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Can't connect to WebRTC?
Good afternoon! I am building Aster 1.8.32.3 in conjunction with Webrtc on Centos 7 x86_64. I set it up according to this guide . There is a sip call between two PCs in LAN. Logs through asterisk cli are visible. But when I try to authorize on the web server, it says that it is not possible to connect to the server. HTML5 sip client deployed on my own web server.
Tell me what's the problem? What is wrong in the instructions? How to view connection attempt logs via webrtc?
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