[Asterisk-Users] LiveVOIP troubleshooting

[Luki]

Hi everyone,

I need some ideas to troubleshoot this issue: I recently got an 800
numbers from LiveVOIP and it works but on most calls the caller gets
hears choppy audio (one drop out per 10 seconds or so).

I know this isn't LiveVOIP's support forum but I'm sure some here use
their 800 service and I'm interested in their feedback and ideas. And
don't get me wrong, LiveVOIP's support has been quite good --
cooperative, fast response, action taken as requested -- but I do not
want to try their patience. At this point I am not blaming them for
this issue either.

Here's the summary:

* I'm connected via IAX2 to 
* The server is in a datacenter with plenty of bandwidth. 
* Using ulaw with "standard" 20 ms frames. 
* I hear the caller perfectly fine, caller hears choppy audio.
* tcpdump shows incoming and outgoing packets right on time,
  every 20 ms in each direction. 
* I'm not using trunking for now. 
* Pings to LiveVOIP are about 35 ms. 
* iax2 show channels shows 1 ms jitter, 42 ms lag.
* Drop outs occur on IVR (or audio generated on the server itself) or
during normal conversation with a SIP client (ATA or phone) connected
to the server remotely. Connection between server and phones is well
tested and working fine.

I have asked LiveVOIP to switch me from their Vancouver node to their
New York node, which reduced ping times from 50 ms to 35 ms. Less
chops but still not perfect.

Note that the same server is already connected to several Broadvoice
accounts, which work flawlessly.

Anyway, if anyone has some ideas of what I can try, please let me
know. I do not want to keep trying all their nodes to find one that
works for me. I do not necessarily want to use a different codec
either since I have the bandwidth and I may be receiving faxes, so I
need ulaw.

Thanks and sorry for the long-ish post.
--Luki