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.
I made a traceroute from a laptop in the same network as the phone to de sip proxy:
traceroute to X.X.X.X (X.X.X.X), 64 hops max, 52 byte packets
1 192.168.0.1 (192.168.0.1) 2.172 ms 1.810 ms 1.787 ms (0% loss)
2 * * * (100% loss)
3 192.168.224.234 (192.168.224.234) 31.195 ms 31.251 ms 12.599 ms (0% loss)
4 192.168.22.45 (192.168.22.45) 12.798 ms 13.900 ms 34.724 ms (0% loss)
5 * * * (100% loss)
6 gw-gtd.pit.ip.telmexchile.cl (201.238.238.73) 11.829 ms 14.006 ms 14.057 ms (0% loss)
7 str12.ge1-46.cn2.gtdinternet.com (201.238.238.6) 14.904 ms 13.770 ms 13.106 ms (0% loss)
8 190.196.126.8 (190.196.126.8) 12.229 ms 13.194 ms 13.002 ms (0% loss)
9 X.X.X.X (X.X.X.X) 15.831 ms 14.749 ms 12.954 ms (0% loss)