hi guys, i am trying to figure it out which protocol is more secure, for webrtc for the moment i am using:
Transport Sip Server = UDP
Transport = 0.0.0.0-ws
Media Encryption = DTLS-SRTP
Enable DTLS = YES
DTLS VERIFY = NO
With these configuration is my traffic encrypted and if not what is the best way to set it up. Thanks
WebRTC endpoints mandate utilizing DTLS-SRTP for media encryption, and, at least with Asterisk/FreePBX use SIP over websocket for the SIP signaling for the call. You can use an SSL cert to make sure that the websocket signaling traffic is encrypted as well if you’d like.
Matthew Fredrickson
I have asterisk 16.20.0 and freepbx 15, i only enabled /wss but when i register my device it says : Added contact 'sip:66e5qhps@myip:2238;transport=ws . I am not sure why is it still using /ws and not /wss. I am using https bind port 8089 ?
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