Hello,
we currently experience "incoming voice loss" after an outgoing
call has been established. i.e. after the remote party picks up the
call you dont hear anything for 1-2 seconds which results in both
sides of the connection saying repeatedly "Hello" until both sides
can hear each other.
This phenomenon occurs with nated and non natted clients so it
doesnt seem to be related to "rtpproxy" or other NAT problems.
It occurs between SIP<>SIP and SIP<>PSTN Calls (via a cisco gw)
independantly of the UA type used.
Has anyone an idea where to look for the bug/where to start debugging?
best regards,
Arnd