Hello,
We are trying to use Kamailio 4.2 with RTPEngine 3.3 behind NAT.
Somehow SIP responses (200 OK) are not handled by Kamailio.
The following INVITE is send to the receiving end (WebRTC Client):
Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on>
Via: SIP/2.0/UDP 104.155.11.255:5060;branch=z9hG4bK865.9db0c023bec2f314e89ae0f18b78d2f0.0
Via: SIP/2.0/UDP 10.240.215.73:5060;rport=5060;received=146.148.113.245;branch=z9hG4bK644600e2
Max-Forwards: 69
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as68f0a26a
Contact: <sip:anonymous@10.240.215.73:5060;alias=146.148.113.245~5060~1>
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Fri, 24 Oct 2014 08:38:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
P-hint: outbound
.......
The client response looks like this:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 104.155.11.255:5060;branch=z9hG4bK865.9db0c023bec2f314e89ae0f18b78d2f0.0;rport=5060;received=104.155.11.255
, SIP/2.0/UDP 10.240.215.73:5060;rport=5060;received=146.148.113.245;branch=z9hG4bK644600e2
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as68f0a26a
CSeq: 102 INVITE
Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on>
Content-Type: application/sdp
Content-Length: 938
.......
I have a xlog call as the first statement in kamailio routing script. I can see the request, which get's handled correctly but the response does not appear in kamailio log, though it arrives at the machine.
I am totally helpless, what might be wrong. If we use kamailio on a public accessable machine everything works as expected.
Maybe anyone might have had similiar problems using kamailio behind NAT.
Kind regards,
Marko