Low-latency audio and video that holds through a weak connection
The audio and video in every call and meeting are low-latency and private.
- A person asks to join a session on your deployment
- The host approves them before they enter
- Audio and video connect, encrypted while they travel
- The call holds through a weak link and comes back after a short drop
How the audio and video travel
The media runs over WebRTC, the browser's own standard for real-time audio and video, at a latency low enough for a natural back and forth in a one to one call or a larger room.
Sessions are private from the start, and audio and video are encrypted while they travel between participants. All of it is self-hosted, answering on the organisation's own domain.
Calls continue through weak connections
Bandwidth changes during a session and connections drop. A call continues through weak connections and short drops, and the deployment grows as you add sites and more people join.
- Calls stay steady and come back after a short drop
- Capacity grows as your organisation adds sites
What holds in every session
- Who is in the session
- The host approves each person before they join. Only approved people can see and hear a session, so a call stays private to the people you invited.
- What a participant controls
- Each participant controls their own microphone and camera during a session, so people share only what they intend to share.
- Which kinds of session use it
- Meetings, calls, classrooms, shared cinema, and activity rooms all use the same live audio and video in the same deployment, and every session gets the same secure, private setup.