[Asterisk-Users] SIP / GrandStream Configuration

[Stephen Varga]

On Thu, 2003-09-25 at 15:41, Michael Koehler wrote:
> It is not a feature of the router, it is the way SIP is handled with
> nikotel.com
> 
> I recently wrote that i'm using just a plain router with my natted
> asterisk because "Stephen Varga" wrote that SIP behind
> NAT (in relation to asterisk) is impossible. It is possible because
> i'm using asterisk this way.
> 
> There is also nothing special to setup with the router for nikotel and
> NAT, except you have a firewall and need
> straight rules, then you may use port forwarding.

Ok maybe I was being to broad in my original statement, so let me
clarify.

There orginal question was does the scenario

 SIP Phone --- NAT --- Internet --- NAT --- Asterisk 

work.

In general this can not be easily accomplished, because of the real ip
address of the devices get embedded in SDP message during the INVITE
process. Most phones can be changed to use the NAT address in this
process, so this solves one side of the conversation. However I have not
found away to do this in the asterisk software, thus SDP message needs
modified to change the ip address to the NATed one outside of * for this
to work. For this I have not discovered a reasonable solution.

In Mike's case, I am guessing the SDP message is being modified when the
packet arrives at the Nikotel's gateway. Which makes this a specialized
case.

So that still leaves us with a general problem of SIP and NATing on both
sides, for the rest of us not having the benefit of the software that
nikotel is using to make this scenario work.