Can you share your config file?
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Ilie Soltanici Sent: quarta-feira, 3 de abril de 2019 14:34 To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org 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:nl7oe4ss@vjbh7r4im6j7.invalid;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:11@my-company.net mailto:sip%3A11@my-company.net >;tag=as1789445c
To: <sip:15@192.168.50.210:5060 http://sip:15@192.168.50.210:5060 >
Contact: <sip:11@192.168.50.237:5060 http://sip:11@192.168.50.237:5060 >
Call-ID: 7fc800de060197fa2315c93763873092@my-company.net mailto: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 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:15@192.168.50.210:5060 http://sip:15@192.168.50.210:5060 >;tag=dk4fa8ftt6
From: "WebRTC" <sip:11@my-company.net mailto:sip%3A11@my-company.net >;tag=as1789445c
Call-ID: 7fc800de060197fa2315c93763873092@my-company.net mailto: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 http://my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes>
Max-Forwards: 70
From: "WebRTC" <sip:11@my-company.net mailto:sip%3A11@my-company.net >;tag=as1789445c
To: <sip:15@192.168.50.210:5060 http://sip:15@192.168.50.210:5060 >;tag=dk4fa8ftt6
Contact: <sip:11@192.168.50.237:5060 http://sip:11@192.168.50.237:5060 >
Call-ID: 7fc800de060197fa2315c93763873092@my-company.net mailto: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 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.