WebRTC peer-to-peer design excels in low-latency videoconferencing. However, having multiple media streams poses challenges if you want to record videoconference or stream it to a larger audience. Doing that requires combining many video and audio tracks first. I’ll explore available tools for mixing live streams and explain why we created an open-source media server for doing that in real time - a LiveCompositor. I’ll dive into our approach for crucial problems like stream synchronization, handling network unreliability, optimizing performance, and designing developers-friendly APIs. Lastly, I’ll show how you can record WebRTC calls with LiveCompositor, using one of our projects as an example.