Hi,

I have recently bought a IP/PABX system with one FXO and one FXS port. I intend to install this on a remote site with a public but dynamic IP (can be resolved via dyndns though) and make calls via clients that are NATTed (inside a home router). I would like to seek advice on the port opening and the recommended settings. I have been reading a lot on VOIP and I am getting feedback that SIP calls are difficult to establish on a NATTed environment. Hope you can help me on the following questions:

1.) SIP port 5060 UDP?
2.) RTP ports - what range should I open for this? I see some use 10000-20000 UDP
3.) STUN server - Is this something that needs to be configured?

How can I ensure that the other party can hear the audio just like a regular telephone? Is it really impossible to do if the client is behind a router in which it is using a Private IP Address?

What other network configurations needs to be done?

Thanks and Regards,
Don

Do you want that your customers use VOIP software and connect to IP-PABX?

Yes. If possible I want flexibility of clients to connect to IP PABX via IP phone whether or not they are using public IP or behind a NAT router. Thanks.

But can your IP-PABX handle connections like this. It has something like SIP server inside. I would suggest to do a test: take switch get one SIP phone connect phone and your IP-PABX to switch and try to call.

I think there is a manual for this IP-PABX where said what ports should be used, or what kind of SIP software it has.

HI nimnull22,

The IP/PABX is based on Asterisk. The IP/PABX device is actually IP04 which has one FXS and one FXO port. I have not receive the device yet but will try. I am just worried on the NAT issue if there is.

Yes, there will be some difficulties, but there is many solutions.
Good page, may be you've seen it: http://www.voip-info.org/wiki/view/NAT+and+VOIP

I've used linphone with STUN under at least 2 NAT - it works.

Can IP changes during a phone call?

The IP will only be changed once you reboot the modem and the router. After the reboot, my router will update dyndns.org with the new IP address. I will set the IP of the asterisk to resolve to my host in dyndns.org so it will know the IP address. What concerns me is the client that will connect to this server. Can they do a NAT traversal so that the asterisk will know where to forward the port?

Good page, may be you've seen it: http://www.voip-info.org/wiki/view/NAT+and+VOIP

I've used linphone with STUN through at least 2 NAT - it works.

Hi nimnull22,

One more question. Is the STUN Server something I should setup?

No doubt, and for the testing purpose on THE SAME computer where Asterisk will be.

If you need help with testing, I can call. Have lots of free time.