WebRTC 5-Channel Conferencing App

Job ID: 39957128

Budget: $30 – $250 USD

I’m looking for a lightweight, browser-based audio-conferencing application that creates five simultaneous audio channels—one host channel that drives the discussion and four guest channels that respond.

Interface 1 (host): opens in a browser, starts a session, generates join links, and sees live controls for every channel.
Interface 2 (guest): is simply the link each guest opens to join the conversation.

Essential channel controls
• Mute / un-mute
• Accept / reject incoming audio
• Individual volume fader
• Echo cancellation and automatic mic gain

Chat panel
• Room-wide text messaging
• Small file sharing (e.g., PDF, images)

Security
• Basic user authentication for both host and guests.

Tech notes
• WebRTC with an SFU; I rely on you to recommend and configure the optimal TURN/STUN setup for low latency and reliability.
• Must run efficiently on a small device; either Linux or Windows is acceptable.
• Feel free to leverage lightweight frameworks such as mediasoup, Janus, livekit or Jitsi if they meet the resource target. I am not sure if cloudflare maybe used but ideally will want all open source.

What I will provide
• A simple prototype mock-up of the intended UI.

What I need from you
1. A functional minimum-viable app that matches the prototype.
2. Source code, server configs, and build scripts.
3. A concise deployment guide so I can install and run it on the target device.

If you have questions or see a smarter way to hit the latency/reliability balance, please outline it in your proposal.