[Asterisk-Users] SIP phone behind NAT

[Andrew Radke]

Hi Olaf,

I've just started working on a SIP and RTP proxy to handle exactly this. 
I'm really just in proof of concept at the moment but just one hour ago 
I got a completely successful connection out over NAT in which both 
endpoints thought they were talking to the proxy. I should have the code 
posted in the next few days. So far it's only tested under Linux but it 
should work on Windows without too many problems. I'll post more info in 
the next few days but feel free to email me directly if you are 
inerested or haven't heard anything from me.

Regards,

Andrew Radke

Olaf Menzel wrote:

> Hi all,
> --------
> 
> I have a Asterisk at a public Network (official IP address). In the local 
> network I have isntalled a Snom 200 IP phone and in my home network (behind 
> NAT) a Snom 100 device.  I can dial the Snom200 device from my home location 
> without any problems but the Snom200 can not dial me. It always gets a "we do 
> not rely". I tried to forward the SIP Port (5060) UDP via UPnP to the 
> internal Snom100 IPadress and a port range forwarding  of 16384 -  32768 
> (UDP) for the RTP traffic. Additionally I tried to change host = dynamic to  
> host = myserver.dyndns.org to ensure the SIP traffic is going to my Linksys 
> ADSL router and be forwarded to the internal SIP 100 phone. But all my effort 
> did not have success. Any suggestions ??
> 
> regards
> 
> Olaf