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

Chirag Desai c.desai at rxhost.co.uk
Tue May 19 21:35:54 CEST 2020


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
certificate = /etc/letsencrypt/live/sip.mydomain.com/fullchain.pem
server_name = 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 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.
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