LiveKit utilizes WebRTC and WebSockets in its infrastructure as follows: • WebRTC: This is the primary transport protocol for real-time media, including audio, video, and data. It is optimized for media with advanced codecs and compression algorithms, ensuring ultra-low latency and network resilience. WebRTC is crucial for synchronized experiences like live sports viewing. • WebSockets: These are used specifically in integrations, such as with OpenAI's Realtime API, for low-latency audio streaming. LiveKit bridges WebSockets and WebRTC, converting WebSocket transport to WebRTC for better performance over long distances or slower networks. This hybrid approach leverages the strengths of both protocols to provide efficient and reliable communication in LiveKit's infrastructure. Sources: LiveKit | LiveKit Docs