A) The Sipura Call Status on the Info page. Keep refreshing the page during the course of a call. It tells you the codecs used (i.e. whether or not your "preferred" codec is being used), the bit rate, packets lost, packets in error, jitter, and other info.

B) Check the round-trip delay. You can use something like ping plotter to determine if there is any packet drops or unusual congestion. There are a few Round Trip Delay fields, but they will not get updated. On the SIP tab, set the RTCP delay to 20 (or more) on both ends of the call. This will cause the Round Trip Delay to get updated every 20ms or so. 300 to 400ms Round Trip Delay is sufficient.

C) Adjust the codecs used, particularly if you have limited upstream bandwidth. G.711 uses the most bandwidth. G.723 should be used when bandwidth is severely constrained. G.726 and G.729a use slightly more bandwidth than G.723, but less than G.711.