[Asterisk-Users] SIP and firewalls?

[John Jensen]

Hi Hagen,
It's not exactly a pleasure to run SIP through firewalls but it can be
done.
At least in under some circumstances.

I have successfull run an Asterisk server from behind a NAT router
and run a SIP trunk to the SIP VoIP provider. The problems tend to 
arise when multiple SIP devices wants to communicate through the 
NAT router. My conclusion was that all my SIP devices should be
connected to the Asterisk box (ip-pbx) and that connected to the
provider.

To make this work I:

1. Set up port forwarding from the NAT router to the asterisk box
of the relevant ports:
UDP: 5004-5082
UDP: 10000-20000

2. Adjusted lan paramers in SIP.conf:
externip = 212.xxx.xxx.xxx       (I don't want your calls guys)
localnet=192.168.0.0/255.255.255.0

3. Set up SIP account for VoIP provider in sip.conf . For good measure

I put nat=yes in this account but I don't think it's required.

4. Make sure your SIP VoIP provider can handle NAT.

5. Extensions from outside the NAT router also works fine with this
setup.
Perhaps you need to set nat=yes in these extensions as well.


In this configuration things are not to terrible to get going.
Good luck.


Cheers,

John

>>> hagen at intellinc.co.za 02/15/06 11:42 am >>>

Hi 

We are currently using Asterisk 1.2.4 with IAX and app_meetme for
conferencing, but are looking to move to SIP because of issues with an
IAX
control we're using. 

The reason we moved from SIP to IAX in the first place was because of
the
poor NAT traversal with SIP. At that stage we were using Asterisk
1.0.*. How
does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the
older
Asterisk? Have there been improvements? Or is SIP (obviously depending
on
what client you use) still poor when it comes to NAT traversal and
firewalling? 

Many thanks

Hagen

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