Hello Everyone, how are you ?
Have someone a step-by-step about how I configure it scenary behind : (client )—> (webrtcproxy) — → (myserver asterisk ). I just need get a way to use webrtc sip phone service, because I dont have a display browser phone in php + html. So a way that I found to use sip phone cliente free is use webrtc services from freepbx. Can someone help me ?
I would not recommend it. I could never get the build in client to work correctly due to how I have the environment segregated. I could use other clients such as the one from Queuemetrics. and Bria and zoiper. Even Mizu worked mostly. I would like to try another client in the future such as round cube.
I’m using a 3rd party proxy from Ingate. However I might not buy any more of them due to how much they have increased license cost by. Up to version 6.3 it was fair pricing. 6.4 and later they want so much money for a webrtc session its not feasible to do. Kamilio is an option but I’m in a corporate environment and its much better to be able to open a ticket and use there knowledge to help work the issue.
for reference, you would be either doing sip TLS or a webrtc over sip. Webrtc is browser based, Sip tls is encrypted sip in any common Client.
Once you decide on a path then its a matter of following the steps dictated by the path.
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