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Unread 02-04-2007, 12:26 PM   #16
pianoquintet
 
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It seems to work differently in my case. I did a tcpdump during both inbound (PSTN to SIP) and outbound (SIP to PSTN) calls and the RTP flow seemed to go through "hosted-by.leaseweb.com" (which is the European Voxalot server) instead of my Italian SIP provider (Messagenet.it) servers.

These were my settings at the time of the experiment:

- my internet provider is Fastweb.it, which puts all its users behind a NAT;
- my SIP client is registered with Voxalot only; STUN is set to stun.voxalot.com.au;
- Messagenet.it, my Italian SIP provider is entered in Voxalot's "Providers" list and both "Active" and "Register" are set to "Yes";
- my Voxalot dial plans point to Messagenet for the relevant outgoing calls;
- I dialled PSTN numbers from my SIP client to test outbound calls and dialled the PSTN number associated with my Messagenet account from a landline and answered them with my SIP client to test inbound calls.

In both cases UDP traffic seemed to go through Voxalot. I would like to have my voice traffic go through the relevant SIP proider instead, both in order to avoid burdening Voxalot's servers and to reduce latency. Can you help?

Thank you,

Carlo
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