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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
   * Recorder/Playout
   * 
   * Screen Sharing
   * NoSIP (SDP/RTP)
   * 
   * Device Selection
   * End-to-end Encryption
   * Multichannel Opus (surround)
   * Canvas Capture
   * Web Audio Processing
   * Virtual Background
   * 
   * 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 previous 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.

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Janus WebRTC Server © Meetecho 2014-2023