I’m working on a lab project, to make call using SIPML5 and Asterisk.
I have implement, TLS for SIP and SRTP for data. Everything works fine with my softphone.
But I tried to make audio/video calls between browser (Chrome and FF) which only support right now DTLS-SRTP.
This is the conf of one of my user.
;; Configuration générale
Now I have this error in the asterisk console, when i tried to make a call.
*CLI> [Jul 21 11:44:43] ERROR[C-00000001]: res_rtp_asterisk.c:934 ast_rtp_dtls_set_configuration: Specified certificate file ‘/etc/asterisk/keys/asterisk.pem’ for RTP instance ‘0x7fd460039338’ could not be used
[Jul 21 11:44:43] ERROR[C-00000001]: chan_sip.c:5858 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP instance ‘0x7fd460039338’
[Jul 21 11:44:43] NOTICE[C-00000001]: chan_sip.c:25679 handle_request_invite: Failed to authenticate device "6001"sip:email@example.com;tag=nmwFc9oTJJIO1iEfWgqK
I can’t find anything to help me on internet. Maybe you could help me understand this issue ?