<gordon+asterisk@drogon.net> wrote:
>However, you'll need to do similar things to your asterisk box & router if
>it's behind NAT for IAX as you do for SIP. (You will need a static IP
>address on the NAT router and port-forward 4569 to the asterisk box, just
>as you'd port-forward 5060 and 10000-20000 for SIP)
Am I wrong to understand that IAX only needs one port, TCP4569 by
default? So I only need one port for each phone, while SIP requires at
least 3 (SIP, and one RTP each way)?
>And a SIP phone behind a NAT router is also solvable if it supports STUN.
But not all NAT routers support STUN, ie. keeping UDP ports open so
that incoming packets can make it.
>I know that SIP behind NAT isn't perfect, but with care, it's very usable
>and workable
But unless I'm mistaken, when NAT is involved, canreinvite must be set
to no, ie. all RTP packets must go through Asterisk instead of flowing
from one phone to the other?
Thanks guys.
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