NAT sipserver doesn't work with phones which are behind NAT

sesci · Post by **sesci** » Mon Oct 31, 2016 4:48 am

1. Brekeke Product Name and Version:
Brekek sip server 3.6.1.2/447
2. Java version:

3. OS type and the version:
windows 10
4. UA (phone), gateway or other hardware/software involved:
ICE implemented sip phones which uses pjsip
5. Your problem:

I have setup a network like below:
https://imageshack.com/i/pljLsBhkj

and i have setup sip server configs like below for NAT and RTP relay off:
https://imageshack.com/i/plEGzOgKj
https://imageshack.com/i/pmOlCCE3j
https://imageshack.com/i/pm9Fobdtj
https://imageshack.com/i/poEk1vKLj

when i tried to make a call from phone-1 to phone-2 ( i have pushed my phones to send RTP stream over TURN server) , call couldn't established. phone-1 invite message sdp part connection information is 10.1.1.3:64545, but sip server invite message -which sended to phone-2 based on phone-1 invite message- sdp part connection info is 20.1.1.1:12536. sip server still use source port instead of sdp.
what should i do for sip server don't change sdp part connection information and send invite message with phone-1 invite message sdp part?

ps: when i move sip server to 10.1.1.x network, everything is working fine and RTP streams can goes over TURN server.

ambrosio · Post by **ambrosio** » Mon Oct 31, 2016 10:18 am

There are several ways..
Try [RTP relay even with ICE] = "yes" in [Configuration]->[RTP] page.

If it doesn't work, disable RTP-relay with the DialPlan rule like the following.
-------------------------
[Matching Patterns]
$request = ^INVITE

[Deploy Patterns]
$rtp = false
&net.nat.force = false
$continue = true
-------------------------

sesci · Post by **sesci** » Tue Nov 01, 2016 1:11 am

thank you ambrosio.
your dial plan rule solved my problem.

ambrosio · Post by **ambrosio** » Tue Nov 01, 2016 9:10 am

Great!