[Speex-dev] Prebuffering best practices

[David Barrett]

What is the best way to pick a prebuffering length for a streaming audio 
application using UDP transport?

I'm using Speex in a VoIP application with RTP transport, currently with 
a fixed 500ms prebuffer on the playback side.  However, I'd like 
something a bit more adaptive to accomodate high-jitter connections.

For example, in one test configuration there is a very low average 
round-trip latency (50ms), but it spikes all over the place (sometimes 
10ms, sometimes 500ms).  Thus I can't make my prebuffer duration 
proportional to latency, but somehow proportional to "jitter".  But I'm 
not sure the best way to quantify this, nor how to tranform that into a 
reasonable prebuffer length.

Thus I'm curious what experience you've had in this area, and what you 
can recommend as a good way to adaptively compute a prebuffer duration. 
  Thanks!

-david