I have successfully configured webRTC in asterisk using ws protocol (JsSIP client). But I need to enable wss in asterisk to test voice calls via https protocol. Can anyone help me ?
Thanks in advance
I have successfully configured webRTC in asterisk using ws protocol (JsSIP client). But I need to enable wss in asterisk to test voice calls via https protocol. Can anyone help me ?
Thanks in advance
This uses the HTTPS support configurable in http.conf. Once TLS is configured and available there, then wss becomes available as well.
if you have a domain, i recommand use a letsencrypt!!
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