16 Oct
2020
16 Oct
'20
11:07 a.m.
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