I have asterisk 13.2 installed in ubuntu 14.04. I was success in register and setup call from my browser and i can hear the audio. But now call gets disconnected when audio start playing after call connected when i updated my browser to chrome version 54. I tired with firefox, it works fine.
This is the error i get
[2016-08-29 11:27:58] ERROR[C-00000003]: res_rtp_asterisk.c:2042 __rtp_recvfrom: DTLS failure occurred on RTP instance ‘0x7fb7e402e778’ due to reason ‘sslv3 alert handshake failure’, terminating
[2016-08-29 11:27:58] WARNING[C-00000003]: res_rtp_asterisk.c:3911 ast_rtcp_read: RTCP Read error: Unspecified. Hanging up.
[2016-08-29 11:27:58] WARNING[C-00000003]: app_playback.c:493 playback_exec: Playback failed on SIP/6001-00000003 for demo-congrats
I googled this issue and found that it was problem with openssl. So i updated my openssl from 1.0.1f to 1.0.1t referring this link. I also rebuild my asterisk 13.2 referring this link. I am using sipml5 to make browser call.
Can any one please help me to solve this issue.
This is my extension detail
generated cert file using below command
sudo ./ast_tls_cert -C pbx.mycompany.com -O “My Super Company” -d /etc/asterisk/keys