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

[Daniel-Constantin Mierla]

Hello,

I used:

openssl s_client -connect MYSERVER:5061 -tlsextdebug -tls1_3

and worked:

Post-Handshake New Session Ticket arrived:
SSL-Session:
    Protocol  : TLSv1.3
    Cipher    : TLS_AES_256_GCM_SHA384

It connected to Kamailio master branch running on a Debian Buster (10).

Might be something specific to your environment.

Cheers,
Daniel

On 21.05.20 02:48, Chirag Desai wrote:
> Hi Daniel,
>
> I solved it! In the tls.cfg file I changed 
> method = TLSv1.2+ 
> to 
> method = TLSv1.2
>
> It seems like Kamailio wasn't liking the TLS 1.3 connections. The
> documentation states it should work when using Open SSL 1.1.1 and I am
> indeed using that, so I'm not sure what went wrong. Could it be a bug?
>
> openssl version -a
> OpenSSL 1.1.1  11 Sep 2018
>
> I'm happy I have the WSS working now. I will test the audio shortly
> and report back.
>
> Thanks again for all your help,
>
> C

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