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

Daniel-Constantin Mierla miconda at gmail.com
Wed May 20 08:51:44 CEST 2020


Hello,

can you try to connect with openssl client tool and see if you get more
hints in the output? Like:

openssl s_client -connect sipserver.com:5061 -tlsextdebug

Cheers,
Daniel

On 19.05.20 21:35, Chirag Desai wrote:
> 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.

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla

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