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

[Daniel-Constantin Mierla]

Ok. Let us know if you find out anything related to tls 1.3 that impacts
Kamailio.

Cheers,
Daniel

On 22.05.20 13:07, Chirag Desai wrote:
> Hi Daniel
>
> Thanks so much for your help. I'm going to roll with TLS 1.2 for now,
> until I find time to debug. I seem to have audio in both directions
> too! Now I can continue building around kamailio. Thank you :)
>
> Chirag

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