[OpenSIPS-Users] OpenSips as simple frontend to Asterisk to deal with NAT

[Bogdan-Andrei Iancu]

Hello,

Nathelper does not require rtpproxy for nat keepalive - the nat 
keepalive is at signaling level, while rtpproxy is exclusively for media 
pinning.

Nathelper is using the usrloc (registration based) for doing the nat 
pinging (to remember which destinations need to be pinged).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 19.02.2014 10:57, Rudy Eschauzier wrote:
> Hi Bogdan,
>
> Thanks for your response. I did investigate both nathelper and nat_traversal. My conclusion was that nathelper doesn't qualify per the requirements below, because it requires rtpproxy to do the nat keepalive. I probably missed something there.
>
> With nat_traversal being the better choice (or at least thinking that that is the case), I started looking for examples, but I couldn't find any (which I thought was a little weird). I _think_ what I would need is to use $source_uri to rewrite the contact header of the REGISTER request, before forwarding it to the Asterisk box. In Asterisk I would then need to define the OpenSips server as the outbound proxy. Does that make sense?
>
> Regards,
> Rudy.
>   
>
>   > What I would like is the following:
>   >
>   > -Run OpenSips on the router
>   > -Have OpenSips act as a stateless proxy, only
>   forwarding and mangling messages
>   > -Handle all registration requests on the Asterisk box
>   > -Make sure NATed clients are kept alive &
>   can be reached
>   > -Avoid OpenSips database use (not to overload the
>   router)
>   > -Not use a RTP proxy (the public RTP ports can be
>   forwarded to Asterisk, so clients can talk directly to the
>   PBX)
>
>