to be transfered to a new Space - Using the WebRTC

On the PBX side the web sockets support is enabled. The performance testing was carried out by the SIP.js library. The sipml5 is also as a working option, but certain usage difficulties are associated with it (for example, a long collection of interface addresses leading to a delay in receiving/sending a call, a problem when working with non-ASCII characters), therefore SIP.js is strongly recommended.

WebRTC diagnostics

In Chrome browser you can use chrome://webrtc-internals/, this built-in service reflects the diagnostic capabilities of the webRTC RTCPeerConnection: the webRTC events, all possible statistics on the conversation, for example, data on the conversation quality (loss).

Also chrome://webrtc-internals/, allows you to record a conversation for further analysis of the conversation recording.