Every call encrypted in transit
Voice and video calls run over standard WebRTC transport encryption (DTLS-SRTP), with waiting rooms, host controls, and a clear notice whenever a meeting is being recorded.
Encrypted in transit, by default
Every call - whether it goes through our LiveKit media server or the peer-to-peer fallback - is encrypted in transit using WebRTC’s standard DTLS-SRTP, the same transport security every major video platform relies on. We describe this precisely as "encrypted in transit," not "end-to-end encrypted" - our media server needs to route audio/video between participants, the same way it does for screen sharing and recording.
- DTLS-SRTP transport encryption on every call, mesh or SFU
- Waiting rooms hold anyone but the host until admitted
- Host controls: mute, deafen, move, or disconnect any participant
You always know when you’re being recorded
A persistent "Recording" indicator shows for the whole call whenever the host has it on, and anyone joining who isn’t the host has to click through a one-time notice before they’re let into a recording-enabled meeting at all.
Bring the whole conversation together.
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