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Janus (multistream) * Home * Demos * Index * * Echo Test * Streaming * Video Call * SIP Gateway * Video Room * Video Room (multistream) * Audio Bridge * Text Room * Voice Mail * Recorder/Playout * * Screen Sharing * NoSIP (SDP/RTP) * * Device Selection * End-to-end Encryption * Multichannel Opus (surround) * Canvas Capture * Web Audio Processing * Virtual Background * VP9-SVC Video Room * * Admin/Monitor * Documentation * Papers * Need help? * Janus (0.x) * JanusCon! * ABOUT JANUS: THE GENERAL PURPOSE WEBRTC SERVER Note Well: these are the demos and documentation for the multistream version of Janus, which is a new version. If you want to check the current version of Janus instead (i.e., 0.x, a.k.a. "legacy") click here instead. Janus is a WebRTC Server developed by Meetecho conceived to be a general purpose one. As such, it doesn't provide any functionality per se other than implementing the means to set up a WebRTC media communication with a browser, exchanging JSON messages with it, and relaying RTP/RTCP and messages between browsers and the server-side application logic they're attached to. Any specific feature/application is provided by server side plugins, that browsers can then contact via Janus to take advantage of the functionality they provide. Example of such plugins can be implementations of applications like echo tests, conference bridges, media recorders, SIP gateways and the like. The reason for this is simple: we wanted something that would have a small footprint (hence a C implementation) and that we could only equip with what was really needed (hence pluggable modules). That is, something that would allow us to deploy either a full-fledged WebRTC gateway on the cloud, or a small nettop/box to handle a specific use case. Check the Documentation for more details about Janus, or check out the Demos to see it in action. -------------------------------------------------------------------------------- Janus WebRTC Server © Meetecho 2014-2022