Roadmap

These are plans, not promises. Nothing here is committed to a date, and the order may change.

Hosted per-participant meetings on Asterisk and FreeSWITCH

Today per-participant audio comes through the WebRTC worker and LiveKit, while a PBX conference reaches the engine as one mixed stream. The aim is to host meetings on a PBX and receive each participant’s audio separately, with gateway adapters for both Asterisk and FreeSWITCH alongside the existing rustpbx adapter. What each PBX can expose per participant differs, and the plan depends on that.

Platform connectors

Connectors that let a flow take part in meetings on hosted platforms, through the platforms’ own supported media interfaces:

• Zoom, via real-time media streams (RTMS).
• Microsoft Teams, via media bots.
• Google Meet, via the Meet Media API.
• Post-meeting artifacts: fetching the platform’s own recordings and transcripts after a meeting for the flow to process.

Each platform has its own approval, hosting and consent requirements, and each connector would arrive as a gateway behind the same media-stream protocol: no new BPMN element.

Smaller items

• Video recording from the WebRTC worker to the object store.
• Phone participants through LiveKit SIP.
• Several rooms per worker.

Follow progress in the repository and its changelog.