Re: [SR-Users] No audio when calling from SIP to WEBRTC

Hi Daniel, Thanks for the response. I'm sure I have everything set up correctly. Here's what's in my tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/sip.mydomain.com/privkey.pem <http://sip.mydomain.com/privkey.pem> certificate = /etc/letsencrypt/live/sip.mydomain.com/fullchain.pem <http://sip.mydomain.com/fullchain.pem> server_name = sip.mydomain.com <http://sip.mydomain.com> Here's my kamailio.cfg  #!ifdef WITH_TLS         # ----- tls params -----         modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")         #!endif If I run cat /etc/letsencrypt/live/sip.mydomain.com/privkey.pem <http://sip.mydomain.com/privkey.pem> I can see the contents of the file. The permissions for the certificates are quite liberal too, so there shouldn't be any issues there. Any other ideas? Thanks so much for your help.