Can you share your config file?
From: sr-users [mailto:[email protected]] On Behalf Of Ilie Soltanici Sent: quarta-feira, 3 de abril de 2019 14:34 To: Kamailio (SER) - Users Mailing List <[email protected]> Subject: [SR-Users] WebRTC ACK Protocol 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:[email protected];transport=wss SIP/2.0 Record-Route: <sip:my-company.net <http://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:[email protected] <mailto:sip%[email protected]> >;tag=as1789445c To: <sip:[email protected]:5060 <http://sip:[email protected]:5060> > Contact: <sip:[email protected]:5060 <http://sip:[email protected]:5060> > Call-ID: [email protected] <mailto:[email protected]> 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 <http://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:[email protected]:5060 <http://sip:[email protected]:5060> >;tag=dk4fa8ftt6 From: "WebRTC" <sip:[email protected] <mailto:sip%[email protected]> >;tag=as1789445c Call-ID: [email protected] <mailto:[email protected]> CSeq: 102 INVITE Contact: <sip:[email protected];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:[email protected]:57421;transport=wss SIP/2.0 Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport Route: <sip:my-company.net <http://my-company.net> ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Max-Forwards: 70 From: "WebRTC" <sip:[email protected] <mailto:sip%[email protected]> >;tag=as1789445c To: <sip:[email protected]:5060 <http://sip:[email protected]:5060> >;tag=dk4fa8ftt6 Contact: <sip:[email protected]:5060 <http://sip:[email protected]:5060> > Call-ID: [email protected] <mailto:[email protected]> 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 <http://123.123.123.123:10443> , to udp:22.22.22.22:23317 <http://22.22.22.22:23317> ) Can someone point me in the right direction, please? Thank you.
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