[Freeswitch-users] Working configuration for FS behind NAT?

[GillesToo]

Hello

I'd like to set things up, where...
- FS and local SIP extensions are on a private LAN behind a NAT firewall,
- remote employees are allowed to register with Freeswitch so they're part
of Freeswitch,
- none of the routers support UPnP or NAT-PMP, and
- to lower load on the Freeswitch server, I'd like to have RTP packets flow
directly between extensions instead of going through Freeswitch, like
Asterisk does with its "canreinvite=no" option. However, Freeswitch should
remain in the loop somehow so that users can access PBX features during
their call.

Here's the picture:

http://img577.imageshack.us/img577/1795/freeswitchnat.png

Apparently, changes to Freeswitch made 
http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios this page 
obsolete. If Freeswitch can do this, does someone have a basic, working
configuration that I could use to get started?

Thank you.

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