Hello,
I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine. So far, everything is working fine, I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. This is how invite is looking:
INVITE sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss SIP/2.0 Record-Route: sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 Max-Forwards: 70 From: "WebRTC" sip:11@my-company.net;tag=as1789445c To: sip:15@192.168.50.210:5060 Contact: sip:11@192.168.50.237:5060 Call-ID: 7fc800de060197fa2315c93763873092@my-company.net CSeq: 102 INVITE User-Agent: Proxy Date: Wed, 03 Apr 2019 17:11:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Alert-Info: Content-Type: application/sdp Content-Length: 596 Server: SIP Proxy
and then WebRTC app is replying with 200 OK:
SIP/2.0 200 OK Record-Route: sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 To: sip:15@192.168.50.210:5060;tag=dk4fa8ftt6 From: "WebRTC" sip:11@my-company.net;tag=as1789445c Call-ID: 7fc800de060197fa2315c93763873092@my-company.net CSeq: 102 INVITE Contact: sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Proxy-WEBRTC Content-Type: application/sdp Content-Length: 901
and finally, Kamailio is trying to send this ack through UDP protocol:
ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0 Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport Route: sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes Max-Forwards: 70 From: "WebRTC" sip:11@my-company.net;tag=as1789445c To: sip:15@192.168.50.210:5060;tag=dk4fa8ftt6 Contact: sip:11@192.168.50.237:5060 Call-ID: 7fc800de060197fa2315c93763873092@my-company.net CSeq: 102 ACK User-Agent: Proxy Content-Length: 0
If i'm trying to force it through TLS, i'm receiving error: get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443, to udp:22.22.22.22:23317)
Can someone point me in the right direction, please? Thank you.