olivier@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