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Running VoIP in a NAT environment

Hi,

We have three offices in different cities. Each office is connected with a T1 internet connection and the computers in the offices connect to the internet through a Linux server using NAT.

We want to install IP phone for each computer. So that the users in one office can talk to users in other offices as well as with them. We will be using IP phone sets from any company.

What I wanted to know, is it possible for users at one site, behind a NAT will be able to talk to users on other side of the NAT without VLAN?

Can we setup exchange or something for that using Linux server without using any hardware gateway or such?

And lastly, can our users get calls from any user on outside our networks who is using any net telephony service like net2phone or ICQ individually?

Thanks in advance.

Abhinay Sinha
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abhinaysinha
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abhinaysinha
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scampgbCommented:
Hi abhinaysinha,

You can do VoIP through routers that use NAT.  I've done this sucessfully before with no problems.

The "standard" for managing VoIP connections is called SIP - Session Initiation Protocol.  You can find out more from http://www.iptel.org

You can use either a SIP-compliant handset or a "softphone" on your PC.

X-Lite (http://www.xten.com/index.php?menu=products&smenu=xlite) is a very good SIP softphone.  

You will need to have a SIP server in order to manage the connections between the VoIP phones, or VoIP and the standard PSTN.
There are variety of organisations that will do this, some of them for free.  I personally use GossipTel (http://www.gossiptel.com)

Alternatively, you can set up your own SIP server on your network. A free one with lots of features is SIP Express Router (http://www.iptel.org/ser/)

As for free calls from outside users, this depends on who you're using for your SIP service.  Most of them will have peering arrangements with other SIP providers, so can route calls between them

I hope that this helps as a starter, let me know if you need any clarification.

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ripariusCommented:
You can also set up full PBX and conferencing with  Asterisk:   www.asterisk.org
It would use its IAX (Inter Asterisk eXchange) to work seamlessly between the Linux servers.  This also provides autoattendant, voicemail,...etc.  You would still need SIP softphones or SIP ethernet IP phones.

This would also allow a user in one city to grab an open telco line from another city and use it  if you add a POTS FXO card to each server.

Asterisk is an Open Source Linux  
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