[Serusers] Help needed : SER / PSTN / NAT

Klaus Darilion klaus.mailinglists at pernau.at
Wed Mar 17 15:11:07 CET 2004



olivier at siteboulevard.com wrote:
> Hi,
> 
> After some testing on the latest release, i have some problem doing the 
> following on LINUX :
> 
latest? du you mean unstable or latest stable?

> Scenario :
> - SIP Phones behind a NAT
> - SER server under linux with rtpproxy launched
> - a 3660 cisco gateway with PSTN connectivity enabled.
> 
> When i call with SIP phone a PSTN number, everything is OK BUT no sound 
> anywhere.
> 

Use ethereal to verfiy that the SDP in the INVITE and 200 OK (or 183 
Early Media) are rewritten by nathelper&rtpproxy to point to the IP:port 
of the rtpproxy. If this is correct, you should see RTP streams to 
rtpproxy (which should be forwarded to the GW and the NAT box)

> I could not find a sample ser.cfg script that reflect this scenario. Could 
> someone send me this scenario ?

this is like any other scenario with a client behind NAT and one client 
with public IP.

> 
> Maybe i missunderstood some things. In particular, do i need to launch two 
> instances of ser (one for outbound proxy, another for request. If yes, how to 
> do that)

You don't need two instances.

Klaus




More information about the sr-users mailing list