We have tried to solve this problem for a while now, and extensive research yielded no results, so I decided that my best bet is this post.
While calls work fine in Chrome (with the exception of Hold which crashes Asterisk), they do not work in Firefox.
(We did not use WebRTC2SIP (just the RC Asterisk that incorporates the patch))
Firefox requires the connection to be set via DTLS-SRTP, for which we had to generate certificates via OpenSSL (.pem)
We have used this information to get those certificates:
After they were generated, they were added to the device via the DTLS settings:
dtlsenable = yes
dtlsverify = no
dtlscertfile=/etc/asterisk/keys/softphone.pem
dtlsprivatekey=/etc/asterisk/keys/key.pem
dtlscafile=/etc/asterisk/keys/key.pem
Asterisk did not display any errors, but the call could not be made (SipML would just say Call in Progress and do nothing).
Asterisk would not accept the call ( Specified certificate file '/usr/local/certs/callision.com.crt' for RTP instance '0x7f69a000ea68' could not be used )
I would highly appreciate your help on the matter, as it has been haunting us for the past two months.