When WebRTC quality degrades on your SFU (jitter spikes, frozen frames, choppy audio), your monitoring tells you something is wrong. It doesn’t tell you what is responsible. Is the network dropping packets? Or is your server too loaded to read them from the socket on time? The
Last year, I attended the RTC.ON conference organized by Software Mansion for the first time. I shared my take on the conference in this WebRTC.ventures blog post: A WebRTC Developer’s Take on RTC.ON 2024. I also spoke at the 2024 conference, a recording of that talk is
Session Initiation Protocol (SIP) and WebRTC are both essential technologies in the field of real-time communications, particularly for voice and video over IP (VoIP). While they serve complementary roles, they operate differently and have distinct functionalities. In this post, we explore how to architect the integration of
WebRTC is more popular than ever, and is supported by more browsers than ever. But that doesn't make building with WebRTC any easier!
