[Asterisk-Users] VoIP bad voice quality

[Ashish Shinde]

Hi,
     We have Asterisk, running on a  P4 box running Suse 9.1, making
calls using IAX through SimpleTelecom and Nufone. What we are looking
for is toll quality voice.
     The problem is that voice over calls routed through SimpleTelecom
and nNufone occassionally breaks. We also have a digium card and the
calls over the digium card using the Zaptel Interface have a very good
quality.
     We have enough bandwidth, the latency to the servers is 100-150ms
and the packet loss is around 1%. We tried using the G711 and G729
codec and also have the jitterbuffer enabled.
   How can I solve this problem of voice quality? Can a better
implementation of jitterbuffer with packet loss concealment  help? If
so how do I get the newer implementation. I would really like to help
out in the development of the new jitterbuffer if it has not yet been
implemented.
   Would be grateful if someone can help me out in this regard.

Thanks and regards,
 - Ashish