One way audio problems

Hi,

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.

Regards,
Christian

I’d start with a traceroute between the end points to look for excessive latency or jitter

This isn’t one way audio, as normally understood.

Are you actually dropping frames, or is Wireshark just throwing them away.

If the latter, has there been a timestamp discontinuity, with no correponding change in SSRC. Asterisk does that but some Cisco phones don’t like it.

Hi,

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)

The network is badly overloaded.