We have the following problem: in random calls from a phone to PSTN and viceversa, the audio received in the PSTN side becomes badly and noise making it impossible to understand.
We’ve done some testing, measuring about 30% of Drop by Jitter buffer using Wireshark, but we don´t know why it happens, anyone can help us?
We have the following topology:
<- SIP -> <- IAX -> <gateway (1TE820BF)> <- E1 ->
The Proxy and Gateway are Dell R310 (Xeon X3450 2.66GHz 4 cores, 4G RAM) with Asterisk 11 connected to a switch PowerConnect 5548 with 10 Mbps to internet and 1Gpbs between both, using 2 NIC cards (bonding).
The phone is a Cisco SPA3102 Phone Adapter with 10 Mbps to internet.
Proxy/Gateway and phone are connected to different ISPs in Chile.
During testing, we have 1 or 2 calls simultaneously.