[SR-Users] WebRTC ACK Protocol

[Ilie Soltanici]

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 at 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 at my-company.net>;tag=as1789445c
To: <sip:15 at 192.168.50.210:5060>
Contact: <sip:11 at 192.168.50.237:5060>
Call-ID: 7fc800de060197fa2315c93763873092 at 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 at 192.168.50.210:5060>;tag=dk4fa8ftt6
From: "WebRTC" <sip:11 at my-company.net>;tag=as1789445c
Call-ID: 7fc800de060197fa2315c93763873092 at my-company.net
CSeq: 102 INVITE
Contact: <sip:nl7oe4ss at 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 at 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 at my-company.net>;tag=as1789445c
To: <sip:15 at 192.168.50.210:5060>;tag=dk4fa8ftt6
Contact: <sip:11 at 192.168.50.237:5060>
Call-ID: 7fc800de060197fa2315c93763873092 at 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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20190403/cf439cad/attachment.html>