[asterisk-users] SIP reinvite _and_ NAT

[Roger Schreiter]

Hi,

I have a sipphone behind a router doing NAT, an asterisk
box in the middle and another asterisk box, which works
as gateway to further destinations.

The asterisk box in the middle should do all call setup
and tear down, but no RTP. RTP should flow directly between
the sipphone via the router to the other asterisk box.

When calling _from_ the sipphone, everthing is fine:
The asterisk box in the middle is reinviting, and the
other asterisk box is finally exchanging RTP with the sipphone
in both directions.

When calling _to_ the sipphone, there is a problem:
The asterisk box in the middle again is reinviting, and
the RTP stream from the sipphone than goes directly to
the other asterisk box.

But the RTP stream from the other asterisk box is sent to
the private IP address of the sipphone (192.168....)
The NAT workaround is not effective in this case.

Any hints?

Just for understanding: Who is responsible in this case
for the NAT workaround: The asterisk box in the middle
who is reinviting or the originating asterisk box who is
then sending the RTP traffic?


Roger.