WebRTC

API Testing

LiveKit Testing Guide: How to Test Real-Time Video and Audio Apps

LiveKit is an open-source infrastructure for real-time audio, video, and data — used by teams building video conferencing, live streaming, voice agents, and collaborative applications. Testing LiveKit-powered apps is notoriously tricky: WebRTC is browser-dependent, media streams are asynchronous, and real room behavior requires multiple connected participants. This

Testing

WebRTC Testing Guide: Unit Tests, Integration with Playwright, and Media Quality

WebRTC applications are notoriously hard to test because they involve browser APIs, network negotiation, and real-time media. This guide covers mocking RTCPeerConnection in Jest, intercepting WebRTC in Playwright, measuring media quality with getStats(), and simulating adverse network conditions. WebRTC powers video calls, screen sharing, and peer-to-peer data