A core solution for reducing real-time audio and video latency
LiveKit achieves low latency of 100 milliseconds with a combination of the following technologies:
- WebRTC Native SupportBuilt-in UDP transmission and NAT penetration technology reduces latency by more than 80% compared to traditional RTMP protocols.
- SFU architecture optimization: Selective forwarding units relay only the necessary media streams, avoiding decoding/multiple encoding delays in the MCU architecture
- Proximity Access Strategy: LiveKit Cloud automatically allocates edge nodes, reducing latency by 5-10ms for every 1000km of physical distance reduction
Specific implementation steps
- Select WebRTC output (instead of RTMP) in OBS pushstream settings
- invocations
Room.connect()Specifies the nearest area when{ region: 'ap-southeast-1' } - pass (a bill or inspection etc)
room.getStats()Monitor end-to-end latency metrics
Advanced Optimization Tools
- Enable SIMULCAST function: automatically adapts to different terminal bandwidths
- Configure TURN server backup links (required for self-hosting)
- utilization
priorityParameters mark important packets
This answer comes from the articleLiveKit: an open source tool for building real-time audio and video applicationsThe
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