A WebRTC based tool to support low latency audio conferencing. This is targeted to allow musicians to be able to jam together.
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988 B

web server component -- rendezvous part, and mixing/aggregation

web client component -- connect to server, send encoded data

Meeting setup flow: 1) Admin uses server to generate room. The server generates a URL and the admin gives out url to participants. As part of this, restrictions may be applied, such as when the meeting is valid.

Participant flow: 1) Participant goes to meeting url obtained from admin. 2) Once meeting is valid, setup the following: a) WebRTC data connection to server b) Input audio c) Output audio 3) Pass the setup items to do encoding/decoding and playback.

Server meeting flow: 1) receive RTC request 2) start up RTC and add audio source to channel 3) mix in audio to channel and send it back to each participant

File information: src/index.js -- main file for web page

Random references: navigator.mediaDevices.getSupportedConstraints(); https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getSupportedConstraints