our one way audio problem is there again
After I ruled out all codec mismatches and got our ISDN PABX (Auerswald Compact) talking perfectly with Asterisk on SIP level, we now encountered similar trouble with the Sipgate Trunk. Outbound calls always work perfectly, but inbound only to 50% or less. In these cases, the calling party cannot be heard, while everything spoken into a local SIP phone (Snom 320s and a few Gigasets) is perfectly reproduced at the callers side.
I would accept NAT problems as the only reason if it were a permanent problem, but it is as intermittent as it can be.
After some tweaking I called in 8-9 times without problems but then it reappeared. It does not matter how long I wait between calls (assuming there’s still an UDP port open on the router) or whether another peer is in use at this moment.
Our router is a Draytek Vigor 2820, it makes no difference whether I forward 49152-49408 (the ports defined in rtp.conf) to the * or not. We’re going to swap the router for a Cisco on wednesday if no other solution comes to help.