Hello,
Kamailio can help with NAT traversal -- there are couple of options, one
of the most common deployments is using nathelper and rtpengine (or
rtpproxy).
A similar solution is provided in the default kamailio.cfg, just look at
WITH_NAT ifdefs to figure out the related parts:
-
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
Cheers,
Daniel
On 07.05.18 12:22, Pinter, Gerd. wrote:
Hi there,
unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway,
I have to maintain a SIP Server for high quality Audio transmission with special
UA's.
Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I
Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP
address issues" most providers use private ranges separated by CGNAT from the real
Internet. This causes a lot of trouble. In some cases stun server entries on the clients
helps, in other cases don't!
But these days I read in announcement of a german company (Mayah Communications), which
is in the ProAudio biz since decades, that they offer a SIP Service where no stun is
recommended whether the clients are behind NAT or not.
After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that
there might exist a solution for my Kamailio SIP server to make it work like the
"Mayah" Sip Server.
What settings have to be done that Kamailio uses IP Header Information for SIP signalling
instead of the content of SIP packages themselves. Without stun you find the Local Address
in the SIP packages, and with Stun the external Address but mostly with the wrong
portnumber, but in the IP header you'll find the correct IP:port.
I already read a lot of the Kamailio documentation but due to the fact that I am not too
deep in IT I am not sure if I got it all. If someone could give me a useful hint on that,
that would make my day.
Hope this description is unterstandable.
Nice Regards
Gerhard Pinter
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