In this presentation, the speaker argues that traditional WebRTC debugging methods—asking users to open Chrome's WebRTC internals, download dump files, and send them—are impractical, especially for high‑value users whose issues may have occurred in the past. He introduces the open‑source RTCStats client SDK as a solution that continuously collects live session statistics and streams them to a server. This enables support teams to instantly access detailed logs and observations for any reported problem, allowing rapid root‑cause identification such as CPU overload, network issues, failed ICE candidates, or firewall blocks. The speaker emphasizes the time savings and efficiency gains when the actual cause of a WebRTC failure is just a click away from a support ticket.
