Hi I seem to be getting about 250-500ms drop outs on receive audio for some calls that are routed over the internet. Probably no news there. My end point is a polycom 500 and was wondering if anyone could recommended jitter buffer settings. codec is alaw min buffer is at 40ms maximum buffer is at 200 ms shrink rate is at 3000ms I think i am getting less than 40ms jitter most of the time - would the approx 250ms audio drop refer to the amount of time it takes to start buffering? Maybe I should reduce the minimum buffer so it is always on? Any tips from the field appreciated. Thanks J __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com