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SIP/VOIP Problem

Hello Mates can someone please help with this?

I cannot get SIP to work behind Manual NAT Hide or Static makes no difference, When caling from two software agents that are in the same network behind the firewall, and the SIP server is in external network, not over the Internet but on a line directly from the ISP. The best I can get is one way audio. When calling to a mobile phone it all works great audio in two directions as it should.

Its shown both in the SIP server and tcpdump in the firewall that many of the UDP packets that belong to the media stream is not NATed. They are sent to the ISP SIP server untranslated and hence are dropped on the way back since there are no route for our internal network from the ISP. Therefore we have to use NAT.

We use version R80.10 in a VSX Cluser.

3 Replies
Admin
Admin

Re: SIP/VOIP Problem

This thread might help:  

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Re: SIP/VOIP Problem

We finaly solved this, changed the hide NAT IP that we used to an IP defined on the correct side of the firewall.

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Re: SIP/VOIP Problem

Now then another problem has arised. 

Our ISP require us to use Static NAT otherwise they wont support our solution.

When I NAT to the IP Addresses that we have configured on the linknet to the ISP the SIP Softphones works just fine. But when I NAT to a "mapped" IP Addresses space the RTP traffic isnt goning as it should (same problem as before) 

Is Checkpoint limited to use the real configured IP Addresses only for RTP to work?

 

 

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