[SR-Users] Error on ACK/BYE After 200

[Neimar]

Hello, 

I am trying to set up a Webrtc by using Kamailio, Asterisk 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 from Asterisk, 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. 

In console, I see: 
Sep 21 20:39:06 vsphone3-sbc /usr/sbin/kamailio[9756]: WARNING: {1 102 ACK 332ee81b4eb178ac156e8d3e53a98900 at 172.16.11.6:5060} <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:172.16.11.57:8443, to udp:187.20.56.83:27730) 


My scenario is: 
Internet -> Firewall -> Kamailio -> Asterisk 

Working: 
Softphone -> Kamailio -> Asterisk (TCP/UDP/TLS(SRTP) 
Asterisk -> Kamailio -> Softphone (TCP/UDP/TLS(SRTP)) 
Webrtc -> Kamailio -> Asterisk (WSS) 

Not Working: 
Asterisk -> Kamailio -> Webrtc 

After a answer from Webrtc, I can listen sound, so, RTP in theory is fine. 

Webrtc send 200 OK to Kamailio 
Kamailio Send 200 OK to Asterisk. 
Asterisk Send ACK to Kamailio 
ACK not forwarded to Webrtc. 





In attachment, my kamailio.cfg 

Every Help is welcome. 

Thanks 

Neimar. 
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