With the new Google Chrome update, removing support for TLS1.0 and TLS1.1, we have had an issue come up concerning our WebRTC which uses the jsSIP client.
The error in asterisk:
[2020-11-18 07:12:34] ERROR[C-00017194]: res_rtp_asterisk.c:2545 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f91884ebd00' due to reason 'tlsv1 alert protocol version', terminating [2020-11-18 07:12:34] WARNING[C-00017194]: res_rtp_asterisk.c:5300 ast_rtp_read: RTP Read error: Unspecified. Hanging up.
Our systems are build on Centos 6.6 and we use openssl 1.0.1e-fips - which supports TLS1.2. The Asterisk version we are using is: 13.19.1
We have ran SIP packet traces and can confirm the type was: DTLS1.0 coming from the Asterisk server
Any ideas on how to fix this issue?