diff --git a/scrolls/voice/mediasoup/latest/.scroll/private/app.js b/scrolls/voice/mediasoup/latest/.scroll/private/app.js new file mode 100644 index 00000000..a2a64368 --- /dev/null +++ b/scrolls/voice/mediasoup/latest/.scroll/private/app.js @@ -0,0 +1,191 @@ +// Druid Voice Server - DruidUI Client +// WebRTC client using mediasoup-client + +class VoiceClient { + constructor() { + this.ws = null; + this.device = null; + this.producerTransport = null; + this.consumerTransport = null; + this.producer = null; + this.consumers = new Map(); + this.connected = false; + this.muted = false; + this.deafened = false; + + this.initUI(); + this.connect(); + } + + initUI() { + // Status indicator + this.statusIndicator = document.getElementById('status-indicator'); + this.statusText = document.getElementById('status-text'); + + // Stats + this.statUsers = document.getElementById('stat-users'); + this.statChannels = document.getElementById('stat-channels'); + this.statUptime = document.getElementById('stat-uptime'); + + // Buttons + this.btnJoin = document.getElementById('btn-join'); + this.btnMute = document.getElementById('btn-mute'); + this.btnDeafen = document.getElementById('btn-deafen'); + + // Log + this.logElement = document.getElementById('log'); + + // Button handlers + this.btnJoin.addEventListener('click', () => this.joinVoice()); + this.btnMute.addEventListener('click', () => this.toggleMute()); + this.btnDeafen.addEventListener('click', () => this.toggleDeafen()); + + // Start uptime counter + this.startTime = Date.now(); + setInterval(() => this.updateUptime(), 1000); + } + + log(message, type = 'info') { + const now = new Date(); + const timestamp = now.toTimeString().split(' ')[0]; + const entry = document.createElement('div'); + entry.className = 'log-entry'; + entry.innerHTML = `[${timestamp}] ${message}`; + this.logElement.appendChild(entry); + this.logElement.scrollTop = this.logElement.scrollHeight; + console.log(`[${type}]`, message); + } + + updateUptime() { + const uptime = Math.floor((Date.now() - this.startTime) / 1000); + const hours = Math.floor(uptime / 3600); + const minutes = Math.floor((uptime % 3600) / 60); + const seconds = uptime % 60; + this.statUptime.textContent = `${hours.toString().padStart(2, '0')}:${minutes.toString().padStart(2, '0')}:${seconds.toString().padStart(2, '0')}`; + } + + connect() { + const protocol = window.location.protocol === 'https:' ? 'wss:' : 'ws:'; + const wsUrl = `${protocol}//${window.location.host}`; + + this.log(`Connecting to ${wsUrl}...`); + + this.ws = new WebSocket(wsUrl); + + this.ws.onopen = () => { + this.log('✅ Connected to voice server'); + this.connected = true; + this.statusIndicator.className = 'status-indicator online'; + this.statusText.textContent = 'Connected'; + this.btnJoin.disabled = false; + + // Request router capabilities + this.send({ type: 'getRouterRtpCapabilities' }); + }; + + this.ws.onclose = () => { + this.log('❌ Disconnected from voice server'); + this.connected = false; + this.statusIndicator.className = 'status-indicator offline'; + this.statusText.textContent = 'Disconnected'; + this.btnJoin.disabled = true; + + // Reconnect after 3 seconds + setTimeout(() => this.connect(), 3000); + }; + + this.ws.onerror = (error) => { + this.log(`WebSocket error: ${error}`, 'error'); + }; + + this.ws.onmessage = (event) => { + const message = JSON.parse(event.data); + this.handleMessage(message); + }; + } + + send(data) { + if (this.ws && this.ws.readyState === WebSocket.OPEN) { + this.ws.send(JSON.stringify(data)); + } + } + + handleMessage(message) { + switch (message.type) { + case 'routerRtpCapabilities': + this.log('Received router RTP capabilities'); + // Store for later use when joining + this.routerRtpCapabilities = message.data; + break; + + case 'transportCreated': + this.log(`Transport created: ${message.data.id}`); + break; + + case 'transportConnected': + this.log('Transport connected'); + break; + + case 'produced': + this.log(`Producer created: ${message.data.id}`); + break; + + case 'consumed': + this.log(`Consumer created: ${message.data.id}`); + break; + + case 'error': + this.log(`Server error: ${message.error}`, 'error'); + break; + } + } + + async joinVoice() { + try { + this.log('🎙️ Joining voice channel...'); + this.btnJoin.disabled = true; + this.btnJoin.textContent = 'Connecting...'; + + // Get user microphone + const stream = await navigator.mediaDevices.getUserMedia({ audio: true }); + this.log('✅ Microphone access granted'); + + // In a real implementation, we would: + // 1. Load mediasoup-client library + // 2. Create Device with router capabilities + // 3. Create transports + // 4. Produce audio track + // 5. Consume other users' audio + + // For now, just update UI + this.btnJoin.textContent = 'Connected'; + this.btnMute.disabled = false; + this.btnDeafen.disabled = false; + + this.statUsers.textContent = '1'; + this.log('✅ Connected to voice channel'); + + } catch (error) { + this.log(`❌ Failed to join: ${error.message}`, 'error'); + this.btnJoin.disabled = false; + this.btnJoin.textContent = 'Join Voice Channel'; + } + } + + toggleMute() { + this.muted = !this.muted; + this.btnMute.textContent = this.muted ? 'Unmute' : 'Mute'; + this.log(this.muted ? '🔇 Microphone muted' : '🔊 Microphone unmuted'); + } + + toggleDeafen() { + this.deafened = !this.deafened; + this.btnDeafen.textContent = this.deafened ? 'Undeafen' : 'Deafen'; + this.log(this.deafened ? '🔇 Audio deafened' : '🔊 Audio enabled'); + } +} + +// Initialize when page loads +document.addEventListener('DOMContentLoaded', () => { + new VoiceClient(); +}); diff --git a/scrolls/voice/mediasoup/latest/.scroll/private/index.html b/scrolls/voice/mediasoup/latest/.scroll/private/index.html new file mode 100644 index 00000000..22d68b8e --- /dev/null +++ b/scrolls/voice/mediasoup/latest/.scroll/private/index.html @@ -0,0 +1,240 @@ + + + + + + Druid Voice Server - DruidUI + + + +
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🎙️ Druid Voice Server

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Powered by mediasoup WebRTC SFU - Self-Hosted Voice Chat

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🔊 General Voice

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+ [00:00:00] Druid Voice Server initializing... +
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+ + + + diff --git a/scrolls/voice/mediasoup/latest/PLATFORM_REQUIREMENTS.md b/scrolls/voice/mediasoup/latest/PLATFORM_REQUIREMENTS.md new file mode 100644 index 00000000..1f1cc93b --- /dev/null +++ b/scrolls/voice/mediasoup/latest/PLATFORM_REQUIREMENTS.md @@ -0,0 +1,302 @@ +# Platform Requirements for Voice Server Support + +This document outlines what needs to be added to the Druid platform to support WebRTC-based voice chat in DruidUI. + +## WIT Interface: `druid-ui-webrtc.wit` + +The WebAssembly Interface Types (WIT) definition provides the contract between DruidUI WASM components and the host platform. + +## Required Platform Features + +### 1. Media Device Access + +**APIs to implement:** +``` +get-user-media(constraints) -> media-stream +enumerate-devices() -> list +``` + +**Host responsibilities:** +- Request browser permissions for microphone access +- Enumerate available audio input/output devices +- Create MediaStream objects and pass handles to WASM +- Handle device switching/hot-plugging + +**Browser APIs needed:** +- `navigator.mediaDevices.getUserMedia()` +- `navigator.mediaDevices.enumerateDevices()` + +--- + +### 2. RTCPeerConnection + +**APIs to implement:** +``` +create-peer-connection(config) -> peer-connection +peer-connection.create-offer() -> session-description +peer-connection.create-answer() -> session-description +peer-connection.add-ice-candidate(candidate) +``` + +**Host responsibilities:** +- Create and manage RTCPeerConnection instances +- Handle SDP offer/answer negotiation +- Process ICE candidates +- Manage connection state transitions +- Relay events to WASM component + +**Browser APIs needed:** +- `new RTCPeerConnection(config)` +- `createOffer()` / `createAnswer()` +- `setLocalDescription()` / `setRemoteDescription()` +- `addIceCandidate()` + +--- + +### 3. Audio Track Control + +**APIs to implement:** +``` +audio-track.set-enabled(bool) // Mute/unmute +audio-track.set-volume(f32) // Volume control +audio-track.get-audio-level() // VU meter +``` + +**Host responsibilities:** +- Enable/disable audio tracks +- Apply volume transformations +- Calculate audio levels for visualization +- Handle audio routing (which speakers, etc.) + +**Browser APIs needed:** +- `MediaStreamTrack.enabled` +- `GainNode` for volume control (Web Audio API) +- `AnalyserNode` for audio level detection + +--- + +### 4. WebSocket Signaling + +**APIs to implement:** +``` +websocket-connect(url) -> websocket +websocket.send-text(string) +websocket.send-binary(bytes) +``` + +**Host responsibilities:** +- Establish WebSocket connections +- Send/receive signaling messages +- Handle connection lifecycle +- Relay messages to WASM component + +**Browser APIs needed:** +- `new WebSocket(url)` +- `send()` for text/binary +- Event handlers: `onopen`, `onmessage`, `onerror`, `onclose` + +--- + +### 5. Event System + +**Event flow:** +``` +Browser Event → Host Platform → WASM Component + ↓ ↓ ↓ +ICE Candidate → webrtc-event → handle-webrtc-event() +Remote Track → track-added → render update +WS Message → ws-event → handle-ws-event() +``` + +**Host responsibilities:** +- Register event listeners on browser APIs +- Serialize events into WIT-compatible format +- Call exported WASM event handlers +- Manage event queue/batching for performance + +--- + +## Implementation Priority + +### Phase 1: Minimum Viable (voice-only, no UI controls) +1. `get-user-media` (audio only) +2. `create-peer-connection` +3. SDP offer/answer +4. ICE candidates +5. WebSocket signaling + +**Deliverable:** Basic voice connection between two peers + +--- + +### Phase 2: Audio Control +1. Mute/unmute +2. Audio level detection +3. Volume control +4. Device enumeration + +**Deliverable:** User can mute, adjust volume, see who's talking + +--- + +### Phase 3: Advanced Features +1. Multi-peer connections (SFU integration) +2. Screen sharing (video tracks) +3. Recording capabilities +4. Network quality metrics + +**Deliverable:** Full-featured voice chat + +--- + +## Security Considerations + +### Permissions +- All media device access requires explicit user permission +- WASM components cannot bypass browser security model +- Host must enforce permission checks before granting access + +### Sandboxing +- WASM components get **handles** to resources, not direct access +- Host owns all WebRTC objects (PeerConnection, MediaStream) +- Component can only invoke operations via WIT interface +- No access to raw UDP/TCP sockets + +### Data Privacy +- Audio data flows through browser's WebRTC stack +- WASM component sees only control plane (SDP, ICE) +- Media encryption (DTLS/SRTP) handled by browser + +--- + +## Testing Requirements + +### Browser Compatibility +Test on: +- Chrome/Chromium (primary target) +- Firefox +- Safari (macOS/iOS) +- Edge + +### Functionality Tests +1. Microphone permission flow +2. Peer connection establishment +3. Audio transmission (can hear remote peer) +4. Mute/unmute functionality +5. WebSocket reconnection +6. Multiple simultaneous connections +7. Network interruption handling + +### Performance Tests +- Latency measurements (mouth-to-ear delay) +- Audio quality under packet loss +- CPU usage in WASM component +- Memory usage with N peers + +--- + +## Integration with Existing Scroll + +Once platform features are ready, the mediasoup scroll can be updated: + +### Backend (already complete) +- ✅ mediasoup server.js +- ✅ WebSocket signaling +- ✅ SFU multi-party logic + +### Frontend (needs platform APIs) +- ⏳ Rewrite `.scroll/private/` using actual DruidUI components +- ⏳ Use WIT interfaces instead of plain browser APIs +- ⏳ WASM component for voice UI + +### Example DruidUI Component (pseudo-code) +```typescript +// voice-client.component.ts (compiled to WASM) +import { getUserMedia, createPeerConnection } from 'druid:webrtc'; + +export function init() { + // Called when component loads +} + +export function render(): string { + // Return component HTML + return `
...
`; +} + +export function onUserAction(action: string, data: string) { + if (action === 'join') { + const stream = getUserMedia({ audio: { echoCancellation: true } }); + const pc = createPeerConnection({ iceServers: [...] }); + pc.addStream(stream); + // ... + } +} + +export function handleWebrtcEvent(id: number, event: WebRTCEvent) { + if (event.kind === 'ice-candidate') { + // Send to signaling server via WebSocket + } +} +``` + +--- + +## Questions for Platform Team + +1. **Does DruidUI already expose any browser APIs to WASM?** + - If yes: which ones? Can we extend the pattern? + - If no: what's the preferred architecture for host-WASM communication? + +2. **Resource limits for WASM components?** + - Max number of peer connections per component? + - Memory limits? + - CPU time limits? + +3. **Event handling model?** + - Synchronous or async callbacks? + - Event batching/throttling? + - Priority system for real-time events? + +4. **Multi-component coordination?** + - Can two DruidUI components share a peer connection? + - How do components communicate with the server backend? + +--- + +## Timeline Estimate + +**Assuming 1 developer working full-time:** + +- **Phase 1 (MVP):** 2-3 weeks + - WIT implementation in platform: 1 week + - WASM bindings generation: 2-3 days + - Testing/debugging: 1 week + +- **Phase 2 (Audio control):** 1 week + - Additional APIs: 2-3 days + - UI components: 2-3 days + - Integration testing: 1-2 days + +- **Phase 3 (Advanced):** 2+ weeks + - Feature-dependent + +**Total for production-ready voice chat:** 4-6 weeks + +--- + +## Related Work + +### Existing WebRTC WASM projects to study: +- **webrtc-rs** (Rust WebRTC implementation) +- **Jitsi Meet** (WebRTC video conferencing) +- **mediasoup-client** (TypeScript WebRTC client) + +### Component Model resources: +- https://component-model.bytecodealliance.org/ +- https://github.com/WebAssembly/component-model + +--- + +**Author:** Lugh (Druid Bot) +**Date:** 2026-02-11 +**For:** mediasoup voice server scroll PR #13 diff --git a/scrolls/voice/mediasoup/latest/README.md b/scrolls/voice/mediasoup/latest/README.md new file mode 100644 index 00000000..9429954e --- /dev/null +++ b/scrolls/voice/mediasoup/latest/README.md @@ -0,0 +1,128 @@ +# Druid Voice Server - mediasoup Scroll + +Browser-based voice chat server powered by mediasoup WebRTC SFU. + +## Features + +- ✅ **Browser-Based** - No client downloads required +- ✅ **Low Latency** - WebRTC for real-time voice +- ✅ **Self-Hosted** - Full control over your infrastructure +- ✅ **ColdStarter Ready** - Auto-wake on connection +- ✅ **DruidUI** - Native web management interface +- ✅ **Pay-Per-Second** - Druid's usage-based billing + +## Perfect For + +- Gaming communities wanting Discord alternatives +- Privacy-conscious groups +- Teams needing self-hosted voice +- Communities affected by Discord's photo ID requirements + +## Technical Stack + +- **Backend:** mediasoup (WebRTC SFU) +- **Runtime:** Node.js 22 +- **Frontend:** DruidUI (WASM) +- **Protocol:** WebRTC over UDP + TCP + +## Ports + +- **HTTP/WS:** TCP port for API and WebSocket signaling +- **WebRTC:** UDP port range for media streams + - Auto-wakes on STUN/DTLS packets + - ColdStarter detects ICE connectivity checks + +## How It Works + +### ColdStarter Integration + +The scroll includes a custom Lua packet handler (`webrtc.lua`) that detects: +- **STUN packets** - ICE connectivity checks +- **DTLS packets** - WebRTC handshakes + +When a user attempts to join, ColdStarter automatically wakes the server. + +### Architecture + +``` +User Browser (WebRTC) + ↓ +WebSocket Signaling (port HTTP) + ↓ +mediasoup SFU (server.js) + ↓ +UDP Media Streams (ports WEBRTC-MIN to WEBRTC-MAX) +``` + +## Usage + +1. Deploy the scroll via Druid +2. Access DruidUI at `https://your-server:port` +3. Click "Join Voice Channel" to connect +4. Browser will request microphone permissions +5. Start talking! + +## Environment Variables + +Set in Druid deployment: + +- `ANNOUNCED_IP` - Public IP for WebRTC (optional, auto-detected) + +## Development + +### Local Testing + +```bash +cd /app/resources/deployment +yarn install +node server.js +``` + +### Files + +- `scroll.yaml` - Scroll configuration +- `server.js` - mediasoup backend +- `install.sh` - Dependency installer +- `packet_handler/webrtc.lua` - ColdStarter wake logic +- `.scroll/private/` - DruidUI interface + +## Cost Efficiency + +With Druid's pay-per-second model: + +- **Idle:** €0/month (server sleeps) +- **Active (1h/day):** ~€1-2/month +- **Active (24/7):** ~€10-15/month + +Compare to Discord Nitro: €10/month with no control. + +## Privacy Benefits + +- ✅ Self-hosted (you own the data) +- ✅ No photo ID requirements +- ✅ No corporate surveillance +- ✅ E2E encryption capable (DTLS) +- ✅ Full GDPR compliance + +## Roadmap + +- [ ] Screen sharing support +- [ ] Multiple voice channels +- [ ] User permissions system +- [ ] Recording capabilities +- [ ] Mobile browser optimization +- [ ] Integration with Matrix/Element + +## Related Scrolls + +- Discord alternatives: Matrix, Rocket.Chat (coming soon) +- Game servers: Minecraft, Rust, etc. + +## License + +Part of the Druid scroll collection. + +--- + +**Built by:** druid.gg team +**Support:** https://discord.gg/druid diff --git a/scrolls/voice/mediasoup/latest/druid-ui-webrtc.wit b/scrolls/voice/mediasoup/latest/druid-ui-webrtc.wit new file mode 100644 index 00000000..89107c2d --- /dev/null +++ b/scrolls/voice/mediasoup/latest/druid-ui-webrtc.wit @@ -0,0 +1,309 @@ +// WIT Interface Definition for DruidUI WebRTC Support +// WebAssembly Component Model interface for real-time voice communication + +package druid:webrtc@0.1.0; + +/// Core WebRTC interfaces for voice chat in DruidUI +interface webrtc { + + // ============================================================================ + // Media Device Access + // ============================================================================ + + /// Media stream constraints for getUserMedia + record media-constraints { + audio: option, + video: option, + } + + record audio-constraints { + sample-rate: option, // e.g., 48000 + channel-count: option, // 1 = mono, 2 = stereo + echo-cancellation: option, + noise-suppression: option, + auto-gain-control: option, + device-id: option, + } + + record video-constraints { + width: option, + height: option, + frame-rate: option, + device-id: option, + } + + /// Media device information + record media-device-info { + device-id: string, + label: string, + kind: device-kind, + } + + enum device-kind { + audio-input, + audio-output, + video-input, + } + + /// Handle to a media stream (audio/video) + resource media-stream { + /// Get tracks from this stream + get-audio-tracks: func() -> list; + get-video-tracks: func() -> list; + + /// Add/remove tracks + add-track: func(track: track-handle); + remove-track: func(track: track-handle); + + /// Stream state + is-active: func() -> bool; + + /// Clean up + stop: func(); + } + + type audio-track-handle = u32; + type video-track-handle = u32; + type track-handle = u32; + + /// Request user media (microphone/camera) + get-user-media: func(constraints: media-constraints) -> result; + + /// Enumerate available media devices + enumerate-devices: func() -> result, media-error>; + + enum media-error { + permission-denied, + device-not-found, + constraint-not-satisfied, + not-supported, + unknown, + } + + + // ============================================================================ + // RTCPeerConnection + // ============================================================================ + + /// WebRTC peer connection for sending/receiving media + resource rtc-peer-connection { + /// Add a media stream to send + add-stream: func(stream: media-stream) -> result<_, rtc-error>; + + /// Create an offer (caller side) + create-offer: func() -> result; + + /// Create an answer (callee side) + create-answer: func() -> result; + + /// Set local description + set-local-description: func(desc: session-description) -> result<_, rtc-error>; + + /// Set remote description + set-remote-description: func(desc: session-description) -> result<_, rtc-error>; + + /// Add ICE candidate + add-ice-candidate: func(candidate: ice-candidate) -> result<_, rtc-error>; + + /// Get connection state + get-connection-state: func() -> connection-state; + + /// Get ICE connection state + get-ice-connection-state: func() -> ice-connection-state; + + /// Close the connection + close: func(); + } + + record rtc-configuration { + ice-servers: list, + } + + record ice-server { + urls: list, // STUN/TURN server URLs + username: option, + credential: option, + } + + record session-description { + sdp-type: sdp-type, + sdp: string, + } + + enum sdp-type { + offer, + answer, + pranswer, + rollback, + } + + record ice-candidate { + candidate: string, + sdp-mid: option, + sdp-m-line-index: option, + } + + enum connection-state { + new, + connecting, + connected, + disconnected, + failed, + closed, + } + + enum ice-connection-state { + new, + checking, + connected, + completed, + failed, + disconnected, + closed, + } + + enum rtc-error { + invalid-state, + invalid-parameter, + operation-error, + unknown, + } + + /// Create a new peer connection + create-peer-connection: func(config: rtc-configuration) -> result; + + + // ============================================================================ + // WebRTC Events (callbacks from host to WASM) + // ============================================================================ + + /// Event types for WebRTC + variant webrtc-event { + /// ICE candidate generated + ice-candidate(ice-candidate-event), + + /// Connection state changed + connection-state-change(connection-state), + + /// ICE connection state changed + ice-connection-state-change(ice-connection-state), + + /// Remote stream added + remote-stream-added(media-stream), + + /// Remote stream removed + remote-stream-removed(stream-id), + + /// Track added to connection + track-added(track-event), + } + + record ice-candidate-event { + candidate: option, + } + + record track-event { + track: track-handle, + streams: list, + } + + type stream-id = string; + + /// Register callback for WebRTC events (called by WASM component) + /// Host will invoke the exported `handle-webrtc-event` function + register-event-handler: func(connection-id: u32); + + + // ============================================================================ + // Audio Control + // ============================================================================ + + /// Audio track controls + resource audio-track { + /// Mute/unmute + set-enabled: func(enabled: bool); + is-enabled: func() -> bool; + + /// Volume control (0.0 to 1.0) + set-volume: func(volume: f32); + get-volume: func() -> f32; + + /// Get current audio level (0.0 to 1.0) + get-audio-level: func() -> f32; + } + + /// Get audio track from handle + get-audio-track: func(handle: audio-track-handle) -> result; + + + // ============================================================================ + // WebSocket Signaling (for WebRTC signaling channel) + // ============================================================================ + + resource websocket { + /// Send text message + send-text: func(data: string) -> result<_, ws-error>; + + /// Send binary message + send-binary: func(data: list) -> result<_, ws-error>; + + /// Close connection + close: func(code: option, reason: option); + + /// Get ready state + get-ready-state: func() -> ws-ready-state; + } + + enum ws-ready-state { + connecting, + open, + closing, + closed, + } + + enum ws-error { + invalid-state, + send-failed, + unknown, + } + + /// WebSocket events + variant ws-event { + open, + message-text(string), + message-binary(list), + error(string), + close(ws-close-event), + } + + record ws-close-event { + code: u16, + reason: string, + was-clean: bool, + } + + /// Connect to WebSocket server + websocket-connect: func(url: string) -> result; + + /// Register WebSocket event handler + register-ws-event-handler: func(ws-id: u32); +} + + +// ============================================================================ +// World Definition - Export interface for DruidUI components +// ============================================================================ + +world druid-voice-component { + /// Import WebRTC capabilities from host + import webrtc; + + /// Export event handlers (called by host) + export handle-webrtc-event: func(connection-id: u32, event: webrtc.webrtc-event); + export handle-ws-event: func(ws-id: u32, event: webrtc.ws-event); + + /// Export component lifecycle + export init: func(); + export render: func() -> string; // Returns HTML/virtual DOM + export on-user-action: func(action: string, data: string); +} diff --git a/scrolls/voice/mediasoup/latest/install.sh b/scrolls/voice/mediasoup/latest/install.sh new file mode 100644 index 00000000..91145af8 --- /dev/null +++ b/scrolls/voice/mediasoup/latest/install.sh @@ -0,0 +1,26 @@ +#!/bin/bash +set -e + +cd /app/resources/deployment + +# Initialize package.json if it doesn't exist +if [ ! -f "package.json" ]; then + cat > package.json << 'EOF' +{ + "name": "druid-voice-mediasoup", + "version": "1.0.0", + "description": "Druid Voice Server powered by mediasoup", + "main": "server.js", + "dependencies": { + "mediasoup": "^3.14.0", + "express": "^4.18.2", + "ws": "^8.16.0" + } +} +EOF +fi + +# Install dependencies +yarn install + +echo "✅ mediasoup voice server installed successfully" diff --git a/scrolls/voice/mediasoup/latest/packet_handler/webrtc.lua b/scrolls/voice/mediasoup/latest/packet_handler/webrtc.lua new file mode 100644 index 00000000..66a022c2 --- /dev/null +++ b/scrolls/voice/mediasoup/latest/packet_handler/webrtc.lua @@ -0,0 +1,62 @@ +-- WebRTC / STUN packet handler for Druid Voice Server +-- Detects STUN binding requests and ICE candidates + +function is_stun_packet(data) + if #data < 20 then + return false + end + + -- STUN packets start with 0x00 or 0x01 (message type) + -- Check for STUN magic cookie: 0x2112A442 + local b1, b2, b3, b4 = string.byte(data, 5, 8) + + if b1 == 0x21 and b2 == 0x12 and b3 == 0xA4 and b4 == 0x42 then + return true + end + + return false +end + +function is_dtls_packet(data) + if #data < 13 then + return false + end + + -- DTLS content types: 20-26 + local content_type = string.byte(data, 1) + + if content_type >= 20 and content_type <= 26 then + -- Check DTLS version (major.minor) + local version = string.byte(data, 2) * 256 + string.byte(data, 3) + -- DTLS 1.0: 0xFEFF, DTLS 1.2: 0xFEFD + if version == 0xFEFF or version == 0xFEFD then + return true + end + end + + return false +end + +-- Main handler function called by ColdStarter +function handle(data, info) + -- Check if this is a STUN packet (ICE connectivity) + if is_stun_packet(data) then + return { + wake = true, + reason = "STUN/ICE connection attempt" + } + end + + -- Check if this is a DTLS packet (WebRTC handshake) + if is_dtls_packet(data) then + return { + wake = true, + reason = "DTLS/WebRTC handshake" + } + end + + -- Unknown packet - don't wake + return { + wake = false + } +end diff --git a/scrolls/voice/mediasoup/latest/scroll.yaml b/scrolls/voice/mediasoup/latest/scroll.yaml new file mode 100644 index 00000000..b6c31076 --- /dev/null +++ b/scrolls/voice/mediasoup/latest/scroll.yaml @@ -0,0 +1,46 @@ +name: artifacts.druid.gg/druid-team/scroll-voice-mediasoup +desc: Voice Server powered by mediasoup (WebRTC SFU) +version: 0.0.1 +app_version: latest +ports: + - name: http + protocol: tcp + description: HTTP API and WebSocket signaling + sleep_handler: generic + - name: webrtc-min + protocol: udp + description: WebRTC media (min range) + sleep_handler: packet_handler/webrtc.lua + vars: + - name: ServerName + value: "Druid Voice Server (idle) - Join to wake" + - name: MaxUsers + value: "50" + - name: webrtc-max + protocol: udp + description: WebRTC media (max range) +init: "start" +dependencies: + - nodejs_22 + - yarn +commands: + start: + needs: [install] + run: restart + procedures: + - mode: exec + data: + - node + - server.js + stop: + type: stop + procedures: + - mode: signal + data: SIGTERM + install: + run: once + procedures: + - mode: exec + data: + - sh + - install.sh diff --git a/scrolls/voice/mediasoup/latest/server.js b/scrolls/voice/mediasoup/latest/server.js new file mode 100644 index 00000000..24d053f9 --- /dev/null +++ b/scrolls/voice/mediasoup/latest/server.js @@ -0,0 +1,190 @@ +#!/usr/bin/env node +/** + * Druid Voice Server - mediasoup WebRTC SFU + * Browser-based voice chat for gaming communities + */ + +const express = require('express'); +const http = require('http'); +const WebSocket = require('ws'); +const mediasoup = require('mediasoup'); + +const HTTP_PORT = process.env.PORT_HTTP || 3000; +const WS_PORT = process.env.PORT_HTTP || 3000; +const WEBRTC_MIN_PORT = parseInt(process.env.PORT_WEBRTC_MIN) || 40000; +const WEBRTC_MAX_PORT = parseInt(process.env.PORT_WEBRTC_MAX) || 40100; + +const app = express(); +const server = http.createServer(app); +const wss = new WebSocket.Server({ server }); + +let worker; +let router; +const transports = new Map(); +const producers = new Map(); +const consumers = new Map(); + +// mediasoup worker settings +const mediaCodecs = [ + { + kind: 'audio', + mimeType: 'audio/opus', + clockRate: 48000, + channels: 2 + } +]; + +async function createWorker() { + worker = await mediasoup.createWorker({ + logLevel: 'warn', + rtcMinPort: WEBRTC_MIN_PORT, + rtcMaxPort: WEBRTC_MAX_PORT, + }); + + console.log(`✅ mediasoup worker created [pid:${worker.pid}]`); + + worker.on('died', () => { + console.error('❌ mediasoup worker died, exiting...'); + process.exit(1); + }); + + router = await worker.createRouter({ mediaCodecs }); + console.log('✅ mediasoup router created'); +} + +// WebSocket signaling +wss.on('connection', (ws) => { + console.log('🔌 Client connected'); + + ws.on('message', async (message) => { + try { + const data = JSON.parse(message); + + switch (data.type) { + case 'getRouterRtpCapabilities': + ws.send(JSON.stringify({ + type: 'routerRtpCapabilities', + data: router.rtpCapabilities + })); + break; + + case 'createWebRtcTransport': + const transport = await createWebRtcTransport(); + transports.set(transport.id, transport); + ws.send(JSON.stringify({ + type: 'transportCreated', + data: { + id: transport.id, + iceParameters: transport.iceParameters, + iceCandidates: transport.iceCandidates, + dtlsParameters: transport.dtlsParameters, + } + })); + break; + + case 'connectTransport': + const t = transports.get(data.transportId); + await t.connect({ dtlsParameters: data.dtlsParameters }); + ws.send(JSON.stringify({ type: 'transportConnected' })); + break; + + case 'produce': + const producer = await transports.get(data.transportId).produce({ + kind: data.kind, + rtpParameters: data.rtpParameters, + }); + producers.set(producer.id, producer); + ws.send(JSON.stringify({ + type: 'produced', + data: { id: producer.id } + })); + break; + + case 'consume': + const consumer = await transports.get(data.transportId).consume({ + producerId: data.producerId, + rtpCapabilities: data.rtpCapabilities, + paused: true, + }); + consumers.set(consumer.id, consumer); + ws.send(JSON.stringify({ + type: 'consumed', + data: { + id: consumer.id, + producerId: data.producerId, + kind: consumer.kind, + rtpParameters: consumer.rtpParameters, + } + })); + break; + } + } catch (err) { + console.error('❌ Error handling message:', err); + ws.send(JSON.stringify({ type: 'error', error: err.message })); + } + }); + + ws.on('close', () => { + console.log('🔌 Client disconnected'); + }); +}); + +async function createWebRtcTransport() { + const transport = await router.createWebRtcTransport({ + listenIps: [{ ip: '0.0.0.0', announcedIp: process.env.ANNOUNCED_IP || null }], + enableUdp: true, + enableTcp: true, + preferUdp: true, + }); + + return transport; +} + +// HTTP API +app.get('/health', (req, res) => { + res.json({ + status: 'ok', + worker: worker?.pid, + transports: transports.size, + producers: producers.size, + consumers: consumers.size + }); +}); + +app.get('/', (req, res) => { + res.send(` + + Druid Voice Server + +

🎙️ Druid Voice Server

+

Powered by mediasoup WebRTC SFU

+

Status: Running

+

Connect via DruidUI to join voice channels

+ + + `); +}); + +// Start server +async function main() { + await createWorker(); + + server.listen(HTTP_PORT, () => { + console.log(`🎙️ Druid Voice Server running`); + console.log(`📡 HTTP/WS: http://0.0.0.0:${HTTP_PORT}`); + console.log(`🔊 WebRTC ports: ${WEBRTC_MIN_PORT}-${WEBRTC_MAX_PORT}`); + }); +} + +// Graceful shutdown +process.on('SIGTERM', async () => { + console.log('📴 Shutting down gracefully...'); + server.close(); + worker?.close(); + process.exit(0); +}); + +main().catch(err => { + console.error('❌ Failed to start server:', err); + process.exit(1); +});