WebRTC Transport
Implement WebRTC rooms, participants, tracks, and real-time streaming features.
All WebRTC Transport Guides
CreateRoom() is taking too long
Learn why CreateRoom() API calls can be slow and how to avoid explicit room creation by leveraging automatic room creation on participant join.
WebRTC Transport
Debugging audio/video sync issues in LiveKit publishing
Learn how to identify and fix audio/video synchronization drift issues when publishing to LiveKit, including root causes and best practices.
WebRTC Transport
Diagnosing Connection Errors with the Connection Test Utility
Learn how to use the Connection Test utility to diagnose and troubleshoot connectivity problems with LiveKit Cloud, including WebRTC, WebSocket, and TURN connectivity checks.
WebRTC Transport
Ensuring Unique Room IDs with Dynamic Room Names
Learn why recreating rooms with the same name can cause Room ID reuse, and how to guarantee unique Room IDs by adding dynamic suffixes to your room names.
WebRTC Transport
Firewall Tips for LiveKit Connectivity
Understand how media connectivity works, troubleshoot firewall issues, and configure TURN for firewall-sensitive environments.
WebRTC Transport
Handling asynchronous metadata updates in LiveKit
Learn how LiveKit propagates room and participant metadata through event-driven architecture, and best practices for handling timing variations across distributed SFU nodes.
WebRTC Transport
How is Connection Quality determined?
Learn how LiveKit calculates connection quality scores (Poor, Good, Excellent) using packet loss, video layers, and bitrates.
WebRTC Transport
Understanding and Estimating Pricing for Video Conference and Livestream Use Cases
Learn how to estimate costs for LiveKit Cloud video conferencing and livestream applications, including connection minutes, bandwidth, and transcoding.
WebRTC Transport
Understanding the STATE_MISMATCH Disconnect Reason
Learn what the STATE_MISMATCH disconnect reason means in LiveKit, including its causes, how it differs from PEER_CONNECTION_DISCONNECT, and what triggers this server-side response.
WebRTC Transport