[asterisk-users] Do I need a sip proxy?

Tue Jan 11 06:38:19 CST 2011

Hello Bruce,

Your understanding of NAT is correct, and your setup should work.

I’m not familiar with Pfsense, but I suspected that your problem was due to a SIP ALG. Pfsense seems to have a SIP ALG and other special handling of VoIP traffic. Hence, you are not using plain NAT. Pfsense is probably rewriting the SIP packets in addition to the IP packets. Try reconfiguring Pfsense or swapping it for something else. A good way to troubleshoot your scenario is to compare the traffic in your end to the traffic on your providers end (or on either side of pfsense). Pay attention to the source and destination IP and ports in addition to the contents of the SIP messages.

http://doc.pfsense.org/index.php/VoIP_Configuration
http://en.wikipedia.org/wiki/Application-level_gateway

With kind regards,
Pan

From: Bruce B 
Sent: Tuesday, January 11, 2011 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: [asterisk-users] Do I need a sip proxy?

Hi Everyone, 

I am running multiple instances of Asterisk in Proxmox and so far I had one central Asterisk feeding all others with trunks from one provider. Now, I want to connect each Asterisk server directly to the provider. Based on my understanding, each connection made to the provider port 5060 would be on a port that is unique to that server. And so other connections made to the same provider will go out through a different port and should receive responses through that different port. At least that is my understanding of NAT. The provider should see me trying to register from the same IP with multiple different ports (high number ports; not talking about 5060 as this is outbound and not inbound) and should be able to differentiate between SIP packets coming from various servers. However, it seems to not happen.

There is some sort of clash and only one of the servers shows registered with the provider and other's trunks go down. I have noticed that keeping one server works. It could also be that my Fail2ban kicks in on all servers if the SIP packets received are broadcasted to all servers which shouldn't really happen and router should take of this by sending it to the server that has the established connection through that port.

My equipment:
Asterisk 1.6x
Pfsense 1.2.3
Dumb Switch

My questions:
A- What is the rational behind this?
B- Do I need a sip proxy server? Something like Siproxd, OpenSIPs, or Kamailio?
C- Which one of the above is the easiest to get running given I never tried any of those.
D- If I am doing an SIP proxy server then it might have to also be redundant. What options do I have in that and which of above or any other suggested package might be great for future expansions.

Clarification on how NAT would work in situations like this would be much appreciated.

Thanks

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