[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
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