[Freeswitch-users] NAT between FS and remote SIP phone
codecomplete at free.fr
Tue Apr 21 04:25:26 PDT 2009
> In my experience your asking for nothing but headaches to have both
> devices behind NAT. If at all possible put the FS server on a public IP
> and you'll get rid of 99% of your problems.
I know, but the Freeswitch-based product I'm working on is meant for SOHO
users who only have a basic ADSL modem that acts as a NAT router, so I
pretty much have to put everything in the private LAN. Some users might need
to have remote SIP phones on the Net, behind their own NAT routers, so I
need to know how to set things up so that SIP/RTP work OK even when both
endpoints are NATted.
However, I can map ports on the Freeswitch side so that the SIP and RTP
ports are always open for remote SIP phones to connect to the server and
VoIP gateway. It's just that remote SIP phones are off-limit (remote
locations I don't have access to).
> I THINK this depends on how you set it up. From what I understand you
> can have FS proxy the media (and handle codec negotiations, etc) if you
> so choose. If both endpoints (linksys gateway and phone) support the same
> codec you should be able to set FS to drop out of the media path.
If someone has already successfully had a Linksys gateway talk to a remote
SIP phone, both behind a NAT router, I'm interested :-)
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