[asterisk-users] Jitter only affecting meetme - and echo testing

[JT]

Greetings List!

I'm currently rolling out a new deployment of Asterisk 1.8 to replace
existing 1.2 servers...and have run into an issue which could use your
assistance!

For testing I have trunked (iax2) two of the servers - one running 1.8 and
the other at 1.2.  Calls placed from SIP --> SIP sound fantastic and crystal
clear.  However, when I place a echo test call (*43) from 1.8 to 1.2 the
result is a pulsing echo and garbled audio.  The same result is found when
dialing into a meetme conference being held on 1.2 from the 1.8 server.

I know that the jitter buffer has gone through quite a bit of work since 1.2
but my results at this point seem to indicate 1.8 is actually worse than
1.2.
I've also trunked together two 1.8 boxes (physical locations are in
different countries), to rule out if 1.2 was somehow causing the extra
jitter.  My results were identical - echo testing/meetme from 1.8 to 1.8
resulted in the same audio issues.

The new servers are in the same locations as our production units, using the
same networks/etc... so I expected identical results between them, not for
1.8 to be substantially worse.

Any recommendations?  Has anyone else experienced a similar issue and if so,
perhaps you could share your experience.

Thanks!
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