[Freeswitch-users] NAT between FS and remote SIP phone

[Fred-145]

Hello

I'd like to know how to set things up when using the following scenario:
- a VoIP gateway on the same LAN as the Freeswitch to handle incoming calls
from a POTS line
- a remote SIP phone somewhere on the Net
- the FS server and the remote SIP phone are both behind a NAT router
- the remote SIP user either doesn't have the computer skills to map ports
on his NAT router, or doesn't have access to it (eg. staying in a hotel or
connecting to FS from a wifi connection @ Starbucks)

Here's the layout:

tp://img14.imageshack.us/img14/2574/freeswitchlinksysnat.

The questions I have:
1. What ports need to be mapped on each router?
2. If I understood it correctly, UPnP is a technology that can open ports
dynamically. Are there ways to tell if a router supports UPnP, are there
other ways to have a remote SIP phone work right out of the box, or are
there cases where mapping ports manually is the only way to get SIP/RTP to
work?
3. When a call comes from the POTS line and meant for the remote SIP
extension... do RTP packets flow directly from the Linksys VoIP gateway to
the remote SIP phone, or do they go through the FS server?

Thank you for any hint.
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