Simple way to show this problem without any WebRTC SIP client is to point Chrome browser to K's TLS listening port:
https://<sip proxy>:5061 and look with wireshark or tshark how the handshake gets terminated by Chrome right after Server Hello. The same with Firefox works fine. -- Juha _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
