffering anywhere.

One thing you should ensure, however, is that you're setting 
"fDurationInMicroseconds" correctly in your "JackSource::doGetNextFrame()" 
implementation.  That variable tells the RTP transmitting code how long to 
delay - after sending a packet - before asking for more data.  If you are 
reading from a 'live' input source - and you can deal with 
"JackSource::doGetNextFrame()" being called when there's no new data 
currently available - then you could set that variable to 0 (or not set it 
at all, which has the same effect).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>