AnonConnect

Truly anonymous real-time chat & video calls — no database, no accounts, no traces.

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What is AnonConnect?

AnonConnect is a production-ready anonymous communication platform. You get a random identity the moment you open the app — no signup, no login, no data stored anywhere. Every message, room, and call lives in server memory only and vanishes on restart.

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Features

💬 Chat

• Random anonymous usernames auto-generated on connect (e.g. SilentPanda42)
• Create or join named rooms — switch freely anytime
• Optional password-protected private rooms
• Last 50 messages synced instantly when you join a room
• Live typing indicators
• Emoji picker
• Message reactions (broadcast live to room)
• Inline image sharing via base64 (< 2MB, no cloud upload)
• Disappearing messages — room-level TTL: off / 5min / 1hr / 24hr
• Empty rooms auto-deleted after 5 minutes

📹 Voice & Video Calls

• Peer-to-peer WebRTC calls — server only handles signaling
• Up to 4 participants per call room
• Mute / camera toggle with live status broadcast
• Screen sharing
• In-call side chat panel
• Live call duration timer
• 6-character shareable room codes

🔒 Security

• helmet — secure HTTP headers
• express-rate-limit — 100 requests / 15 min per IP
• Per-socket rate limit — 30 messages / min
• Zero persistence — no DB, no logs, no cookies
• Configurable CORS via CORS_ORIGIN env variable

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Quick Start

git clone https://github.com/your-username/AnonConnect.git
cd AnonConnect
npm install
cp .env.example .env
npm start

Open → http://localhost:3000

For development with auto-restart:

npm run dev

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Environment Variables

Variable	Default	Description
PORT	3000	Server port
NODE_ENV	development	Runtime mode
CORS_ORIGIN	*	Allowed origin (lock down in production)

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Project Structure

AnonConnect/
├── server.js                  # Express + Socket.io backend
├── package.json
├── .env.example
└── public/
    ├── index.html             # Landing page
    ├── chat.html              # Chat room page
    ├── call.html              # Voice/video call page
    ├── css/
    │   ├── main.css           # Global theme + variables
    │   ├── chat.css           # Chat interface
    │   ├── call.css           # Call interface
    │   └── components.css     # Reusable components
    └── js/
        ├── socket-client.js   # Socket.io connection manager
        ├── chat.js            # Chat logic
        ├── webrtc.js          # WebRTC peer connection
        ├── call.js            # Call UI + controls
        └── utils.js           # Helpers + utilities

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Socket Events

/chat Namespace

Event	Direction	Description
welcome	server → client	Assigns random username on connect
join-room	client → server	Join a named room
join-room-error	server → client	Wrong password or room not found
send-message	client → server	Send text message to room
send-file	client → server	Send base64 image to room
add-reaction	client → server	Add emoji reaction to a message
reaction-updated	server → room	Reactions changed on a message
message-deleted	server → room	TTL expired, message removed
typing-start	client → server	User started typing
typing-stop	client → server	User stopped typing
get-rooms	client → server	Fetch active room list
create-room	client → server	Create new room with options
online-count	server → all	Total connected users updated
rate-limit-hit	server → client	Message rate limit exceeded

/call Namespace

Event	Direction	Description
join-call-room	client → server	Join call room by code
webrtc-offer	client → server	Forward SDP offer to peer
webrtc-answer	client → server	Forward SDP answer to caller
webrtc-ice-candidate	client → server	Relay ICE candidate between peers
call-request	client → server	Notify room of incoming call
call-accepted	client → server	Callee accepted the call
call-rejected	client → server	Callee rejected the call
call-ended	client → server	End call for all participants
mute-toggle	client → server	Broadcast mic mute state
video-toggle	client → server	Broadcast camera state

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Architecture

Browser ──── Socket.io ────► Node.js Server
   │          (signaling)     (in-memory only)
   │                               │
   └──── WebRTC P2P ───────────────┘
         (direct media,
          no server relay)

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Production Notes

• HTTPS required — WebRTC won't work on plain HTTP in modern browsers
• TURN servers — Add Coturn for users behind strict NATs/firewalls
• Scaling — Use @socket.io/redis-adapter for multi-instance deployments
• CORS — Set CORS_ORIGIN to your exact domain, never leave * public

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License

MIT — do whatever you want with it.