Brekeke SIP can't recognize NAT environment.

1. Brekeke Product Name and Version:
Brekeke SIP Server Standard 3.7

UA1 in NAT register to brekeke SIP Server, but brekeke can't recognize UA1 as NAT. So SIP Server send rtp data to UA1's private ip.

UA2 in NAT(another network) register to brekeke SIP Server, brekeke recognize UA2 as NAT well.

Both UA1 and UA2 are NAT environment certainly.

I found the difference with register packet on [Dial Plan] - [History] page.

UA2 has "P-behind-NAT = true" in register packet. But UA1 has not "P-behind-NAT = true".

How brekeke sip server recognize UA as in NAT?
Why sip server can't recognize UA1 as in NAT?

Thank you in advance.

Are both UA behind same NAT?

It is an IP address in the valid private IP address range?
10.0.0.0-10.255.255.255, 172.16.0.0-172.31.255.255 or 192.168.0.0-192.168.255.255

What kind of SIP client is it? Does it support STUN or equivalent?

Thank you for reply.

No, there are not behind same NAT.
I found out why brekeke sip server can't recognize UA1 as NAT.
Becuase UA1's private ip is 172.168.0.8.

I know basic private ip ranges are
10.0.0.0-10.255.255.255
172.16.0.0-172.31.255.255
192.168.0.0-192.168.255.255

But there are a lot of pcs(sip client) on internet has another private ip address like UA1. All these sip client can't receive RTP data through brekeke sip server because brekeke can't recognize it as NAT.
How should brekeke sip server do to recognize all private network as NAT? (Configuration or anything.)
I set the configuration "Relay On".

I think there are couple of ways..
Set [Send UA's remote address] = "Yes" at [Configuration]->[RTP] page.


> But there are a lot of pcs(sip client) on internet has another private ip address like UA1.

But it is not recommended and it might cause another issue..
This is because 172.168.0.8 (for example) is assigned to someone officially.

If a router has sip ALG turned on, nat devices behind the router will appear as not nat because SIP ALG replaces the nat address with the wan address in the sip packets. This simulates the same problem you are having.

Since 99% of our customers are behind nat we added this to the sip server dial plan to force nat (which mainly serves to make sure the keep alive packets are sent every specified number of seconds.) No harm in sending the keep alive packets to the few that are not behind nat. Maybe this will solve your problem.

If the address conflicts with another actual address you may want to turn sip alg on in the router and do the below. Otherwise just do the below.

Matching:
$request=^REGISTER

Deploy
$register.contact.nat=true
$continue=true