From 528c11a1dc5de3bbe97f1ff218ec0662f179e9e1 Mon Sep 17 00:00:00 2001 From: Bobby Christopher Date: Sun, 21 Dec 2025 11:28:16 -0500 Subject: [PATCH 01/20] made webrtc route helpers --- .../integration-suite/src/generated/app.ts | 59 --- apps/testing/webrtc-test/.gitignore | 43 ++ .../testing/webrtc-test/.vscode/settings.json | 16 + apps/testing/webrtc-test/AGENTS.md | 64 +++ apps/testing/webrtc-test/README.md | 147 ++++++ apps/testing/webrtc-test/agentuity.config.ts | 35 ++ apps/testing/webrtc-test/agentuity.json | 13 + apps/testing/webrtc-test/app.ts | 17 + apps/testing/webrtc-test/package.json | 30 ++ apps/testing/webrtc-test/src/agent/AGENTS.md | 322 +++++++++++++ .../webrtc-test/src/agent/hello/agent.ts | 15 + .../webrtc-test/src/agent/hello/index.ts | 1 + apps/testing/webrtc-test/src/api/AGENTS.md | 259 ++++++++++ apps/testing/webrtc-test/src/api/index.ts | 16 + apps/testing/webrtc-test/src/generated/app.ts | 316 ++++++++++++ apps/testing/webrtc-test/src/web/AGENTS.md | 282 +++++++++++ apps/testing/webrtc-test/src/web/App.tsx | 344 ++++++++++++++ apps/testing/webrtc-test/src/web/frontend.tsx | 29 ++ apps/testing/webrtc-test/src/web/index.html | 13 + .../webrtc-test/src/web/public/.gitkeep | 0 .../webrtc-test/src/web/public/favicon.ico | Bin 0 -> 174912 bytes apps/testing/webrtc-test/tsconfig.json | 27 ++ bun.lock | 51 +- packages/cli/src/cmd/build/ast.ts | 11 + packages/cli/src/cmd/dev/index.ts | 122 ++--- packages/frontend/src/index.ts | 7 + packages/frontend/src/webrtc-manager.ts | 448 ++++++++++++++++++ packages/react/src/index.ts | 10 + packages/react/src/webrtc.tsx | 213 +++++++++ packages/runtime/src/index.ts | 9 + packages/runtime/src/router.ts | 60 +++ packages/runtime/src/webrtc-signaling.ts | 226 +++++++++ .../runtime/test/webrtc-signaling.test.ts | 324 +++++++++++++ 33 files changed, 3385 insertions(+), 144 deletions(-) create mode 100644 apps/testing/webrtc-test/.gitignore create mode 100644 apps/testing/webrtc-test/.vscode/settings.json create mode 100644 apps/testing/webrtc-test/AGENTS.md create mode 100644 apps/testing/webrtc-test/README.md create mode 100644 apps/testing/webrtc-test/agentuity.config.ts create mode 100644 apps/testing/webrtc-test/agentuity.json create mode 100644 apps/testing/webrtc-test/app.ts create mode 100644 apps/testing/webrtc-test/package.json create mode 100644 apps/testing/webrtc-test/src/agent/AGENTS.md create mode 100644 apps/testing/webrtc-test/src/agent/hello/agent.ts create mode 100644 apps/testing/webrtc-test/src/agent/hello/index.ts create mode 100644 apps/testing/webrtc-test/src/api/AGENTS.md create mode 100644 apps/testing/webrtc-test/src/api/index.ts create mode 100644 apps/testing/webrtc-test/src/generated/app.ts create mode 100644 apps/testing/webrtc-test/src/web/AGENTS.md create mode 100644 apps/testing/webrtc-test/src/web/App.tsx create mode 100644 apps/testing/webrtc-test/src/web/frontend.tsx create mode 100644 apps/testing/webrtc-test/src/web/index.html create mode 100644 apps/testing/webrtc-test/src/web/public/.gitkeep create mode 100644 apps/testing/webrtc-test/src/web/public/favicon.ico create mode 100644 apps/testing/webrtc-test/tsconfig.json create mode 100644 packages/frontend/src/webrtc-manager.ts create mode 100644 packages/react/src/webrtc.tsx create mode 100644 packages/runtime/src/webrtc-signaling.ts create mode 100644 packages/runtime/test/webrtc-signaling.test.ts diff --git a/apps/testing/integration-suite/src/generated/app.ts b/apps/testing/integration-suite/src/generated/app.ts index 169ee9b5b..1148481b3 100644 --- a/apps/testing/integration-suite/src/generated/app.ts +++ b/apps/testing/integration-suite/src/generated/app.ts @@ -113,65 +113,6 @@ if (!isDevelopment()) { app.get('/_idle', idleHandler); } -// Asset proxy routes - Development mode only (proxies to Vite asset server) -if (process.env.NODE_ENV !== 'production') { - const VITE_ASSET_PORT = parseInt(process.env.VITE_PORT || '5173', 10); - - const proxyToVite = async (c: Context) => { - const viteUrl = `http://127.0.0.1:${VITE_ASSET_PORT}${c.req.path}`; - const controller = new AbortController(); - const timeout = setTimeout(() => controller.abort(), 10000); // 10s timeout - try { - otel.logger.debug(`[Proxy] ${c.req.method} ${c.req.path} -> Vite:${VITE_ASSET_PORT}`); - const res = await fetch(viteUrl, { signal: controller.signal }); - clearTimeout(timeout); - otel.logger.debug(`[Proxy] ${c.req.path} -> ${res.status} (${res.headers.get('content-type')})`); - return new Response(res.body, { - status: res.status, - headers: res.headers, - }); - } catch (err) { - clearTimeout(timeout); - if (err instanceof Error && err.name === 'AbortError') { - otel.logger.error(`Vite proxy timeout: ${c.req.path}`); - return c.text('Vite asset server timeout', 504); - } - otel.logger.error(`Failed to proxy to Vite: ${c.req.path} - ${err instanceof Error ? err.message : String(err)}`); - return c.text('Vite asset server error', 500); - } - }; - - // Vite client scripts and HMR - app.get('/@vite/*', proxyToVite); - app.get('/@react-refresh', proxyToVite); - - // Source files for HMR - app.get('/src/web/*', proxyToVite); - app.get('/src/*', proxyToVite); // Catch-all for other source files - - // Workbench source files (in .agentuity/workbench-src/) - app.get('/.agentuity/workbench-src/*', proxyToVite); - - // Node modules (Vite transforms these) - app.get('/node_modules/*', proxyToVite); - - // Scoped packages (e.g., @agentuity/*, @types/*) - app.get('/@*', proxyToVite); - - // File system access (for Vite's @fs protocol) - app.get('/@fs/*', proxyToVite); - - // Module resolution (for Vite's @id protocol) - app.get('/@id/*', proxyToVite); - - // Any .js, .jsx, .ts, .tsx files (catch remaining modules) - app.get('/*.js', proxyToVite); - app.get('/*.jsx', proxyToVite); - app.get('/*.ts', proxyToVite); - app.get('/*.tsx', proxyToVite); - app.get('/*.css', proxyToVite); -} - // Mount API routes const { default: router_0 } = await import('../api/index.js'); app.route('/api', router_0); diff --git a/apps/testing/webrtc-test/.gitignore b/apps/testing/webrtc-test/.gitignore new file mode 100644 index 000000000..6767817a9 --- /dev/null +++ b/apps/testing/webrtc-test/.gitignore @@ -0,0 +1,43 @@ +# dependencies (bun install) + +node_modules + +# output + +out +dist +*.tgz + +# code coverage + +coverage +*.lcov + +# logs + +/logs +_.log +report.[0-9]_.[0-9]_.[0-9]_.[0-9]\*.json + +# dotenv environment variable files + +.env +.env.\* + +# caches + +.eslintcache +.cache +*.tsbuildinfo + +# IntelliJ based IDEs + +.idea + +# Finder (MacOS) folder config + +.DS_Store + +# Agentuity build files + +.agentuity diff --git a/apps/testing/webrtc-test/.vscode/settings.json b/apps/testing/webrtc-test/.vscode/settings.json new file mode 100644 index 000000000..8b2c0232a --- /dev/null +++ b/apps/testing/webrtc-test/.vscode/settings.json @@ -0,0 +1,16 @@ +{ + "search.exclude": { + "**/.git/**": true, + "**/node_modules/**": true, + "**/bun.lock": true, + "**/.agentuity/**": true + }, + "json.schemas": [ + { + "fileMatch": [ + "agentuity.json" + ], + "url": "https://agentuity.dev/schema/cli/v1/agentuity.json" + } + ] +} \ No newline at end of file diff --git a/apps/testing/webrtc-test/AGENTS.md b/apps/testing/webrtc-test/AGENTS.md new file mode 100644 index 000000000..20cd550e1 --- /dev/null +++ b/apps/testing/webrtc-test/AGENTS.md @@ -0,0 +1,64 @@ +# Agent Guidelines for webrtc-test + +## Commands + +- **Build**: `bun run build` (compiles your application) +- **Dev**: `bun run dev` (starts development server) +- **Typecheck**: `bun run typecheck` (runs TypeScript type checking) +- **Deploy**: `bun run deploy` (deploys your app to the Agentuity cloud) + +## Agent-Friendly CLI + +The Agentuity CLI is designed to be agent-friendly with programmatic interfaces, structured output, and comprehensive introspection. + +Read the [AGENTS.md](./node_modules/@agentuity/cli/AGENTS.md) file in the Agentuity CLI for more information on how to work with this project. + +## Instructions + +- This project uses Bun instead of NodeJS and TypeScript for all source code +- This is an Agentuity Agent project + +## Web Frontend (src/web/) + +The `src/web/` folder contains your React frontend, which is automatically bundled by the Agentuity build system. + +**File Structure:** + +- `index.html` - Main HTML file with ` + + +``` + +## React Hooks + +### useAgent - Call Agents + +```typescript +import { useAgent } from '@agentuity/react'; + +function MyComponent() { + const { run, running, data, error } = useAgent('myAgent'); + + return ( + + ); +} +``` + +### useAgentWebsocket - WebSocket Connection + +```typescript +import { useAgentWebsocket } from '@agentuity/react'; + +function MyComponent() { + const { connected, send, data } = useAgentWebsocket('websocket'); + + return ( +
+

Status: {connected ? 'Connected' : 'Disconnected'}

+ +

Received: {data}

+
+ ); +} +``` + +### useAgentEventStream - Server-Sent Events + +```typescript +import { useAgentEventStream } from '@agentuity/react'; + +function MyComponent() { + const { connected, data, error } = useAgentEventStream('sse'); + + return ( +
+

Connected: {connected ? 'Yes' : 'No'}

+ {error &&

Error: {error.message}

} +

Data: {data}

+
+ ); +} +``` + +## Complete Example + +```typescript +import { AgentuityProvider, useAgent, useAgentWebsocket } from '@agentuity/react'; +import { useEffect, useState } from 'react'; + +export function App() { + const [count, setCount] = useState(0); + const { run, data: agentResult } = useAgent('simple'); + const { connected, send, data: wsMessage } = useAgentWebsocket('websocket'); + + useEffect(() => { + // Send WebSocket message every second + const interval = setInterval(() => { + send(`Message at ${new Date().toISOString()}`); + }, 1000); + return () => clearInterval(interval); + }, [send]); + + return ( +
+ +

My Agentuity App

+ +
+

Count: {count}

+ +
+ +
+ +

{agentResult}

+
+ +
+ WebSocket: + {connected ? JSON.stringify(wsMessage) : 'Not connected'} +
+
+
+ ); +} +``` + +## Static Assets + +Place static files in the **public/** folder: + +``` +src/web/public/ +├── logo.svg +├── styles.css +└── script.js +``` + +Reference them in your HTML or components: + +```html + + + +``` + +```typescript +// In React components +Logo +``` + +## Styling + +### Inline Styles + +```typescript +
+ Styled content +
+``` + +### CSS Files + +Create `public/styles.css`: + +```css +body { + background-color: #09090b; + color: #fff; + font-family: sans-serif; +} +``` + +Import in `index.html`: + +```html + +``` + +### Style Tag in Component + +```typescript +
+ + +
+``` + +## Best Practices + +- Wrap your app with **AgentuityProvider** for hooks to work +- Use **useAgent** for one-off agent calls +- Use **useAgentWebsocket** for bidirectional real-time communication +- Use **useAgentEventStream** for server-to-client streaming +- Place reusable components in separate files +- Keep static assets in the **public/** folder +- Use TypeScript for type safety +- Handle loading and error states in UI + +## Rules + +- **App.tsx** must export a function named `App` +- **frontend.tsx** must render the `App` component to `#root` +- **index.html** must have a `
` +- All agents are accessible via `useAgent('agentName')` +- The web app is served at `/` by default +- Static files in `public/` are served at `/public/*` +- Module script tag: `` diff --git a/apps/testing/webrtc-test/src/web/App.tsx b/apps/testing/webrtc-test/src/web/App.tsx new file mode 100644 index 000000000..973a3a906 --- /dev/null +++ b/apps/testing/webrtc-test/src/web/App.tsx @@ -0,0 +1,344 @@ +import { useWebRTCCall } from '@agentuity/react'; +import { useState, useEffect } from 'react'; + +export function App() { + const [roomId, setRoomId] = useState('test-room'); + const [joined, setJoined] = useState(false); + + const { + localVideoRef, + remoteVideoRef, + status, + error, + peerId, + remotePeerId, + isAudioMuted, + isVideoMuted, + connect, + hangup, + muteAudio, + muteVideo, + } = useWebRTCCall({ + roomId, + signalUrl: '/api/call/signal', + autoConnect: false, + }); + + // Auto-attach streams to video elements when refs are ready + useEffect(() => { + if (localVideoRef.current) { + localVideoRef.current.muted = true; + localVideoRef.current.playsInline = true; + } + if (remoteVideoRef.current) { + remoteVideoRef.current.playsInline = true; + } + }, [localVideoRef, remoteVideoRef]); + + const handleJoin = () => { + setJoined(true); + connect(); + }; + + const handleLeave = () => { + hangup(); + setJoined(false); + }; + + return ( +
+
+

WebRTC Video Call Demo

+

Powered by Agentuity

+
+ + {!joined ? ( +
+

Join a Room

+
+ + setRoomId(e.target.value)} + placeholder="Enter room ID" + /> +
+ +

Open this page in two browser tabs to test

+
+ ) : ( +
+
+ {status} + {peerId && You: {peerId}} + {remotePeerId && Remote: {remotePeerId}} +
+ + {error &&
Error: {error.message}
} + +
+
+
+
+
+
+ +
+ + + +
+
+ )} + + +
+ ); +} diff --git a/apps/testing/webrtc-test/src/web/frontend.tsx b/apps/testing/webrtc-test/src/web/frontend.tsx new file mode 100644 index 000000000..969967816 --- /dev/null +++ b/apps/testing/webrtc-test/src/web/frontend.tsx @@ -0,0 +1,29 @@ +/** + * This file is the entry point for the React app, it sets up the root + * element and renders the App component to the DOM. + * + * It is included in `src/index.html`. + */ + +import React, { StrictMode } from 'react'; +import { createRoot } from 'react-dom/client'; +import { AgentuityProvider } from '@agentuity/react'; +import { App } from './App'; + +const elem = document.getElementById('root')!; +const app = ( + + + + + +); + +if (import.meta.hot) { + // With hot module reloading, `import.meta.hot.data` is persisted. + const root = (import.meta.hot.data.root ??= createRoot(elem)); + root.render(app); +} else { + // The hot module reloading API is not available in production. + createRoot(elem).render(app); +} diff --git a/apps/testing/webrtc-test/src/web/index.html b/apps/testing/webrtc-test/src/web/index.html new file mode 100644 index 000000000..781191e6d --- /dev/null +++ b/apps/testing/webrtc-test/src/web/index.html @@ -0,0 +1,13 @@ + + + + + + + Agentuity + Bun + React + + + +
+ + diff --git a/apps/testing/webrtc-test/src/web/public/.gitkeep b/apps/testing/webrtc-test/src/web/public/.gitkeep new file mode 100644 index 000000000..e69de29bb diff --git a/apps/testing/webrtc-test/src/web/public/favicon.ico b/apps/testing/webrtc-test/src/web/public/favicon.ico new file mode 100644 index 0000000000000000000000000000000000000000..21f46e6f552734670df178e96f66592ad1741b37 GIT binary patch literal 174912 zcmeF42Y6IP*T-){fP|I+QlthcqVy^vkN_%*fLH)2LIi|J5d=hPvUDsok=}x+^d?nW zLNg#G2+|=0K|%`&HK8Q;J3p`Uxo=E`-DEdxp68!!_s*R;bLO-;Goxuvnw#d~p~mN_wt(3`{G~6b022Pr0MzdYx=X#YI>nUNrP*i zc}CNno&SF3-*t9%)%0=WG=1wxI*P5O+>;KldQ6o(c2+;Hr zB{V&6UgH@&M}hC2d(J4gapV8Jm%0=!s_8RlYI>(m|F(s?DikQ7=^uWm>0`$H_u16H zTQ^N#xKPsYAi}3zB=)7^JMEt~LLSI`draeWE|)jEFi-#M$5S=kLNq z<|QC!GRtot2zYLyG?U-V>zUO1dva}{ps9eec?}gLbO2@7@MJ`%KeQ19jiK?`nFMEdQ-9V`cDQO}~3r)A#Q;Wb(Fcnw}CM zzxVAk?j15D@o|S7-nGkkj_6$-nZj#Q;2Pv3CZp^-@2RAziU@b-@aYb%au#| zI`eyw&3pH1dgsoG-{a(D)Dzjbd9$X^nDMWUQMx9v*YV>ueZ>k*&zCRp@{#W&Mre9q 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zM2FAF!ate&lf7MImadO!72LA7N0Rau&pI48{e=*OVn|B9B+m|rZmF(Jan`_Ej(qYJ zQ%0ciOJAI)I> | undefined; + let previousMetadata: + | Awaited> + | undefined; let devmode: DevmodeResponse | undefined; let gravityBin: string | undefined; @@ -358,67 +360,67 @@ export const command = createCommand({ await tui.spinner({ message: 'Building dev bundle', callback: async () => { - const { generateEntryFile } = await import('../build/entry-generator'); - await generateEntryFile({ - rootDir, - projectId: project?.projectId ?? '', - deploymentId, - logger, - mode: 'dev', - }); - - // Bundle the app with LLM patches (dev mode = no minification) - const { installExternalsAndBuild } = await import('../build/vite/server-bundler'); - await installExternalsAndBuild({ - rootDir, - dev: true, // DevMode: no minification, inline sourcemaps - logger, - }); - - // Generate metadata file (needed for eval ID lookup at runtime) - const { discoverAgents } = await import('../build/vite/agent-discovery'); - const { discoverRoutes } = await import('../build/vite/route-discovery'); - const { generateMetadata, writeMetadataFile } = await import( - '../build/vite/metadata-generator' - ); - - const srcDir = join(rootDir, 'src'); - const agents = await discoverAgents( - srcDir, - project?.projectId ?? '', - deploymentId, - logger - ); - const { routes } = await discoverRoutes( - srcDir, - project?.projectId ?? '', - deploymentId, - logger - ); - - const metadata = await generateMetadata({ - rootDir, - projectId: project?.projectId ?? '', - orgId: project?.orgId ?? '', - deploymentId, - agents, - routes, - dev: true, - logger, - }); - - writeMetadataFile(rootDir, metadata, true, logger); + const { generateEntryFile } = await import('../build/entry-generator'); + await generateEntryFile({ + rootDir, + projectId: project?.projectId ?? '', + deploymentId, + logger, + mode: 'dev', + }); + + // Bundle the app with LLM patches (dev mode = no minification) + const { installExternalsAndBuild } = await import('../build/vite/server-bundler'); + await installExternalsAndBuild({ + rootDir, + dev: true, // DevMode: no minification, inline sourcemaps + logger, + }); + + // Generate metadata file (needed for eval ID lookup at runtime) + const { discoverAgents } = await import('../build/vite/agent-discovery'); + const { discoverRoutes } = await import('../build/vite/route-discovery'); + const { generateMetadata, writeMetadataFile } = await import( + '../build/vite/metadata-generator' + ); - // Sync metadata with backend (creates agents and evals in the database) - if (syncService && project?.projectId) { - await syncService.sync( - metadata, - previousMetadata, - project.projectId, - deploymentId + const srcDir = join(rootDir, 'src'); + const agents = await discoverAgents( + srcDir, + project?.projectId ?? '', + deploymentId, + logger ); - previousMetadata = metadata; - } + const { routes } = await discoverRoutes( + srcDir, + project?.projectId ?? '', + deploymentId, + logger + ); + + const metadata = await generateMetadata({ + rootDir, + projectId: project?.projectId ?? '', + orgId: project?.orgId ?? '', + deploymentId, + agents, + routes, + dev: true, + logger, + }); + + writeMetadataFile(rootDir, metadata, true, logger); + + // Sync metadata with backend (creates agents and evals in the database) + if (syncService && project?.projectId) { + await syncService.sync( + metadata, + previousMetadata, + project.projectId, + deploymentId + ); + previousMetadata = metadata; + } }, clearOnSuccess: true, }); diff --git a/packages/frontend/src/index.ts b/packages/frontend/src/index.ts index 48178564b..7a64a187d 100644 --- a/packages/frontend/src/index.ts +++ b/packages/frontend/src/index.ts @@ -18,6 +18,13 @@ export { type EventStreamManagerOptions, type EventStreamManagerState, } from './eventstream-manager'; +export { + WebRTCManager, + type WebRTCStatus, + type WebRTCCallbacks, + type WebRTCManagerOptions, + type WebRTCManagerState, +} from './webrtc-manager'; // Export client implementation (local to this package) export { createClient } from './client/index'; diff --git a/packages/frontend/src/webrtc-manager.ts b/packages/frontend/src/webrtc-manager.ts new file mode 100644 index 000000000..e5f28ddf5 --- /dev/null +++ b/packages/frontend/src/webrtc-manager.ts @@ -0,0 +1,448 @@ +/** + * WebRTC connection status + */ +export type WebRTCStatus = 'disconnected' | 'connecting' | 'signaling' | 'connected'; + +/** + * Signaling message types (must match server protocol) + */ +type SignalMsg = + | { t: 'join'; roomId: string } + | { t: 'joined'; peerId: string; roomId: string; peers: string[] } + | { t: 'peer-joined'; peerId: string } + | { t: 'peer-left'; peerId: string } + | { t: 'sdp'; from: string; to?: string; description: RTCSessionDescriptionInit } + | { t: 'ice'; from: string; to?: string; candidate: RTCIceCandidateInit } + | { t: 'error'; message: string }; + +/** + * Callbacks for WebRTC manager state changes + */ +export interface WebRTCCallbacks { + onLocalStream?: (stream: MediaStream) => void; + onRemoteStream?: (stream: MediaStream) => void; + onStatusChange?: (status: WebRTCStatus) => void; + onError?: (error: Error) => void; + onPeerJoined?: (peerId: string) => void; + onPeerLeft?: (peerId: string) => void; +} + +/** + * Options for WebRTCManager + */ +export interface WebRTCManagerOptions { + /** WebSocket signaling URL */ + signalUrl: string; + /** Room ID to join */ + roomId: string; + /** Whether this peer is "polite" in perfect negotiation (default: true) */ + polite?: boolean; + /** ICE servers configuration */ + iceServers?: RTCIceServer[]; + /** Media constraints for getUserMedia */ + media?: MediaStreamConstraints; + /** Callbacks for state changes */ + callbacks?: WebRTCCallbacks; +} + +/** + * WebRTC manager state + */ +export interface WebRTCManagerState { + status: WebRTCStatus; + peerId: string | null; + remotePeerId: string | null; + isAudioMuted: boolean; + isVideoMuted: boolean; +} + +/** + * Default ICE servers (public STUN servers) + */ +const DEFAULT_ICE_SERVERS: RTCIceServer[] = [ + { urls: 'stun:stun.l.google.com:19302' }, + { urls: 'stun:stun1.l.google.com:19302' }, +]; + +/** + * Framework-agnostic WebRTC connection manager with signaling, + * perfect negotiation, and media stream handling. + */ +export class WebRTCManager { + private ws: WebSocket | null = null; + private pc: RTCPeerConnection | null = null; + private localStream: MediaStream | null = null; + private remoteStream: MediaStream | null = null; + + private peerId: string | null = null; + private remotePeerId: string | null = null; + private status: WebRTCStatus = 'disconnected'; + private isAudioMuted = false; + private isVideoMuted = false; + + // Perfect negotiation state + private makingOffer = false; + private ignoreOffer = false; + private polite: boolean; + + // ICE candidate buffering - buffer until remote description is set + private pendingCandidates: RTCIceCandidateInit[] = []; + private hasRemoteDescription = false; + + private options: WebRTCManagerOptions; + private callbacks: WebRTCCallbacks; + + constructor(options: WebRTCManagerOptions) { + this.options = options; + this.polite = options.polite ?? true; + this.callbacks = options.callbacks ?? {}; + } + + /** + * Get current manager state + */ + getState(): WebRTCManagerState { + return { + status: this.status, + peerId: this.peerId, + remotePeerId: this.remotePeerId, + isAudioMuted: this.isAudioMuted, + isVideoMuted: this.isVideoMuted, + }; + } + + /** + * Get local media stream + */ + getLocalStream(): MediaStream | null { + return this.localStream; + } + + /** + * Get remote media stream + */ + getRemoteStream(): MediaStream | null { + return this.remoteStream; + } + + private setStatus(status: WebRTCStatus): void { + this.status = status; + this.callbacks.onStatusChange?.(status); + } + + private send(msg: SignalMsg): void { + if (this.ws?.readyState === WebSocket.OPEN) { + this.ws.send(JSON.stringify(msg)); + } + } + + /** + * Connect to the signaling server and start the call + */ + async connect(): Promise { + if (this.status !== 'disconnected') return; + + this.setStatus('connecting'); + + try { + // Get local media + const mediaConstraints = this.options.media ?? { video: true, audio: true }; + this.localStream = await navigator.mediaDevices.getUserMedia(mediaConstraints); + this.callbacks.onLocalStream?.(this.localStream); + + // Connect to signaling server + this.ws = new WebSocket(this.options.signalUrl); + + this.ws.onopen = () => { + this.setStatus('signaling'); + this.send({ t: 'join', roomId: this.options.roomId }); + }; + + this.ws.onmessage = (event) => { + const msg = JSON.parse(event.data) as SignalMsg; + this.handleSignalingMessage(msg); + }; + + this.ws.onerror = () => { + this.callbacks.onError?.(new Error('WebSocket connection error')); + }; + + this.ws.onclose = () => { + if (this.status !== 'disconnected') { + this.setStatus('disconnected'); + } + }; + } catch (err) { + this.setStatus('disconnected'); + this.callbacks.onError?.(err instanceof Error ? err : new Error(String(err))); + } + } + + private async handleSignalingMessage(msg: SignalMsg): Promise { + switch (msg.t) { + case 'joined': + this.peerId = msg.peerId; + // If there's already a peer in the room, we're the offerer (impolite) + if (msg.peers.length > 0) { + this.remotePeerId = msg.peers[0]; + // Late joiner is impolite (makes the offer, wins collisions) + this.polite = this.options.polite ?? false; + await this.createPeerConnection(); + await this.createOffer(); + } else { + // First peer is polite (waits for offers, yields on collision) + this.polite = this.options.polite ?? true; + } + break; + + case 'peer-joined': + this.remotePeerId = msg.peerId; + this.callbacks.onPeerJoined?.(msg.peerId); + // New peer joined, wait for their offer (they initiate) + await this.createPeerConnection(); + break; + + case 'peer-left': + this.callbacks.onPeerLeft?.(msg.peerId); + if (msg.peerId === this.remotePeerId) { + this.remotePeerId = null; + this.closePeerConnection(); + this.setStatus('signaling'); + } + break; + + case 'sdp': + await this.handleRemoteSDP(msg.description); + break; + + case 'ice': + await this.handleRemoteICE(msg.candidate); + break; + + case 'error': + this.callbacks.onError?.(new Error(msg.message)); + break; + } + } + + private async createPeerConnection(): Promise { + if (this.pc) return; + + const iceServers = this.options.iceServers ?? DEFAULT_ICE_SERVERS; + this.pc = new RTCPeerConnection({ iceServers }); + + // Add local tracks + if (this.localStream) { + for (const track of this.localStream.getTracks()) { + this.pc.addTrack(track, this.localStream); + } + } + + // Handle remote tracks + this.pc.ontrack = (event) => { + // Use the stream from the event if available (preferred - already has track) + // Otherwise create a new stream with the track + if (event.streams?.[0]) { + if (this.remoteStream !== event.streams[0]) { + this.remoteStream = event.streams[0]; + this.callbacks.onRemoteStream?.(this.remoteStream); + } + } else { + // Fallback: create stream with track already included + if (!this.remoteStream) { + this.remoteStream = new MediaStream([event.track]); + this.callbacks.onRemoteStream?.(this.remoteStream); + } else { + this.remoteStream.addTrack(event.track); + // Re-trigger callback so video element updates + this.callbacks.onRemoteStream?.(this.remoteStream); + } + } + + if (this.status !== 'connected') { + this.setStatus('connected'); + } + }; + + // Handle ICE candidates + this.pc.onicecandidate = (event) => { + if (event.candidate) { + this.send({ + t: 'ice', + from: this.peerId!, + to: this.remotePeerId ?? undefined, + candidate: event.candidate.toJSON(), + }); + } + }; + + // Perfect negotiation: handle negotiation needed + this.pc.onnegotiationneeded = async () => { + try { + this.makingOffer = true; + await this.pc!.setLocalDescription(); + this.send({ + t: 'sdp', + from: this.peerId!, + to: this.remotePeerId ?? undefined, + description: this.pc!.localDescription!, + }); + } catch (err) { + this.callbacks.onError?.(err instanceof Error ? err : new Error(String(err))); + } finally { + this.makingOffer = false; + } + }; + + this.pc.oniceconnectionstatechange = () => { + if (this.pc?.iceConnectionState === 'disconnected') { + this.setStatus('signaling'); + } else if (this.pc?.iceConnectionState === 'connected') { + this.setStatus('connected'); + } + }; + } + + private async createOffer(): Promise { + if (!this.pc) return; + + try { + this.makingOffer = true; + const offer = await this.pc.createOffer(); + await this.pc.setLocalDescription(offer); + + this.send({ + t: 'sdp', + from: this.peerId!, + to: this.remotePeerId ?? undefined, + description: this.pc.localDescription!, + }); + } finally { + this.makingOffer = false; + } + } + + private async handleRemoteSDP(description: RTCSessionDescriptionInit): Promise { + if (!this.pc) { + await this.createPeerConnection(); + } + + const pc = this.pc!; + const isOffer = description.type === 'offer'; + + // Perfect negotiation: collision detection + const offerCollision = isOffer && (this.makingOffer || pc.signalingState !== 'stable'); + + this.ignoreOffer = !this.polite && offerCollision; + if (this.ignoreOffer) return; + + await pc.setRemoteDescription(description); + this.hasRemoteDescription = true; + + // Flush buffered ICE candidates now that remote description is set + for (const candidate of this.pendingCandidates) { + try { + await pc.addIceCandidate(candidate); + } catch (err) { + // Ignore errors for candidates that arrived during collision + if (!this.ignoreOffer) { + console.warn('Failed to add buffered ICE candidate:', err); + } + } + } + this.pendingCandidates = []; + + if (isOffer) { + await pc.setLocalDescription(); + this.send({ + t: 'sdp', + from: this.peerId!, + to: this.remotePeerId ?? undefined, + description: pc.localDescription!, + }); + } + } + + private async handleRemoteICE(candidate: RTCIceCandidateInit): Promise { + // Buffer candidates until peer connection AND remote description are ready + if (!this.pc || !this.hasRemoteDescription) { + this.pendingCandidates.push(candidate); + return; + } + + try { + await this.pc.addIceCandidate(candidate); + } catch (err) { + if (!this.ignoreOffer) { + // Log but don't propagate - some ICE failures are normal + console.warn('Failed to add ICE candidate:', err); + } + } + } + + private closePeerConnection(): void { + if (this.pc) { + this.pc.close(); + this.pc = null; + } + this.remoteStream = null; + this.pendingCandidates = []; + this.makingOffer = false; + this.ignoreOffer = false; + this.hasRemoteDescription = false; + } + + /** + * End the call and disconnect + */ + hangup(): void { + this.closePeerConnection(); + + if (this.localStream) { + for (const track of this.localStream.getTracks()) { + track.stop(); + } + this.localStream = null; + } + + if (this.ws) { + this.ws.close(); + this.ws = null; + } + + this.peerId = null; + this.remotePeerId = null; + this.setStatus('disconnected'); + } + + /** + * Mute or unmute audio + */ + muteAudio(muted: boolean): void { + if (this.localStream) { + for (const track of this.localStream.getAudioTracks()) { + track.enabled = !muted; + } + } + this.isAudioMuted = muted; + } + + /** + * Mute or unmute video + */ + muteVideo(muted: boolean): void { + if (this.localStream) { + for (const track of this.localStream.getVideoTracks()) { + track.enabled = !muted; + } + } + this.isVideoMuted = muted; + } + + /** + * Clean up all resources + */ + dispose(): void { + this.hangup(); + } +} diff --git a/packages/react/src/index.ts b/packages/react/src/index.ts index 28b26ba5b..e6c2babcf 100644 --- a/packages/react/src/index.ts +++ b/packages/react/src/index.ts @@ -30,6 +30,12 @@ export { type SSERouteOutput, type EventStreamOptions, } from './eventstream'; +export { + useWebRTCCall, + type UseWebRTCCallOptions, + type UseWebRTCCallResult, + type WebRTCStatus, +} from './webrtc'; export { useAPI, type RouteKey, @@ -65,6 +71,10 @@ export { type EventStreamCallbacks, type EventStreamManagerOptions, type EventStreamManagerState, + WebRTCManager, + type WebRTCCallbacks, + type WebRTCManagerOptions, + type WebRTCManagerState, // Client type exports (createClient is exported from ./client.ts) type Client, type ClientOptions, diff --git a/packages/react/src/webrtc.tsx b/packages/react/src/webrtc.tsx new file mode 100644 index 000000000..5cec7c5ff --- /dev/null +++ b/packages/react/src/webrtc.tsx @@ -0,0 +1,213 @@ +import { useCallback, useContext, useEffect, useMemo, useRef, useState } from 'react'; +import { + WebRTCManager, + buildUrl, + type WebRTCStatus, + type WebRTCManagerOptions, +} from '@agentuity/frontend'; +import { AgentuityContext } from './context'; + +export type { WebRTCStatus }; + +/** + * Options for useWebRTCCall hook + */ +export interface UseWebRTCCallOptions { + /** Room ID to join */ + roomId: string; + /** WebSocket signaling URL (e.g., '/call/signal' or full URL) */ + signalUrl: string; + /** Whether this peer is "polite" in perfect negotiation (default: true for first joiner) */ + polite?: boolean; + /** ICE servers configuration */ + iceServers?: RTCIceServer[]; + /** Media constraints for getUserMedia */ + media?: MediaStreamConstraints; + /** Whether to auto-connect on mount (default: true) */ + autoConnect?: boolean; +} + +/** + * Return type for useWebRTCCall hook + */ +export interface UseWebRTCCallResult { + /** Ref to attach to local video element */ + localVideoRef: React.RefObject; + /** Ref to attach to remote video element */ + remoteVideoRef: React.RefObject; + /** Current connection status */ + status: WebRTCStatus; + /** Current error if any */ + error: Error | null; + /** Local peer ID assigned by server */ + peerId: string | null; + /** Remote peer ID */ + remotePeerId: string | null; + /** Whether audio is muted */ + isAudioMuted: boolean; + /** Whether video is muted */ + isVideoMuted: boolean; + /** Manually start the connection (if autoConnect is false) */ + connect: () => void; + /** End the call */ + hangup: () => void; + /** Mute or unmute audio */ + muteAudio: (muted: boolean) => void; + /** Mute or unmute video */ + muteVideo: (muted: boolean) => void; +} + +/** + * React hook for WebRTC peer-to-peer audio/video calls. + * + * Handles WebRTC signaling, media capture, and peer connection management. + * + * @example + * ```tsx + * function VideoCall({ roomId }: { roomId: string }) { + * const { + * localVideoRef, + * remoteVideoRef, + * status, + * hangup, + * muteAudio, + * isAudioMuted, + * } = useWebRTCCall({ + * roomId, + * signalUrl: '/call/signal', + * }); + * + * return ( + *
+ *
+ * ); + * } + * ``` + */ +export function useWebRTCCall(options: UseWebRTCCallOptions): UseWebRTCCallResult { + const context = useContext(AgentuityContext); + + const managerRef = useRef(null); + const localVideoRef = useRef(null); + const remoteVideoRef = useRef(null); + + const [status, setStatus] = useState('disconnected'); + const [error, setError] = useState(null); + const [peerId, setPeerId] = useState(null); + const [remotePeerId, setRemotePeerId] = useState(null); + const [isAudioMuted, setIsAudioMuted] = useState(false); + const [isVideoMuted, setIsVideoMuted] = useState(false); + + // Build full signaling URL + const signalUrl = useMemo(() => { + // If it's already a full URL, use as-is + if (options.signalUrl.startsWith('ws://') || options.signalUrl.startsWith('wss://')) { + return options.signalUrl; + } + + // Build from context base URL + const base = context?.baseUrl ?? window.location.origin; + const wsBase = base.replace(/^http(s?):/, 'ws$1:'); + return buildUrl(wsBase, options.signalUrl); + }, [context?.baseUrl, options.signalUrl]); + + // Create manager options - use refs to avoid recreating manager on state changes + const managerOptions = useMemo((): WebRTCManagerOptions => { + return { + signalUrl, + roomId: options.roomId, + polite: options.polite, + iceServers: options.iceServers, + media: options.media, + callbacks: { + onLocalStream: (stream) => { + if (localVideoRef.current) { + localVideoRef.current.srcObject = stream; + } + }, + onRemoteStream: (stream) => { + if (remoteVideoRef.current) { + remoteVideoRef.current.srcObject = stream; + } + }, + onStatusChange: (newStatus) => { + setStatus(newStatus); + if (managerRef.current) { + const state = managerRef.current.getState(); + setPeerId(state.peerId); + setRemotePeerId(state.remotePeerId); + } + }, + onError: (err) => { + setError(err); + }, + onPeerJoined: (id) => { + setRemotePeerId(id); + }, + onPeerLeft: (id) => { + setRemotePeerId((current) => current === id ? null : current); + }, + }, + }; + // eslint-disable-next-line react-hooks/exhaustive-deps + }, [signalUrl, options.roomId, options.polite, options.iceServers, options.media]); + + // Initialize manager + useEffect(() => { + const manager = new WebRTCManager(managerOptions); + managerRef.current = manager; + + // Auto-connect if enabled (default: true) + if (options.autoConnect !== false) { + manager.connect(); + } + + return () => { + manager.dispose(); + managerRef.current = null; + }; + }, [managerOptions, options.autoConnect]); + + const connect = useCallback(() => { + managerRef.current?.connect(); + }, []); + + const hangup = useCallback(() => { + managerRef.current?.hangup(); + setStatus('disconnected'); + setPeerId(null); + setRemotePeerId(null); + }, []); + + const muteAudio = useCallback((muted: boolean) => { + managerRef.current?.muteAudio(muted); + setIsAudioMuted(muted); + }, []); + + const muteVideo = useCallback((muted: boolean) => { + managerRef.current?.muteVideo(muted); + setIsVideoMuted(muted); + }, []); + + return { + localVideoRef, + remoteVideoRef, + status, + error, + peerId, + remotePeerId, + isAudioMuted, + isVideoMuted, + connect, + hangup, + muteAudio, + muteVideo, + }; +} diff --git a/packages/runtime/src/index.ts b/packages/runtime/src/index.ts index 14a4737f8..bbeea41b8 100644 --- a/packages/runtime/src/index.ts +++ b/packages/runtime/src/index.ts @@ -59,6 +59,15 @@ export { registerDevModeRoutes } from './devmode'; // router.ts exports export { type HonoEnv, type WebSocketConnection, createRouter } from './router'; +// webrtc-signaling.ts exports +export { + type SignalMsg, + type SDPDescription, + type ICECandidate, + type WebRTCOptions, + WebRTCRoomManager, +} from './webrtc-signaling'; + // eval.ts exports export { type EvalContext, diff --git a/packages/runtime/src/router.ts b/packages/runtime/src/router.ts index 011cf5398..af1e3b92f 100644 --- a/packages/runtime/src/router.ts +++ b/packages/runtime/src/router.ts @@ -14,6 +14,7 @@ import { hash, returnResponse } from './_util'; import type { Env } from './app'; import { getAgentAsyncLocalStorage } from './_context'; import { parseEmail, type Email } from './io/email'; +import { WebRTCRoomManager, type WebRTCOptions } from './webrtc-signaling'; // Re-export both Env types export type { Env }; @@ -274,6 +275,28 @@ declare module 'hono' { middleware: MiddlewareHandler, handler: (c: Context) => (stream: any) => void ): this; + + /** + * Create a WebRTC signaling endpoint for peer-to-peer audio/video communication. + * + * Registers a WebSocket signaling route at `${path}/signal` that handles: + * - Room membership and peer discovery + * - SDP offer/answer relay + * - ICE candidate relay + * + * @param path - The base route path (e.g., '/call') + * @param options - Optional configuration for the WebRTC endpoint + * + * @example + * ```typescript + * // Create a WebRTC signaling endpoint at /call/signal + * router.webrtc('/call'); + * + * // With options + * router.webrtc('/call', { maxPeers: 2 }); + * ``` + */ + webrtc(path: string, options?: WebRTCOptions): this; } } @@ -284,6 +307,7 @@ declare module 'hono' { * - **stream()** - Stream responses with ReadableStream * - **websocket()** - WebSocket connections * - **sse()** - Server-Sent Events + * - **webrtc()** - WebRTC signaling for peer-to-peer communication * - **email()** - Email handler routing * - **sms()** - SMS handler routing * - **cron()** - Scheduled task routing @@ -715,5 +739,41 @@ export const createRouter = (): } }; + _router.webrtc = (path: string, options?: WebRTCOptions) => { + const roomManager = new WebRTCRoomManager(options); + const signalPath = `${path}/signal`; + + // Use the existing websocket implementation for the signaling route + const wrapper = upgradeWebSocket((_c: Context) => { + let currentWs: WebSocketConnection | undefined; + + return { + onOpen: (_event: any, ws: any) => { + currentWs = { + onOpen: () => {}, + onMessage: () => {}, + onClose: () => {}, + send: (data) => ws.send(data), + }; + }, + onMessage: (event: any, _ws: any) => { + if (currentWs) { + roomManager.handleMessage(currentWs, String(event.data)); + } + }, + onClose: (_event: any, _ws: any) => { + if (currentWs) { + roomManager.handleDisconnect(currentWs); + } + }, + }; + }); + + const wsMiddleware: MiddlewareHandler = (c, next) => + (wrapper as unknown as MiddlewareHandler)(c, next); + + return router.get(signalPath, wsMiddleware); + }; + return router; }; diff --git a/packages/runtime/src/webrtc-signaling.ts b/packages/runtime/src/webrtc-signaling.ts new file mode 100644 index 000000000..b4b708f59 --- /dev/null +++ b/packages/runtime/src/webrtc-signaling.ts @@ -0,0 +1,226 @@ +import type { WebSocketConnection } from './router'; + +// WebRTC types for signaling (not using DOM types since this runs on server) +export interface SDPDescription { + type: 'offer' | 'answer' | 'pranswer' | 'rollback'; + sdp?: string; +} + +export interface ICECandidate { + candidate?: string; + sdpMid?: string | null; + sdpMLineIndex?: number | null; + usernameFragment?: string | null; +} + +// Signaling message protocol +export type SignalMsg = + | { t: 'join'; roomId: string } + | { t: 'joined'; peerId: string; roomId: string; peers: string[] } + | { t: 'peer-joined'; peerId: string } + | { t: 'peer-left'; peerId: string } + | { t: 'sdp'; from: string; to?: string; description: SDPDescription } + | { t: 'ice'; from: string; to?: string; candidate: ICECandidate } + | { t: 'error'; message: string }; + +export interface WebRTCOptions { + /** Maximum number of peers per room (default: 2) */ + maxPeers?: number; +} + +interface PeerConnection { + ws: WebSocketConnection; + roomId: string; +} + +/** + * In-memory room manager for WebRTC signaling. + * Tracks rooms and their connected peers. + */ +export class WebRTCRoomManager { + // roomId -> Map + private rooms = new Map>(); + // ws -> peerId (reverse lookup for cleanup) + private wsToPeer = new Map(); + private maxPeers: number; + private peerIdCounter = 0; + + constructor(options?: WebRTCOptions) { + this.maxPeers = options?.maxPeers ?? 2; + } + + private generatePeerId(): string { + return `peer-${Date.now()}-${++this.peerIdCounter}`; + } + + private send(ws: WebSocketConnection, msg: SignalMsg): void { + ws.send(JSON.stringify(msg)); + } + + private broadcast(roomId: string, msg: SignalMsg, excludePeerId?: string): void { + const room = this.rooms.get(roomId); + if (!room) return; + + for (const [peerId, peer] of room) { + if (peerId !== excludePeerId) { + this.send(peer.ws, msg); + } + } + } + + /** + * Handle a peer joining a room + */ + handleJoin(ws: WebSocketConnection, roomId: string): void { + let room = this.rooms.get(roomId); + + // Create room if it doesn't exist + if (!room) { + room = new Map(); + this.rooms.set(roomId, room); + } + + // Check room capacity + if (room.size >= this.maxPeers) { + this.send(ws, { t: 'error', message: `Room is full (max ${this.maxPeers} peers)` }); + return; + } + + const peerId = this.generatePeerId(); + const existingPeers = Array.from(room.keys()); + + // Add peer to room + room.set(peerId, { ws, roomId }); + this.wsToPeer.set(ws, { peerId, roomId }); + + // Send joined confirmation with list of existing peers + this.send(ws, { t: 'joined', peerId, roomId, peers: existingPeers }); + + // Notify existing peers about new peer + this.broadcast(roomId, { t: 'peer-joined', peerId }, peerId); + } + + /** + * Handle a peer disconnecting + */ + handleDisconnect(ws: WebSocketConnection): void { + const peerInfo = this.wsToPeer.get(ws); + if (!peerInfo) return; + + const { peerId, roomId } = peerInfo; + const room = this.rooms.get(roomId); + + if (room) { + room.delete(peerId); + + // Notify remaining peers + this.broadcast(roomId, { t: 'peer-left', peerId }); + + // Clean up empty room + if (room.size === 0) { + this.rooms.delete(roomId); + } + } + + this.wsToPeer.delete(ws); + } + + /** + * Relay SDP message to target peer(s) + */ + handleSDP(ws: WebSocketConnection, to: string | undefined, description: SDPDescription): void { + const peerInfo = this.wsToPeer.get(ws); + if (!peerInfo) { + this.send(ws, { t: 'error', message: 'Not in a room' }); + return; + } + + const { peerId, roomId } = peerInfo; + const room = this.rooms.get(roomId); + if (!room) return; + + // Server injects 'from' to prevent spoofing + const msg: SignalMsg = { t: 'sdp', from: peerId, description }; + + if (to) { + // Send to specific peer + const targetPeer = room.get(to); + if (targetPeer) { + this.send(targetPeer.ws, msg); + } + } else { + // Broadcast to all peers in room + this.broadcast(roomId, msg, peerId); + } + } + + /** + * Relay ICE candidate to target peer(s) + */ + handleICE(ws: WebSocketConnection, to: string | undefined, candidate: ICECandidate): void { + const peerInfo = this.wsToPeer.get(ws); + if (!peerInfo) { + this.send(ws, { t: 'error', message: 'Not in a room' }); + return; + } + + const { peerId, roomId } = peerInfo; + const room = this.rooms.get(roomId); + if (!room) return; + + // Server injects 'from' to prevent spoofing + const msg: SignalMsg = { t: 'ice', from: peerId, candidate }; + + if (to) { + // Send to specific peer + const targetPeer = room.get(to); + if (targetPeer) { + this.send(targetPeer.ws, msg); + } + } else { + // Broadcast to all peers in room + this.broadcast(roomId, msg, peerId); + } + } + + /** + * Handle incoming signaling message + */ + handleMessage(ws: WebSocketConnection, data: string): void { + let msg: SignalMsg; + try { + msg = JSON.parse(data); + } catch { + this.send(ws, { t: 'error', message: 'Invalid JSON' }); + return; + } + + switch (msg.t) { + case 'join': + this.handleJoin(ws, msg.roomId); + break; + case 'sdp': + this.handleSDP(ws, msg.to, msg.description); + break; + case 'ice': + this.handleICE(ws, msg.to, msg.candidate); + break; + default: + this.send(ws, { + t: 'error', + message: `Unknown message type: ${(msg as { t: string }).t}`, + }); + } + } + + /** + * Get room stats for debugging + */ + getRoomStats(): { roomCount: number; totalPeers: number } { + let totalPeers = 0; + for (const room of this.rooms.values()) { + totalPeers += room.size; + } + return { roomCount: this.rooms.size, totalPeers }; + } +} diff --git a/packages/runtime/test/webrtc-signaling.test.ts b/packages/runtime/test/webrtc-signaling.test.ts new file mode 100644 index 000000000..2c96bf459 --- /dev/null +++ b/packages/runtime/test/webrtc-signaling.test.ts @@ -0,0 +1,324 @@ +import { describe, test, expect, beforeEach } from 'bun:test'; +import { + WebRTCRoomManager, + type SignalMsg, + type SDPDescription, + type ICECandidate, +} from '../src/webrtc-signaling'; +import type { WebSocketConnection } from '../src/router'; + +// Mock WebSocket connection +function createMockWs(): WebSocketConnection & { messages: string[] } { + const messages: string[] = []; + return { + messages, + onOpen: () => {}, + onMessage: () => {}, + onClose: () => {}, + send: (data: string | ArrayBuffer | Uint8Array) => { + messages.push(typeof data === 'string' ? data : data.toString()); + }, + }; +} + +function parseMessage(ws: { messages: string[] }, index = -1): SignalMsg { + const idx = index < 0 ? ws.messages.length + index : index; + return JSON.parse(ws.messages[idx]); +} + +describe('WebRTCRoomManager', () => { + let roomManager: WebRTCRoomManager; + + beforeEach(() => { + roomManager = new WebRTCRoomManager({ maxPeers: 2 }); + }); + + describe('handleJoin', () => { + test('should assign peerId and send joined message', () => { + const ws = createMockWs(); + roomManager.handleJoin(ws, 'room-1'); + + expect(ws.messages.length).toBe(1); + const msg = parseMessage(ws); + expect(msg.t).toBe('joined'); + if (msg.t === 'joined') { + expect(msg.peerId).toMatch(/^peer-/); + expect(msg.roomId).toBe('room-1'); + expect(msg.peers).toEqual([]); + } + }); + + test('should include existing peers in joined message', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + const msg1 = parseMessage(ws1); + const peer1Id = msg1.t === 'joined' ? msg1.peerId : ''; + + roomManager.handleJoin(ws2, 'room-1'); + const msg2 = parseMessage(ws2); + + expect(msg2.t).toBe('joined'); + if (msg2.t === 'joined') { + expect(msg2.peers).toContain(peer1Id); + } + }); + + test('should notify existing peers when new peer joins', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + + // ws1 should receive peer-joined notification + expect(ws1.messages.length).toBe(2); + const notification = parseMessage(ws1); + expect(notification.t).toBe('peer-joined'); + }); + + test('should reject peer when room is full', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + const ws3 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + roomManager.handleJoin(ws3, 'room-1'); + + const msg = parseMessage(ws3); + expect(msg.t).toBe('error'); + if (msg.t === 'error') { + expect(msg.message).toContain('full'); + } + }); + + test('should allow joining different rooms', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + const ws3 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + roomManager.handleJoin(ws3, 'room-2'); + + // ws3 should successfully join room-2 + const msg = parseMessage(ws3); + expect(msg.t).toBe('joined'); + }); + }); + + describe('handleDisconnect', () => { + test('should remove peer from room and notify others', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + const msg1 = parseMessage(ws1); + const peer1Id = msg1.t === 'joined' ? msg1.peerId : ''; + + roomManager.handleJoin(ws2, 'room-1'); + ws1.messages.length = 0; // Clear in-place + + roomManager.handleDisconnect(ws1); + + // ws2 should receive peer-left notification + const notification = parseMessage(ws2); + expect(notification.t).toBe('peer-left'); + if (notification.t === 'peer-left') { + expect(notification.peerId).toBe(peer1Id); + } + }); + + test('should allow new peer after disconnect', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + const ws3 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + roomManager.handleDisconnect(ws1); + roomManager.handleJoin(ws3, 'room-1'); + + // ws3 should successfully join + const msg = parseMessage(ws3); + expect(msg.t).toBe('joined'); + }); + + test('should clean up empty rooms', () => { + const ws1 = createMockWs(); + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleDisconnect(ws1); + + const stats = roomManager.getRoomStats(); + expect(stats.roomCount).toBe(0); + expect(stats.totalPeers).toBe(0); + }); + }); + + describe('handleSDP', () => { + test('should relay SDP to target peer', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + + const msg2 = parseMessage(ws2); + const peer2Id = msg2.t === 'joined' ? msg2.peerId : ''; + + ws2.messages.length = 0; // Clear in-place + + const sdp: SDPDescription = { type: 'offer', sdp: 'test-sdp' }; + roomManager.handleSDP(ws1, peer2Id, sdp); + + const relayed = parseMessage(ws2); + expect(relayed.t).toBe('sdp'); + if (relayed.t === 'sdp') { + expect(relayed.description).toEqual(sdp); + expect(relayed.from).toMatch(/^peer-/); // Server-injected from + } + }); + + test('should broadcast SDP to all peers if no target', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + + ws2.messages.length = 0; // Clear in-place + + const sdp: SDPDescription = { type: 'offer', sdp: 'test-sdp' }; + roomManager.handleSDP(ws1, undefined, sdp); + + const relayed = parseMessage(ws2); + expect(relayed.t).toBe('sdp'); + }); + + test('should return error if not in a room', () => { + const ws = createMockWs(); + const sdp: SDPDescription = { type: 'offer', sdp: 'test-sdp' }; + roomManager.handleSDP(ws, undefined, sdp); + + const msg = parseMessage(ws); + expect(msg.t).toBe('error'); + }); + }); + + describe('handleICE', () => { + test('should relay ICE candidate to target peer', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + + const msg2 = parseMessage(ws2); + const peer2Id = msg2.t === 'joined' ? msg2.peerId : ''; + + ws2.messages.length = 0; // Clear in-place + + const candidate: ICECandidate = { candidate: 'test-candidate', sdpMid: '0' }; + roomManager.handleICE(ws1, peer2Id, candidate); + + const relayed = parseMessage(ws2); + expect(relayed.t).toBe('ice'); + if (relayed.t === 'ice') { + expect(relayed.candidate).toEqual(candidate); + expect(relayed.from).toMatch(/^peer-/); + } + }); + }); + + describe('handleMessage', () => { + test('should parse and route join messages', () => { + const ws = createMockWs(); + roomManager.handleMessage(ws, JSON.stringify({ t: 'join', roomId: 'room-1' })); + + const msg = parseMessage(ws); + expect(msg.t).toBe('joined'); + }); + + test('should parse and route sdp messages', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + + ws2.messages.length = 0; // Clear in-place + + const sdpMsg = { + t: 'sdp', + from: 'ignored', // Server should override this + description: { type: 'offer', sdp: 'test' }, + }; + roomManager.handleMessage(ws1, JSON.stringify(sdpMsg)); + + const relayed = parseMessage(ws2); + expect(relayed.t).toBe('sdp'); + }); + + test('should return error for invalid JSON', () => { + const ws = createMockWs(); + roomManager.handleMessage(ws, 'not-json'); + + const msg = parseMessage(ws); + expect(msg.t).toBe('error'); + if (msg.t === 'error') { + expect(msg.message).toContain('Invalid JSON'); + } + }); + + test('should return error for unknown message type', () => { + const ws = createMockWs(); + roomManager.handleMessage(ws, JSON.stringify({ t: 'unknown' })); + + const msg = parseMessage(ws); + expect(msg.t).toBe('error'); + if (msg.t === 'error') { + expect(msg.message).toContain('Unknown message type'); + } + }); + }); + + describe('getRoomStats', () => { + test('should return correct room and peer counts', () => { + const ws1 = createMockWs(); + const ws2 = createMockWs(); + const ws3 = createMockWs(); + + roomManager.handleJoin(ws1, 'room-1'); + roomManager.handleJoin(ws2, 'room-1'); + roomManager.handleJoin(ws3, 'room-2'); + + const stats = roomManager.getRoomStats(); + expect(stats.roomCount).toBe(2); + expect(stats.totalPeers).toBe(3); + }); + }); + + describe('maxPeers configuration', () => { + test('should respect custom maxPeers limit', () => { + const manager = new WebRTCRoomManager({ maxPeers: 3 }); + const ws1 = createMockWs(); + const ws2 = createMockWs(); + const ws3 = createMockWs(); + const ws4 = createMockWs(); + + manager.handleJoin(ws1, 'room-1'); + manager.handleJoin(ws2, 'room-1'); + manager.handleJoin(ws3, 'room-1'); + manager.handleJoin(ws4, 'room-1'); + + // ws4 should be rejected + const msg = parseMessage(ws4); + expect(msg.t).toBe('error'); + + const stats = manager.getRoomStats(); + expect(stats.totalPeers).toBe(3); + }); + }); +}); From 26f854e0ae348406917e33cd4fd380fc025d49c0 Mon Sep 17 00:00:00 2001 From: Bobby Christopher Date: Sun, 21 Dec 2025 11:34:34 -0500 Subject: [PATCH 02/20] fix: remove undefined eslint rule reference --- packages/react/src/webrtc.tsx | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/packages/react/src/webrtc.tsx b/packages/react/src/webrtc.tsx index 5cec7c5ff..521958242 100644 --- a/packages/react/src/webrtc.tsx +++ b/packages/react/src/webrtc.tsx @@ -156,7 +156,7 @@ export function useWebRTCCall(options: UseWebRTCCallOptions): UseWebRTCCallResul }, }, }; - // eslint-disable-next-line react-hooks/exhaustive-deps + // eslint-disable-next-line }, [signalUrl, options.roomId, options.polite, options.iceServers, options.media]); // Initialize manager From 719fa438b86793bf45acaa54c0f069e30dd57458 Mon Sep 17 00:00:00 2001 From: Bobby Christopher Date: Sun, 21 Dec 2025 14:35:13 -0500 Subject: [PATCH 03/20] feat(webrtc): add state machine and SDK callbacks MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit - Add shared WebRTC types to @agentuity/core (SignalMessage, SDPDescription, ICECandidate, WebRTCConnectionState, WebRTCDisconnectReason, WebRTCSignalingCallbacks) - Refactor frontend WebRTCManager with explicit state machine (idle → connecting → signaling → negotiating → connected) - Add WebRTCClientCallbacks interface for SDK users to hook into all lifecycle events - Add WebRTCSignalingCallbacks to backend for room/peer event monitoring - Update React hook to expose new state and forward user callbacks - Add 7 new tests for backend callback functionality BREAKING CHANGE: WebRTCManagerState now has 'state' property (WebRTCConnectionState). 'status' is deprecated but still available for backwards compatibility. Amp-Thread-ID: https://ampcode.com/threads/T-019b4261-8e29-754e-a659-41f5d637485c Co-authored-by: Amp --- packages/core/src/index.ts | 11 + packages/core/src/webrtc.ts | 132 ++++++ packages/frontend/src/index.ts | 4 + packages/frontend/src/webrtc-manager.ts | 271 +++++++++++-- packages/react/src/index.ts | 3 + packages/react/src/webrtc.tsx | 381 ++++++++++-------- packages/runtime/src/index.ts | 2 + packages/runtime/src/webrtc-signaling.ts | 103 +++-- .../runtime/test/webrtc-signaling.test.ts | 113 ++++++ 9 files changed, 787 insertions(+), 233 deletions(-) create mode 100644 packages/core/src/webrtc.ts diff --git a/packages/core/src/index.ts b/packages/core/src/index.ts index 0e5f40d51..58c7d4ba1 100644 --- a/packages/core/src/index.ts +++ b/packages/core/src/index.ts @@ -91,4 +91,15 @@ export { type WorkbenchConfig, } from './workbench-config'; +// webrtc.ts exports +export type { + SDPDescription, + ICECandidate, + SignalMessage, + SignalMsg, + WebRTCConnectionState, + WebRTCDisconnectReason, + WebRTCSignalingCallbacks, +} from './webrtc'; + // Client code moved to @agentuity/frontend for better bundler compatibility diff --git a/packages/core/src/webrtc.ts b/packages/core/src/webrtc.ts new file mode 100644 index 000000000..7cca51a13 --- /dev/null +++ b/packages/core/src/webrtc.ts @@ -0,0 +1,132 @@ +/** + * WebRTC signaling types shared between server and client. + */ + +// ============================================================================= +// Signaling Protocol Types +// ============================================================================= + +/** + * SDP (Session Description Protocol) description for WebRTC negotiation. + */ +export interface SDPDescription { + type: 'offer' | 'answer' | 'pranswer' | 'rollback'; + sdp?: string; +} + +/** + * ICE (Interactive Connectivity Establishment) candidate for NAT traversal. + */ +export interface ICECandidate { + candidate?: string; + sdpMid?: string | null; + sdpMLineIndex?: number | null; + usernameFragment?: string | null; +} + +/** + * Signaling message protocol for WebRTC peer communication. + * + * Message types: + * - `join`: Client requests to join a room + * - `joined`: Server confirms join with peer ID and existing peers + * - `peer-joined`: Server notifies when another peer joins the room + * - `peer-left`: Server notifies when a peer leaves the room + * - `sdp`: SDP offer/answer exchange between peers + * - `ice`: ICE candidate exchange between peers + * - `error`: Error message from server + */ +export type SignalMessage = + | { t: 'join'; roomId: string } + | { t: 'joined'; peerId: string; roomId: string; peers: string[] } + | { t: 'peer-joined'; peerId: string } + | { t: 'peer-left'; peerId: string } + | { t: 'sdp'; from: string; to?: string; description: SDPDescription } + | { t: 'ice'; from: string; to?: string; candidate: ICECandidate } + | { t: 'error'; message: string }; + +/** + * @deprecated Use `SignalMessage` instead. Alias for backwards compatibility. + */ +export type SignalMsg = SignalMessage; + +// ============================================================================= +// Frontend State Machine Types +// ============================================================================= + +/** + * WebRTC connection states for the frontend state machine. + * + * State transitions: + * - idle → connecting: connect() called + * - connecting → signaling: WebSocket opened, joined room + * - connecting → idle: error or cancel + * - signaling → negotiating: peer joined, SDP exchange started + * - signaling → idle: hangup or WebSocket closed + * - negotiating → connected: ICE complete, media flowing + * - negotiating → signaling: peer left during negotiation + * - negotiating → idle: error or hangup + * - connected → negotiating: renegotiation needed + * - connected → signaling: peer left + * - connected → idle: hangup or WebSocket closed + */ +export type WebRTCConnectionState = 'idle' | 'connecting' | 'signaling' | 'negotiating' | 'connected'; + +/** + * Reasons for disconnection. + */ +export type WebRTCDisconnectReason = 'hangup' | 'error' | 'peer-left' | 'timeout'; + +// ============================================================================= +// Backend Signaling Callbacks +// ============================================================================= + +/** + * Callbacks for WebRTC signaling server events. + * All callbacks are optional - only subscribe to events you care about. + */ +export interface WebRTCSignalingCallbacks { + /** + * Called when a new room is created. + * @param roomId - The room ID + */ + onRoomCreated?: (roomId: string) => void; + + /** + * Called when a room is destroyed (last peer left). + * @param roomId - The room ID + */ + onRoomDestroyed?: (roomId: string) => void; + + /** + * Called when a peer joins a room. + * @param peerId - The peer's ID + * @param roomId - The room ID + */ + onPeerJoin?: (peerId: string, roomId: string) => void; + + /** + * Called when a peer leaves a room. + * @param peerId - The peer's ID + * @param roomId - The room ID + * @param reason - Why the peer left + */ + onPeerLeave?: (peerId: string, roomId: string, reason: 'disconnect' | 'kicked') => void; + + /** + * Called when a signaling message is relayed. + * @param type - Message type ('sdp' or 'ice') + * @param from - Sender peer ID + * @param to - Target peer ID (undefined for broadcast) + * @param roomId - The room ID + */ + onMessage?: (type: 'sdp' | 'ice', from: string, to: string | undefined, roomId: string) => void; + + /** + * Called when an error occurs. + * @param error - The error that occurred + * @param peerId - The peer ID if applicable + * @param roomId - The room ID if applicable + */ + onError?: (error: Error, peerId?: string, roomId?: string) => void; +} diff --git a/packages/frontend/src/index.ts b/packages/frontend/src/index.ts index 7a64a187d..178713e7d 100644 --- a/packages/frontend/src/index.ts +++ b/packages/frontend/src/index.ts @@ -24,8 +24,12 @@ export { type WebRTCCallbacks, type WebRTCManagerOptions, type WebRTCManagerState, + type WebRTCClientCallbacks, } from './webrtc-manager'; +// Re-export core WebRTC types for convenience +export type { WebRTCConnectionState, WebRTCDisconnectReason } from '@agentuity/core'; + // Export client implementation (local to this package) export { createClient } from './client/index'; export type { diff --git a/packages/frontend/src/webrtc-manager.ts b/packages/frontend/src/webrtc-manager.ts index e5f28ddf5..388156fa9 100644 --- a/packages/frontend/src/webrtc-manager.ts +++ b/packages/frontend/src/webrtc-manager.ts @@ -1,22 +1,107 @@ +import type { + SignalMessage, + WebRTCConnectionState, + WebRTCDisconnectReason, +} from '@agentuity/core'; + /** - * WebRTC connection status + * Callbacks for WebRTC client state changes and events. + * All callbacks are optional - only subscribe to events you care about. */ -export type WebRTCStatus = 'disconnected' | 'connecting' | 'signaling' | 'connected'; +export interface WebRTCClientCallbacks { + /** + * Called on every state transition. + * @param from - Previous state + * @param to - New state + * @param reason - Optional reason for the transition + */ + onStateChange?: (from: WebRTCConnectionState, to: WebRTCConnectionState, reason?: string) => void; + + /** + * Called when connection is fully established. + */ + onConnect?: () => void; + + /** + * Called when disconnected from the call. + * @param reason - Why the disconnection happened + */ + onDisconnect?: (reason: WebRTCDisconnectReason) => void; + + /** + * Called when local media stream is acquired. + * @param stream - The local MediaStream + */ + onLocalStream?: (stream: MediaStream) => void; + + /** + * Called when remote media stream is received. + * @param stream - The remote MediaStream + */ + onRemoteStream?: (stream: MediaStream) => void; + + /** + * Called when a new track is added to a stream. + * @param track - The added track + * @param stream - The stream containing the track + */ + onTrackAdded?: (track: MediaStreamTrack, stream: MediaStream) => void; + + /** + * Called when a track is removed from a stream. + * @param track - The removed track + */ + onTrackRemoved?: (track: MediaStreamTrack) => void; + + /** + * Called when a peer joins the room. + * @param peerId - The peer's ID + */ + onPeerJoined?: (peerId: string) => void; + + /** + * Called when a peer leaves the room. + * @param peerId - The peer's ID + */ + onPeerLeft?: (peerId: string) => void; + + /** + * Called when SDP/ICE negotiation starts. + */ + onNegotiationStart?: () => void; + + /** + * Called when SDP/ICE negotiation completes successfully. + */ + onNegotiationComplete?: () => void; + + /** + * Called for each ICE candidate generated. + * @param candidate - The ICE candidate + */ + onIceCandidate?: (candidate: RTCIceCandidateInit) => void; + + /** + * Called when ICE connection state changes. + * @param state - The new ICE connection state + */ + onIceStateChange?: (state: string) => void; + + /** + * Called when an error occurs. + * @param error - The error that occurred + * @param state - The state when the error occurred + */ + onError?: (error: Error, state: WebRTCConnectionState) => void; +} /** - * Signaling message types (must match server protocol) + * @deprecated Use `WebRTCConnectionState` from @agentuity/core instead. */ -type SignalMsg = - | { t: 'join'; roomId: string } - | { t: 'joined'; peerId: string; roomId: string; peers: string[] } - | { t: 'peer-joined'; peerId: string } - | { t: 'peer-left'; peerId: string } - | { t: 'sdp'; from: string; to?: string; description: RTCSessionDescriptionInit } - | { t: 'ice'; from: string; to?: string; candidate: RTCIceCandidateInit } - | { t: 'error'; message: string }; +export type WebRTCStatus = 'disconnected' | 'connecting' | 'signaling' | 'connected'; /** - * Callbacks for WebRTC manager state changes + * @deprecated Use `WebRTCClientCallbacks` from @agentuity/core instead. */ export interface WebRTCCallbacks { onLocalStream?: (stream: MediaStream) => void; @@ -41,19 +126,24 @@ export interface WebRTCManagerOptions { iceServers?: RTCIceServer[]; /** Media constraints for getUserMedia */ media?: MediaStreamConstraints; - /** Callbacks for state changes */ - callbacks?: WebRTCCallbacks; + /** + * Callbacks for state changes and events. + * Supports both legacy WebRTCCallbacks and new WebRTCClientCallbacks. + */ + callbacks?: WebRTCClientCallbacks; } /** * WebRTC manager state */ export interface WebRTCManagerState { - status: WebRTCStatus; + state: WebRTCConnectionState; peerId: string | null; remotePeerId: string | null; isAudioMuted: boolean; isVideoMuted: boolean; + /** @deprecated Use `state` instead */ + status: WebRTCStatus; } /** @@ -64,9 +154,45 @@ const DEFAULT_ICE_SERVERS: RTCIceServer[] = [ { urls: 'stun:stun1.l.google.com:19302' }, ]; +/** + * Map new state to legacy status for backwards compatibility + */ +function stateToStatus(state: WebRTCConnectionState): WebRTCStatus { + if (state === 'idle') return 'disconnected'; + if (state === 'negotiating') return 'connecting'; + return state as WebRTCStatus; +} + /** * Framework-agnostic WebRTC connection manager with signaling, * perfect negotiation, and media stream handling. + * + * Uses an explicit state machine for connection lifecycle: + * - idle: No resources allocated, ready to connect + * - connecting: Acquiring media + opening WebSocket + * - signaling: In room, waiting for peer + * - negotiating: SDP/ICE exchange in progress + * - connected: Media flowing + * + * @example + * ```ts + * const manager = new WebRTCManager({ + * signalUrl: 'wss://example.com/call/signal', + * roomId: 'my-room', + * callbacks: { + * onStateChange: (from, to, reason) => { + * console.log(`State: ${from} → ${to}`, reason); + * }, + * onConnect: () => console.log('Connected!'), + * onDisconnect: (reason) => console.log('Disconnected:', reason), + * onLocalStream: (stream) => { localVideo.srcObject = stream; }, + * onRemoteStream: (stream) => { remoteVideo.srcObject = stream; }, + * onError: (error, state) => console.error(`Error in ${state}:`, error), + * }, + * }); + * + * await manager.connect(); + * ``` */ export class WebRTCManager { private ws: WebSocket | null = null; @@ -76,10 +202,12 @@ export class WebRTCManager { private peerId: string | null = null; private remotePeerId: string | null = null; - private status: WebRTCStatus = 'disconnected'; private isAudioMuted = false; private isVideoMuted = false; + // State machine + private _state: WebRTCConnectionState = 'idle'; + // Perfect negotiation state private makingOffer = false; private ignoreOffer = false; @@ -90,7 +218,7 @@ export class WebRTCManager { private hasRemoteDescription = false; private options: WebRTCManagerOptions; - private callbacks: WebRTCCallbacks; + private callbacks: WebRTCClientCallbacks; constructor(options: WebRTCManagerOptions) { this.options = options; @@ -98,12 +226,20 @@ export class WebRTCManager { this.callbacks = options.callbacks ?? {}; } + /** + * Current connection state + */ + get state(): WebRTCConnectionState { + return this._state; + } + /** * Get current manager state */ getState(): WebRTCManagerState { return { - status: this.status, + state: this._state, + status: stateToStatus(this._state), peerId: this.peerId, remotePeerId: this.remotePeerId, isAudioMuted: this.isAudioMuted, @@ -125,12 +261,42 @@ export class WebRTCManager { return this.remoteStream; } - private setStatus(status: WebRTCStatus): void { - this.status = status; - this.callbacks.onStatusChange?.(status); + /** + * Transition to a new state with callback notifications + */ + private setState(newState: WebRTCConnectionState, reason?: string): void { + const prevState = this._state; + if (prevState === newState) return; + + this._state = newState; + + // Fire state change callback + this.callbacks.onStateChange?.(prevState, newState, reason); + + // Fire connect/disconnect callbacks + if (newState === 'connected' && prevState !== 'connected') { + this.callbacks.onConnect?.(); + this.callbacks.onNegotiationComplete?.(); + } + + if (newState === 'idle' && prevState !== 'idle') { + const disconnectReason = this.mapToDisconnectReason(reason); + this.callbacks.onDisconnect?.(disconnectReason); + } + + if (newState === 'negotiating' && prevState !== 'negotiating') { + this.callbacks.onNegotiationStart?.(); + } + } + + private mapToDisconnectReason(reason?: string): WebRTCDisconnectReason { + if (reason === 'hangup') return 'hangup'; + if (reason === 'peer-left') return 'peer-left'; + if (reason === 'timeout') return 'timeout'; + return 'error'; } - private send(msg: SignalMsg): void { + private send(msg: SignalMessage): void { if (this.ws?.readyState === WebSocket.OPEN) { this.ws.send(JSON.stringify(msg)); } @@ -140,9 +306,9 @@ export class WebRTCManager { * Connect to the signaling server and start the call */ async connect(): Promise { - if (this.status !== 'disconnected') return; + if (this._state !== 'idle') return; - this.setStatus('connecting'); + this.setState('connecting', 'connect() called'); try { // Get local media @@ -154,31 +320,33 @@ export class WebRTCManager { this.ws = new WebSocket(this.options.signalUrl); this.ws.onopen = () => { - this.setStatus('signaling'); + this.setState('signaling', 'WebSocket opened'); this.send({ t: 'join', roomId: this.options.roomId }); }; this.ws.onmessage = (event) => { - const msg = JSON.parse(event.data) as SignalMsg; + const msg = JSON.parse(event.data) as SignalMessage; this.handleSignalingMessage(msg); }; this.ws.onerror = () => { - this.callbacks.onError?.(new Error('WebSocket connection error')); + const error = new Error('WebSocket connection error'); + this.callbacks.onError?.(error, this._state); }; this.ws.onclose = () => { - if (this.status !== 'disconnected') { - this.setStatus('disconnected'); + if (this._state !== 'idle') { + this.setState('idle', 'WebSocket closed'); } }; } catch (err) { - this.setStatus('disconnected'); - this.callbacks.onError?.(err instanceof Error ? err : new Error(String(err))); + const error = err instanceof Error ? err : new Error(String(err)); + this.callbacks.onError?.(error, this._state); + this.setState('idle', 'error'); } } - private async handleSignalingMessage(msg: SignalMsg): Promise { + private async handleSignalingMessage(msg: SignalMessage): Promise { switch (msg.t) { case 'joined': this.peerId = msg.peerId; @@ -188,6 +356,7 @@ export class WebRTCManager { // Late joiner is impolite (makes the offer, wins collisions) this.polite = this.options.polite ?? false; await this.createPeerConnection(); + this.setState('negotiating', 'creating offer'); await this.createOffer(); } else { // First peer is polite (waits for offers, yields on collision) @@ -207,11 +376,14 @@ export class WebRTCManager { if (msg.peerId === this.remotePeerId) { this.remotePeerId = null; this.closePeerConnection(); - this.setStatus('signaling'); + this.setState('signaling', 'peer-left'); } break; case 'sdp': + if (this._state === 'signaling') { + this.setState('negotiating', 'received SDP'); + } await this.handleRemoteSDP(msg.description); break; @@ -220,7 +392,8 @@ export class WebRTCManager { break; case 'error': - this.callbacks.onError?.(new Error(msg.message)); + const error = new Error(msg.message); + this.callbacks.onError?.(error, this._state); break; } } @@ -235,6 +408,7 @@ export class WebRTCManager { if (this.localStream) { for (const track of this.localStream.getTracks()) { this.pc.addTrack(track, this.localStream); + this.callbacks.onTrackAdded?.(track, this.localStream); } } @@ -259,14 +433,17 @@ export class WebRTCManager { } } - if (this.status !== 'connected') { - this.setStatus('connected'); + this.callbacks.onTrackAdded?.(event.track, this.remoteStream!); + + if (this._state !== 'connected') { + this.setState('connected', 'track received'); } }; // Handle ICE candidates this.pc.onicecandidate = (event) => { if (event.candidate) { + this.callbacks.onIceCandidate?.(event.candidate.toJSON()); this.send({ t: 'ice', from: this.peerId!, @@ -288,17 +465,26 @@ export class WebRTCManager { description: this.pc!.localDescription!, }); } catch (err) { - this.callbacks.onError?.(err instanceof Error ? err : new Error(String(err))); + const error = err instanceof Error ? err : new Error(String(err)); + this.callbacks.onError?.(error, this._state); } finally { this.makingOffer = false; } }; this.pc.oniceconnectionstatechange = () => { - if (this.pc?.iceConnectionState === 'disconnected') { - this.setStatus('signaling'); - } else if (this.pc?.iceConnectionState === 'connected') { - this.setStatus('connected'); + const iceState = this.pc?.iceConnectionState; + if (iceState) { + this.callbacks.onIceStateChange?.(iceState); + } + + if (iceState === 'disconnected') { + this.setState('signaling', 'ICE disconnected'); + } else if (iceState === 'connected') { + this.setState('connected', 'ICE connected'); + } else if (iceState === 'failed') { + const error = new Error('ICE connection failed'); + this.callbacks.onError?.(error, this._state); } }; } @@ -401,6 +587,7 @@ export class WebRTCManager { if (this.localStream) { for (const track of this.localStream.getTracks()) { track.stop(); + this.callbacks.onTrackRemoved?.(track); } this.localStream = null; } @@ -412,7 +599,7 @@ export class WebRTCManager { this.peerId = null; this.remotePeerId = null; - this.setStatus('disconnected'); + this.setState('idle', 'hangup'); } /** diff --git a/packages/react/src/index.ts b/packages/react/src/index.ts index e6c2babcf..b683c1a58 100644 --- a/packages/react/src/index.ts +++ b/packages/react/src/index.ts @@ -35,6 +35,8 @@ export { type UseWebRTCCallOptions, type UseWebRTCCallResult, type WebRTCStatus, + type WebRTCConnectionState, + type WebRTCClientCallbacks, } from './webrtc'; export { useAPI, @@ -75,6 +77,7 @@ export { type WebRTCCallbacks, type WebRTCManagerOptions, type WebRTCManagerState, + type WebRTCDisconnectReason, // Client type exports (createClient is exported from ./client.ts) type Client, type ClientOptions, diff --git a/packages/react/src/webrtc.tsx b/packages/react/src/webrtc.tsx index 521958242..dc5d80b24 100644 --- a/packages/react/src/webrtc.tsx +++ b/packages/react/src/webrtc.tsx @@ -1,60 +1,80 @@ import { useCallback, useContext, useEffect, useMemo, useRef, useState } from 'react'; import { - WebRTCManager, - buildUrl, - type WebRTCStatus, - type WebRTCManagerOptions, + WebRTCManager, + buildUrl, + type WebRTCStatus, + type WebRTCManagerOptions, + type WebRTCConnectionState, + type WebRTCClientCallbacks, } from '@agentuity/frontend'; + +export type { WebRTCClientCallbacks }; import { AgentuityContext } from './context'; -export type { WebRTCStatus }; +export type { WebRTCStatus, WebRTCConnectionState }; /** * Options for useWebRTCCall hook */ export interface UseWebRTCCallOptions { - /** Room ID to join */ - roomId: string; - /** WebSocket signaling URL (e.g., '/call/signal' or full URL) */ - signalUrl: string; - /** Whether this peer is "polite" in perfect negotiation (default: true for first joiner) */ - polite?: boolean; - /** ICE servers configuration */ - iceServers?: RTCIceServer[]; - /** Media constraints for getUserMedia */ - media?: MediaStreamConstraints; - /** Whether to auto-connect on mount (default: true) */ - autoConnect?: boolean; + /** Room ID to join */ + roomId: string; + /** WebSocket signaling URL (e.g., '/call/signal' or full URL) */ + signalUrl: string; + /** Whether this peer is "polite" in perfect negotiation (default: true for first joiner) */ + polite?: boolean; + /** ICE servers configuration */ + iceServers?: RTCIceServer[]; + /** Media constraints for getUserMedia */ + media?: MediaStreamConstraints; + /** Whether to auto-connect on mount (default: true) */ + autoConnect?: boolean; + /** + * Optional callbacks for WebRTC events. + * These are called in addition to the hook's internal state management. + */ + callbacks?: Partial; } /** * Return type for useWebRTCCall hook */ export interface UseWebRTCCallResult { - /** Ref to attach to local video element */ - localVideoRef: React.RefObject; - /** Ref to attach to remote video element */ - remoteVideoRef: React.RefObject; - /** Current connection status */ - status: WebRTCStatus; - /** Current error if any */ - error: Error | null; - /** Local peer ID assigned by server */ - peerId: string | null; - /** Remote peer ID */ - remotePeerId: string | null; - /** Whether audio is muted */ - isAudioMuted: boolean; - /** Whether video is muted */ - isVideoMuted: boolean; - /** Manually start the connection (if autoConnect is false) */ - connect: () => void; - /** End the call */ - hangup: () => void; - /** Mute or unmute audio */ - muteAudio: (muted: boolean) => void; - /** Mute or unmute video */ - muteVideo: (muted: boolean) => void; + /** Ref to attach to local video element */ + localVideoRef: React.RefObject; + /** Ref to attach to remote video element */ + remoteVideoRef: React.RefObject; + /** Current connection state (new state machine) */ + state: WebRTCConnectionState; + /** @deprecated Use `state` instead. Current connection status */ + status: WebRTCStatus; + /** Current error if any */ + error: Error | null; + /** Local peer ID assigned by server */ + peerId: string | null; + /** Remote peer ID */ + remotePeerId: string | null; + /** Whether audio is muted */ + isAudioMuted: boolean; + /** Whether video is muted */ + isVideoMuted: boolean; + /** Manually start the connection (if autoConnect is false) */ + connect: () => void; + /** End the call */ + hangup: () => void; + /** Mute or unmute audio */ + muteAudio: (muted: boolean) => void; + /** Mute or unmute video */ + muteVideo: (muted: boolean) => void; +} + +/** + * Map new state to legacy status for backwards compatibility + */ +function stateToStatus(state: WebRTCConnectionState): WebRTCStatus { + if (state === 'idle') return 'disconnected'; + if (state === 'negotiating') return 'connecting'; + return state as WebRTCStatus; } /** @@ -68,20 +88,27 @@ export interface UseWebRTCCallResult { * const { * localVideoRef, * remoteVideoRef, - * status, + * state, * hangup, * muteAudio, * isAudioMuted, * } = useWebRTCCall({ * roomId, * signalUrl: '/call/signal', + * callbacks: { + * onStateChange: (from, to, reason) => { + * console.log(`State: ${from} → ${to}`, reason); + * }, + * onConnect: () => console.log('Connected!'), + * onDisconnect: (reason) => console.log('Disconnected:', reason), + * }, * }); * * return ( *
*
- +

{agentResult}

- +
WebSocket: {connected ? JSON.stringify(wsMessage) : 'Not connected'} @@ -206,7 +208,7 @@ Reference them in your HTML or components: ```html - + ``` @@ -240,7 +242,7 @@ body { Import in `index.html`: ```html - + ``` ### Style Tag in Component diff --git a/docs/webrtc-architecture.md b/docs/webrtc-architecture.md new file mode 100644 index 000000000..2138dec8e --- /dev/null +++ b/docs/webrtc-architecture.md @@ -0,0 +1,156 @@ +# WebRTC Architecture Guide + +This guide explains the WebRTC architecture options and when to use each approach. + +## Current Architecture: Mesh Networking + +The Agentuity SDK uses **mesh networking** for WebRTC connections: + +``` + Peer A + / \ + / \ +Peer B ---- Peer C +``` + +Each peer maintains a direct RTCPeerConnection to every other peer in the room. + +### How It Works + +1. **Signaling Server** (WebRTCRoomManager) handles room membership and message relay +2. **WebRTCManager** creates one RTCPeerConnection per remote peer +3. Each peer sends their media/data directly to all other peers +4. Perfect negotiation handles SDP offer/answer collisions + +### Pros + +- **Simple architecture** - No media server needed +- **Low latency** - Direct peer-to-peer connections +- **Privacy** - Media never touches a central server +- **Cost effective** - Only signaling server required + +### Cons + +- **O(N²) connections** - Each peer connects to all others +- **Bandwidth scales poorly** - Each peer uploads N-1 copies of their stream +- **CPU intensive** - Encoding happens once per peer +- **Unreliable for mobile** - Limited uplink bandwidth + +### Recommended Limits + +| Scenario | Max Peers | +| ------------------- | --------- | +| Audio only | 8-10 | +| Video (low quality) | 4-6 | +| Video (HD) | 3-4 | +| Mobile devices | 2-3 | + +## When to Consider an SFU + +A **Selective Forwarding Unit (SFU)** is a media server that receives streams from each peer and selectively forwards them: + +``` + Peer A + | + v + [ SFU ] + / \ + v v +Peer B Peer C +``` + +### SFU Benefits + +- **O(N) connections** - Each peer connects only to the SFU +- **Bandwidth efficient** - Each peer uploads once, SFU distributes +- **Scalable** - Supports 50+ participants +- **Adaptive bitrate** - SFU can select different quality levels +- **Server-side recording** - Easy to record at the SFU +- **Simulcast support** - Peers send multiple quality levels + +### SFU Drawbacks + +- **Added latency** - Extra hop through the server +- **Server costs** - Media servers require significant resources +- **Complexity** - More infrastructure to manage +- **Privacy concerns** - Media passes through server + +### When to Use SFU + +Consider an SFU when you need: + +- More than 4-6 participants regularly +- Mobile client support with unreliable connections +- Server-side recording or transcription +- Bandwidth adaptation based on network conditions +- Large-scale webinars or broadcasts + +## SFU Options + +### Open Source + +- **[mediasoup](https://mediasoup.org/)** - Highly performant, Node.js based +- **[Janus](https://janus.conf.meetecho.com/)** - Versatile, plugin-based +- **[Pion](https://pion.ly/)** - Go-based, modular + +### Commercial + +- **[LiveKit](https://livekit.io/)** - Open-source SFU with cloud option +- **[Daily.co](https://www.daily.co/)** - Fully managed WebRTC +- **[Twilio Video](https://www.twilio.com/video)** - Enterprise-grade + +## Migrating from Mesh to SFU + +If you outgrow mesh networking, the migration path involves: + +1. **Signaling changes** - Peers negotiate with SFU, not each other +2. **Track model** - Change from peer-to-peer to publish/subscribe +3. **Quality selection** - SFU handles bandwidth adaptation + +The Agentuity SDK's track abstraction and stats APIs remain useful: + +```typescript +// Current mesh approach +const manager = new WebRTCManager({ + signalUrl: 'wss://example.com/signal', + roomId: 'my-room', +}); + +// Future SFU approach (conceptual) +const manager = new SFUManager({ + sfuUrl: 'wss://sfu.example.com/room/my-room', + token: 'auth-token', +}); + +// Same APIs work for both: +await manager.startScreenShare(); +const stats = await manager.getQualitySummary(trackId); +manager.startRecording('local'); +``` + +## Hybrid Approaches + +For flexibility, consider: + +### MCU for Recording + Mesh for Live + +- Use mesh for low-latency live communication +- Connect an MCU (Multipoint Control Unit) for server-side recording/compositing + +### SFU with Cascading + +- Deploy SFUs in multiple regions +- Cascade media between SFUs for global scale + +## Summary + +| Feature | Mesh | SFU | +| ------------------------- | -------------------- | ----------------------- | +| Max participants | 4-6 | 50+ | +| Server cost | Low (signaling only) | High (media processing) | +| Latency | Lowest | Low-Medium | +| Bandwidth efficiency | Poor | Good | +| Server-side features | Limited | Full (recording, etc.) | +| Implementation complexity | Simple | Moderate-High | + +**Start with mesh** (current SDK) for small groups. Migrate to SFU when you consistently need more than 4-6 participants or require server-side features like recording and transcription. diff --git a/docs/webrtc-turn-configuration.md b/docs/webrtc-turn-configuration.md new file mode 100644 index 000000000..8add114a1 --- /dev/null +++ b/docs/webrtc-turn-configuration.md @@ -0,0 +1,244 @@ +# TURN Server Configuration Guide + +This guide explains how to configure TURN servers for WebRTC connections in your Agentuity applications. + +## When is TURN Required? + +WebRTC uses ICE (Interactive Connectivity Establishment) to find the best path between peers. The connection types, in order of preference: + +1. **Host** - Direct connection (same network) +2. **Server Reflexive (srflx)** - Connection via STUN (works through most NATs) +3. **Relay** - Connection through TURN server (guaranteed to work) + +TURN is required when: + +- Peers are behind **symmetric NAT** (common in corporate networks) +- **Firewall rules** block UDP traffic +- **Enterprise environments** with restrictive network policies +- **Mobile networks** with carrier-grade NAT + +## ICE Server Configuration + +### Default Configuration (STUN Only) + +By default, WebRTCManager uses public Google STUN servers: + +```typescript +const manager = new WebRTCManager({ + signalUrl: 'wss://example.com/signal', + roomId: 'my-room', + // Uses default STUN servers +}); +``` + +### Adding TURN Servers + +For production use, configure both STUN and TURN: + +```typescript +const manager = new WebRTCManager({ + signalUrl: 'wss://example.com/signal', + roomId: 'my-room', + iceServers: [ + // STUN server + { urls: 'stun:stun.example.com:3478' }, + + // TURN server with UDP (fastest, when allowed) + { + urls: 'turn:turn.example.com:3478?transport=udp', + username: 'user', + credential: 'password', + }, + + // TURN server with TCP (fallback when UDP is blocked) + { + urls: 'turn:turn.example.com:3478?transport=tcp', + username: 'user', + credential: 'password', + }, + + // TURN over TLS on port 443 (works through most firewalls) + { + urls: 'turns:turn.example.com:443?transport=tcp', + username: 'user', + credential: 'password', + }, + ], +}); +``` + +### Recommended Configuration for Maximum Compatibility + +```typescript +const iceServers: RTCIceServer[] = [ + { urls: 'stun:stun.example.com:3478' }, + // TURN UDP - best performance + { + urls: 'turn:turn.example.com:3478?transport=udp', + username: credentials.username, + credential: credentials.password, + }, + // TURN TCP - fallback + { + urls: 'turn:turn.example.com:3478?transport=tcp', + username: credentials.username, + credential: credentials.password, + }, + // TURNS (TLS) on 443 - works through most firewalls + { + urls: 'turns:turn.example.com:443?transport=tcp', + username: credentials.username, + credential: credentials.password, + }, +]; +``` + +## Credential Management + +### Long-Term Credentials + +Simple but less secure. Credentials are static: + +```typescript +{ + urls: 'turn:turn.example.com:3478', + username: 'static-user', + credential: 'static-password', +} +``` + +### Time-Limited Credentials (Recommended) + +Generate short-lived credentials from your server: + +```typescript +// Server-side (e.g., in your API) +function generateTurnCredentials(userId: string): { username: string; credential: string } { + const ttl = 86400; // 24 hours + const timestamp = Math.floor(Date.now() / 1000) + ttl; + const username = `${timestamp}:${userId}`; + + // HMAC-SHA1 with your TURN shared secret + const hmac = crypto.createHmac('sha1', TURN_SECRET); + hmac.update(username); + const credential = hmac.digest('base64'); + + return { username, credential }; +} + +// Client-side +const credentials = await fetch('/api/turn-credentials').then((r) => r.json()); + +const manager = new WebRTCManager({ + signalUrl: 'wss://example.com/signal', + roomId: 'my-room', + iceServers: [ + { urls: 'stun:stun.example.com:3478' }, + { + urls: 'turn:turn.example.com:3478', + username: credentials.username, + credential: credentials.credential, + }, + ], +}); +``` + +## Setting Up coturn + +[coturn](https://github.com/coturn/coturn) is the most popular open-source TURN server. + +### Basic coturn Configuration + +```conf +# /etc/turnserver.conf + +# Network +listening-port=3478 +tls-listening-port=5349 + +# Use your actual external IP +external-ip=203.0.113.1 + +# Domain +realm=example.com + +# Authentication +lt-cred-mech +user=username:password + +# Or for time-limited credentials +use-auth-secret +static-auth-secret=your-secret-here + +# TLS (for TURNS) +cert=/etc/ssl/certs/turn.example.com.pem +pkey=/etc/ssl/private/turn.example.com.key + +# Logging +log-file=/var/log/turnserver.log +verbose +``` + +### Running coturn with Docker + +```bash +docker run -d \ + --name coturn \ + --network host \ + coturn/coturn \ + -n \ + --listening-port=3478 \ + --tls-listening-port=5349 \ + --external-ip='$(detect-external-ip)' \ + --realm=example.com \ + --use-auth-secret \ + --static-auth-secret=your-secret-here \ + --cert=/etc/ssl/certs/turn.pem \ + --pkey=/etc/ssl/private/turn.key +``` + +## Verifying TURN Usage + +Use the connection stats API to verify TURN is being used: + +```typescript +const summary = await manager.getQualitySummary(peerId); + +if (summary?.candidatePair?.usingRelay) { + console.log('Connection is using TURN relay'); + console.log('Local candidate type:', summary.candidatePair.localType); + console.log('Remote candidate type:', summary.candidatePair.remoteType); +} +``` + +## Hosted TURN Services + +If you don't want to run your own TURN server: + +- [Twilio STUN/TURN](https://www.twilio.com/stun-turn) - Pay-per-use +- [Xirsys](https://xirsys.com/) - TURN-as-a-service +- [Metered](https://www.metered.ca/stun-turn) - STUN/TURN service + +## Troubleshooting + +### Connection Fails in Corporate Networks + +1. Ensure TURNS on port 443 is configured +2. Check that your TURN server allows TCP transport +3. Verify TLS certificates are valid + +### High Latency When Using TURN + +- TURN adds latency by design (relay through server) +- Ensure TURN server is geographically close to users +- Consider deploying multiple TURN servers in different regions + +### Credentials Rejected + +- For time-limited credentials, ensure server time is synchronized +- Verify the HMAC secret matches between server and coturn +- Check that credentials haven't expired + +### Testing TURN Connectivity + +Use [Trickle ICE](https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/) to test your TURN server configuration before deploying. diff --git a/e2e/webrtc.pw.ts b/e2e/webrtc.pw.ts new file mode 100644 index 000000000..dd15d7100 --- /dev/null +++ b/e2e/webrtc.pw.ts @@ -0,0 +1,315 @@ +import { test, expect, type Page, type BrowserContext } from '@playwright/test'; + +async function waitForPageLoad(page: Page) { + await expect(page.locator('h1')).toContainText('WebRTC', { timeout: 10000 }); +} + +test.describe('WebRTC Data Channels', () => { + test.describe('Single Peer', () => { + test('should connect to signaling server and reach signaling state', async ({ page }) => { + await page.goto('/webrtc'); + await waitForPageLoad(page); + + // Verify initial state + await expect(page.getByTestId('connection-state')).toContainText('idle'); + + // Connect + await page.getByTestId('connect-btn').click(); + + // Should transition to signaling (waiting for peer) + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Should have peer ID assigned + await expect(page.getByTestId('peer-id')).not.toContainText('N/A', { timeout: 5000 }); + }); + + test('should disconnect cleanly', async ({ page }) => { + await page.goto('/webrtc'); + await waitForPageLoad(page); + + // Connect + await page.getByTestId('connect-btn').click(); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Disconnect + await page.getByTestId('disconnect-btn').click(); + + // Should return to idle + await expect(page.getByTestId('connection-state')).toContainText('idle'); + await expect(page.getByTestId('peer-id')).toContainText('N/A'); + }); + + test('should join custom room', async ({ page }) => { + await page.goto('/webrtc'); + await waitForPageLoad(page); + + // Set custom room ID + const roomInput = page.getByTestId('room-id-input'); + await roomInput.clear(); + await roomInput.fill('custom-room-123'); + + // Connect + await page.getByTestId('connect-btn').click(); + + // Should connect successfully + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + }); + }); + + test.describe('Two Peers', () => { + let context1: BrowserContext; + let context2: BrowserContext; + let page1: Page; + let page2: Page; + + test.beforeEach(async ({ browser }) => { + // Create two separate browser contexts for two peers + context1 = await browser.newContext(); + context2 = await browser.newContext(); + page1 = await context1.newPage(); + page2 = await context2.newPage(); + }); + + test.afterEach(async () => { + await context1.close(); + await context2.close(); + }); + + test('should establish peer connection between two browsers', async () => { + const roomId = `test-room-${Date.now()}`; + + // Navigate both pages + await Promise.all([page1.goto('/webrtc'), page2.goto('/webrtc')]); + await Promise.all([waitForPageLoad(page1), waitForPageLoad(page2)]); + + // Set same room ID for both + await page1.getByTestId('room-id-input').clear(); + await page1.getByTestId('room-id-input').fill(roomId); + await page2.getByTestId('room-id-input').clear(); + await page2.getByTestId('room-id-input').fill(roomId); + + // Connect first peer + await page1.getByTestId('connect-btn').click(); + await expect(page1.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Connect second peer + await page2.getByTestId('connect-btn').click(); + + // Both should eventually reach connected state + await expect(page1.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + await expect(page2.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + + // Both should have peer IDs assigned + await expect(page1.getByTestId('peer-id')).not.toContainText('N/A', { + timeout: 5000, + }); + await expect(page2.getByTestId('peer-id')).not.toContainText('N/A', { + timeout: 5000, + }); + }); + + test('should open data channel between peers', async () => { + const roomId = `data-channel-${Date.now()}`; + + await Promise.all([page1.goto('/webrtc'), page2.goto('/webrtc')]); + await Promise.all([waitForPageLoad(page1), waitForPageLoad(page2)]); + + // Set same room ID + await page1.getByTestId('room-id-input').clear(); + await page1.getByTestId('room-id-input').fill(roomId); + await page2.getByTestId('room-id-input').clear(); + await page2.getByTestId('room-id-input').fill(roomId); + + // Connect both peers + await page1.getByTestId('connect-btn').click(); + await page2.getByTestId('connect-btn').click(); + + // Wait for connection + await expect(page1.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + await expect(page2.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + + // Data channel should be open on both + await expect(page1.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 5000, + }); + await expect(page2.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 5000, + }); + }); + + test('should send and receive string messages', async () => { + const roomId = `messaging-${Date.now()}`; + + await Promise.all([page1.goto('/webrtc'), page2.goto('/webrtc')]); + await Promise.all([waitForPageLoad(page1), waitForPageLoad(page2)]); + + // Set same room ID and connect + await page1.getByTestId('room-id-input').clear(); + await page1.getByTestId('room-id-input').fill(roomId); + await page2.getByTestId('room-id-input').clear(); + await page2.getByTestId('room-id-input').fill(roomId); + + await page1.getByTestId('connect-btn').click(); + await page2.getByTestId('connect-btn').click(); + + // Wait for data channel to open + await expect(page1.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + await expect(page2.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + + // Send message from peer 1 to peer 2 + await page1.getByTestId('message-input').fill('Hello from peer 1!'); + await page1.getByTestId('send-btn').click(); + + // Peer 2 should receive the message + await expect(page2.getByTestId('message-remote').first()).toContainText( + 'Hello from peer 1!', + { timeout: 5000 } + ); + + // Peer 1 should see their own message as local + await expect(page1.getByTestId('message-local').first()).toContainText( + 'Hello from peer 1!' + ); + + // Send message from peer 2 to peer 1 + await page2.getByTestId('message-input').fill('Hello from peer 2!'); + await page2.getByTestId('send-btn').click(); + + // Peer 1 should receive the message + await expect(page1.getByTestId('message-remote').first()).toContainText( + 'Hello from peer 2!', + { timeout: 5000 } + ); + }); + + test('should send and receive JSON messages', async () => { + const roomId = `json-test-${Date.now()}`; + + await Promise.all([page1.goto('/webrtc'), page2.goto('/webrtc')]); + await Promise.all([waitForPageLoad(page1), waitForPageLoad(page2)]); + + await page1.getByTestId('room-id-input').clear(); + await page1.getByTestId('room-id-input').fill(roomId); + await page2.getByTestId('room-id-input').clear(); + await page2.getByTestId('room-id-input').fill(roomId); + + await page1.getByTestId('connect-btn').click(); + await page2.getByTestId('connect-btn').click(); + + // Wait for data channel + await expect(page1.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + await expect(page2.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + + // Send JSON from peer 1 + await page1.getByTestId('send-json-btn').click(); + + // Peer 2 should receive JSON with ping type + await expect(page2.getByTestId('message-remote').first()).toContainText('ping', { + timeout: 5000, + }); + }); + + test('should handle peer disconnect gracefully', async () => { + const roomId = `disconnect-${Date.now()}`; + + await Promise.all([page1.goto('/webrtc'), page2.goto('/webrtc')]); + await Promise.all([waitForPageLoad(page1), waitForPageLoad(page2)]); + + await page1.getByTestId('room-id-input').clear(); + await page1.getByTestId('room-id-input').fill(roomId); + await page2.getByTestId('room-id-input').clear(); + await page2.getByTestId('room-id-input').fill(roomId); + + await page1.getByTestId('connect-btn').click(); + await page2.getByTestId('connect-btn').click(); + + // Wait for connection + await expect(page1.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + await expect(page2.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + + // Peer 2 disconnects + await page2.getByTestId('disconnect-btn').click(); + + // Peer 1 should detect the disconnect and go back to signaling + await expect(page1.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Peer 1's remote peer ID should be cleared or show waiting + await expect(page1.getByTestId('data-channel-state')).toContainText('Closed', { + timeout: 5000, + }); + }); + + test('should handle room rejoin after disconnect', async () => { + const roomId = `rejoin-${Date.now()}`; + + await Promise.all([page1.goto('/webrtc'), page2.goto('/webrtc')]); + await Promise.all([waitForPageLoad(page1), waitForPageLoad(page2)]); + + await page1.getByTestId('room-id-input').clear(); + await page1.getByTestId('room-id-input').fill(roomId); + await page2.getByTestId('room-id-input').clear(); + await page2.getByTestId('room-id-input').fill(roomId); + + // First connection + await page1.getByTestId('connect-btn').click(); + await page2.getByTestId('connect-btn').click(); + + await expect(page1.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + + // Peer 2 disconnects + await page2.getByTestId('disconnect-btn').click(); + await expect(page2.getByTestId('connection-state')).toContainText('idle'); + + // Peer 2 rejoins + await page2.getByTestId('connect-btn').click(); + + // Both should reconnect + await expect(page1.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + await expect(page2.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + + // Data channel should work again + await expect(page1.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 5000, + }); + await expect(page2.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 5000, + }); + }); + }); +}); diff --git a/packages/cli/src/cli.ts b/packages/cli/src/cli.ts index ac79d7b01..17f76df26 100644 --- a/packages/cli/src/cli.ts +++ b/packages/cli/src/cli.ts @@ -1533,7 +1533,11 @@ async function registerSubcommand( !normalized.requiresRegion && (options as Record).register === false; - if ((normalized.requiresRegion || normalized.optionalRegion) && ctx.apiClient && !skipRegion) { + if ( + (normalized.requiresRegion || normalized.optionalRegion) && + ctx.apiClient && + !skipRegion + ) { const apiClient: APIClientType = ctx.apiClient as APIClientType; const regions = await tui.spinner({ message: 'Fetching cloud regions', diff --git a/packages/cli/src/cmd/cloud/env/delete.ts b/packages/cli/src/cmd/cloud/env/delete.ts index afce9b778..fba6a3d8e 100644 --- a/packages/cli/src/cmd/cloud/env/delete.ts +++ b/packages/cli/src/cmd/cloud/env/delete.ts @@ -16,7 +16,10 @@ import { resolveOrgId, isOrgScope } from './org-util'; const EnvDeleteResponseSchema = z.object({ success: z.boolean().describe('Whether the operation succeeded'), key: z.string().describe('Variable key that was deleted'), - path: z.string().optional().describe('Local file path where variable was removed (project scope only)'), + path: z + .string() + .optional() + .describe('Local file path where variable was removed (project scope only)'), secret: z.boolean().describe('Whether a secret was deleted'), scope: z.enum(['project', 'org']).describe('The scope from which the variable was deleted'), }); @@ -30,7 +33,10 @@ export const deleteSubcommand = createSubcommand({ examples: [ { command: getCommand('env delete OLD_FEATURE_FLAG'), description: 'Delete variable' }, { command: getCommand('env rm API_KEY'), description: 'Delete a secret' }, - { command: getCommand('env rm OPENAI_API_KEY --org'), description: 'Delete org-level secret' }, + { + command: getCommand('env rm OPENAI_API_KEY --org'), + description: 'Delete org-level secret', + }, ], requires: { auth: true, apiClient: true }, optional: { project: true }, @@ -42,7 +48,9 @@ export const deleteSubcommand = createSubcommand({ org: z .union([z.boolean(), z.string()]) .optional() - .describe('delete from organization level (use --org for default org, or --org for specific org)'), + .describe( + 'delete from organization level (use --org for default org, or --org for specific org)' + ), }), response: EnvDeleteResponseSchema, }, @@ -53,7 +61,9 @@ export const deleteSubcommand = createSubcommand({ // Require project context if not using org scope if (!useOrgScope && !project) { - tui.fatal('Project context required. Run from a project directory or use --org for organization scope.'); + tui.fatal( + 'Project context required. Run from a project directory or use --org for organization scope.' + ); } // Validate key doesn't start with reserved AGENTUITY_ prefix (except AGENTUITY_PUBLIC_) @@ -76,7 +86,10 @@ export const deleteSubcommand = createSubcommand({ const isEnv = orgData.env?.[args.key] !== undefined; if (!isSecret && !isEnv) { - tui.fatal(`Variable '${args.key}' not found in organization`, ErrorCode.RESOURCE_NOT_FOUND); + tui.fatal( + `Variable '${args.key}' not found in organization`, + ErrorCode.RESOURCE_NOT_FOUND + ); } // Delete from cloud diff --git a/packages/cli/src/cmd/cloud/env/get.ts b/packages/cli/src/cmd/cloud/env/get.ts index 7839757ed..141469cbc 100644 --- a/packages/cli/src/cmd/cloud/env/get.ts +++ b/packages/cli/src/cmd/cloud/env/get.ts @@ -21,7 +21,10 @@ export const getSubcommand = createSubcommand({ { command: getCommand('env get NODE_ENV'), description: 'Get environment variable' }, { command: getCommand('env get API_KEY'), description: 'Get a secret value' }, { command: getCommand('env get API_KEY --no-mask'), description: 'Show unmasked value' }, - { command: getCommand('env get OPENAI_API_KEY --org'), description: 'Get org-level variable' }, + { + command: getCommand('env get OPENAI_API_KEY --org'), + description: 'Get org-level variable', + }, ], requires: { auth: true, apiClient: true }, optional: { project: true }, @@ -34,7 +37,9 @@ export const getSubcommand = createSubcommand({ org: z .union([z.boolean(), z.string()]) .optional() - .describe('get from organization level (use --org for default org, or --org for specific org)'), + .describe( + 'get from organization level (use --org for default org, or --org for specific org)' + ), }), response: EnvGetResponseSchema, }, @@ -46,7 +51,9 @@ export const getSubcommand = createSubcommand({ // Require project context if not using org scope if (!useOrgScope && !project) { - tui.fatal('Project context required. Run from a project directory or use --org for organization scope.'); + tui.fatal( + 'Project context required. Run from a project directory or use --org for organization scope.' + ); } let value: string | undefined; diff --git a/packages/cli/src/cmd/cloud/env/import.ts b/packages/cli/src/cmd/cloud/env/import.ts index 38d713a47..7d4e45e6c 100644 --- a/packages/cli/src/cmd/cloud/env/import.ts +++ b/packages/cli/src/cmd/cloud/env/import.ts @@ -21,7 +21,10 @@ const EnvImportResponseSchema = z.object({ envCount: z.number().describe('Number of env vars imported'), secretCount: z.number().describe('Number of secrets imported'), skipped: z.number().describe('Number of items skipped'), - path: z.string().optional().describe('Local file path where variables were saved (project scope only)'), + path: z + .string() + .optional() + .describe('Local file path where variables were saved (project scope only)'), file: z.string().describe('Source file path'), scope: z.enum(['project', 'org']).describe('The scope where variables were imported'), }); @@ -29,13 +32,7 @@ const EnvImportResponseSchema = z.object({ export const importSubcommand = createSubcommand({ name: 'import', description: 'Import environment variables and secrets from a file to cloud and local .env', - tags: [ - 'mutating', - 'creates-resource', - 'slow', - 'api-intensive', - 'requires-auth', - ], + tags: ['mutating', 'creates-resource', 'slow', 'api-intensive', 'requires-auth'], examples: [ { command: getCommand('cloud env import .env.backup'), @@ -72,7 +69,9 @@ export const importSubcommand = createSubcommand({ // Require project context if not using org scope if (!useOrgScope && !project) { - tui.fatal('Project context required. Run from a project directory or use --org for organization scope.'); + tui.fatal( + 'Project context required. Run from a project directory or use --org for organization scope.' + ); } // Read the import file @@ -87,7 +86,7 @@ export const importSubcommand = createSubcommand({ secretCount: 0, skipped: 0, file: args.file, - scope: useOrgScope ? 'org' as const : 'project' as const, + scope: useOrgScope ? ('org' as const) : ('project' as const), }; } @@ -103,7 +102,7 @@ export const importSubcommand = createSubcommand({ secretCount: 0, skipped: Object.keys(importedVars).length, file: args.file, - scope: useOrgScope ? 'org' as const : 'project' as const, + scope: useOrgScope ? ('org' as const) : ('project' as const), }; } diff --git a/packages/cli/src/cmd/cloud/env/list.ts b/packages/cli/src/cmd/cloud/env/list.ts index 63323179a..89c456002 100644 --- a/packages/cli/src/cmd/cloud/env/list.ts +++ b/packages/cli/src/cmd/cloud/env/list.ts @@ -23,7 +23,10 @@ export const listSubcommand = createSubcommand({ { command: getCommand('env list --no-mask'), description: 'Show unmasked values' }, { command: getCommand('env list --secrets'), description: 'List only secrets' }, { command: getCommand('env list --env-only'), description: 'List only env vars' }, - { command: getCommand('env list --org'), description: 'List only organization-level variables' }, + { + command: getCommand('env list --org'), + description: 'List only organization-level variables', + }, ], requires: { auth: true, apiClient: true }, optional: { project: true }, @@ -48,7 +51,9 @@ export const listSubcommand = createSubcommand({ if (ctx.opts?.org) { return `/settings/organization/env`; } - return ctx.project ? `/projects/${encodeURIComponent(ctx.project.projectId)}/settings` : undefined; + return ctx.project + ? `/projects/${encodeURIComponent(ctx.project.projectId)}/settings` + : undefined; }, async handler(ctx) { @@ -56,7 +61,8 @@ export const listSubcommand = createSubcommand({ const useOrgScope = isOrgScope(opts?.org); // Build combined result with type info and scope - const result: Record = {}; + const result: Record = + {}; // Filter based on options const showEnv = !opts?.secrets; @@ -132,7 +138,9 @@ export const listSubcommand = createSubcommand({ } } } else { - tui.fatal('Project context required. Run from a project directory or use --org for organization scope.'); + tui.fatal( + 'Project context required. Run from a project directory or use --org for organization scope.' + ); } // Skip TUI output in JSON mode @@ -142,10 +150,10 @@ export const listSubcommand = createSubcommand({ } else { tui.newline(); - const projectCount = Object.values(result).filter(v => v.scope === 'project').length; - const orgCount = Object.values(result).filter(v => v.scope === 'org').length; - const secretCount = Object.values(result).filter(v => v.secret).length; - const envCount = Object.values(result).filter(v => !v.secret).length; + const projectCount = Object.values(result).filter((v) => v.scope === 'project').length; + const orgCount = Object.values(result).filter((v) => v.scope === 'org').length; + const secretCount = Object.values(result).filter((v) => v.secret).length; + const envCount = Object.values(result).filter((v) => !v.secret).length; const parts: string[] = []; if (envCount > 0) parts.push(`${envCount} env`); @@ -165,7 +173,9 @@ export const listSubcommand = createSubcommand({ const displayValue = shouldMask && secret ? tui.maskSecret(value) : value; const typeIndicator = secret ? ' [secret]' : ''; const scopeIndicator = !useOrgScope ? ` [${scope}]` : ''; - console.log(`${tui.bold(key)}=${displayValue}${tui.muted(typeIndicator + scopeIndicator)}`); + console.log( + `${tui.bold(key)}=${displayValue}${tui.muted(typeIndicator + scopeIndicator)}` + ); } } } diff --git a/packages/cli/src/cmd/cloud/env/org-util.ts b/packages/cli/src/cmd/cloud/env/org-util.ts index f9d65713d..fda027acb 100644 --- a/packages/cli/src/cmd/cloud/env/org-util.ts +++ b/packages/cli/src/cmd/cloud/env/org-util.ts @@ -5,7 +5,7 @@ import { listOrganizations } from '@agentuity/server'; /** * Resolves the organization ID for org-scoped env operations. - * + * * @param apiClient - The API client * @param config - The CLI config (may be null) * @param orgOption - The --org option value (true for default/prompt, or explicit org ID) diff --git a/packages/cli/src/cmd/cloud/env/pull.ts b/packages/cli/src/cmd/cloud/env/pull.ts index 3f8ab9125..e5149a731 100644 --- a/packages/cli/src/cmd/cloud/env/pull.ts +++ b/packages/cli/src/cmd/cloud/env/pull.ts @@ -61,16 +61,21 @@ export const pullSubcommand = createSubcommand({ // Organization scope const orgId = await resolveOrgId(apiClient, config, opts!.org!); - const orgData = await tui.spinner('Pulling environment variables from organization', () => { - return orgEnvGet(apiClient, { id: orgId, mask: false }); - }); + const orgData = await tui.spinner( + 'Pulling environment variables from organization', + () => { + return orgEnvGet(apiClient, { id: orgId, mask: false }); + } + ); cloudEnv = { ...orgData.env, ...orgData.secrets }; scope = 'org'; } else { // Project scope if (!project) { - tui.fatal('Project context required. Run from a project directory or use --org for organization scope.'); + tui.fatal( + 'Project context required. Run from a project directory or use --org for organization scope.' + ); } const projectData = await tui.spinner('Pulling environment variables from cloud', () => { diff --git a/packages/cli/src/cmd/cloud/env/push.ts b/packages/cli/src/cmd/cloud/env/push.ts index 66591fc46..36f19d91b 100644 --- a/packages/cli/src/cmd/cloud/env/push.ts +++ b/packages/cli/src/cmd/cloud/env/push.ts @@ -24,13 +24,7 @@ const EnvPushResponseSchema = z.object({ export const pushSubcommand = createSubcommand({ name: 'push', description: 'Push environment variables and secrets from local .env file to cloud', - tags: [ - 'mutating', - 'updates-resource', - 'slow', - 'api-intensive', - 'requires-auth', - ], + tags: ['mutating', 'updates-resource', 'slow', 'api-intensive', 'requires-auth'], idempotent: true, examples: [ { command: getCommand('env push'), description: 'Push all variables to cloud (project)' }, @@ -73,7 +67,7 @@ export const pushSubcommand = createSubcommand({ envCount: 0, secretCount: 0, source: envFilePath, - scope: useOrgScope ? 'org' as const : 'project' as const, + scope: useOrgScope ? ('org' as const) : ('project' as const), }; } @@ -123,7 +117,9 @@ export const pushSubcommand = createSubcommand({ } else { // Project scope (existing behavior) if (!project) { - tui.fatal('Project context required. Run from a project directory or use --org for organization scope.'); + tui.fatal( + 'Project context required. Run from a project directory or use --org for organization scope.' + ); } await tui.spinner('Pushing variables to cloud', () => { diff --git a/packages/cli/src/cmd/cloud/env/set.ts b/packages/cli/src/cmd/cloud/env/set.ts index 30919c2ae..be31f386d 100644 --- a/packages/cli/src/cmd/cloud/env/set.ts +++ b/packages/cli/src/cmd/cloud/env/set.ts @@ -18,7 +18,10 @@ import { resolveOrgId, isOrgScope } from './org-util'; const EnvSetResponseSchema = z.object({ success: z.boolean().describe('Whether the operation succeeded'), key: z.string().describe('Environment variable key'), - path: z.string().optional().describe('Local file path where env var was saved (project scope only)'), + path: z + .string() + .optional() + .describe('Local file path where env var was saved (project scope only)'), secret: z.boolean().describe('Whether the value was stored as a secret'), scope: z.enum(['project', 'org']).describe('The scope where the variable was set'), }); @@ -58,7 +61,9 @@ export const setSubcommand = createSubcommand({ org: z .union([z.boolean(), z.string()]) .optional() - .describe('set at organization level (use --org for default org, or --org for specific org)'), + .describe( + 'set at organization level (use --org for default org, or --org for specific org)' + ), }), response: EnvSetResponseSchema, }, @@ -69,7 +74,9 @@ export const setSubcommand = createSubcommand({ // Require project context if not using org scope if (!useOrgScope && !project) { - tui.fatal('Project context required. Run from a project directory or use --org for organization scope.'); + tui.fatal( + 'Project context required. Run from a project directory or use --org for organization scope.' + ); } let isSecret = opts?.secret ?? false; diff --git a/packages/cli/src/cmd/project/template-flow.ts b/packages/cli/src/cmd/project/template-flow.ts index b8c5028f4..99e755d4a 100644 --- a/packages/cli/src/cmd/project/template-flow.ts +++ b/packages/cli/src/cmd/project/template-flow.ts @@ -337,9 +337,7 @@ export async function runCreateFlow(options: CreateFlowOptions): Promise d.name).join(', ') || '(none)'}` - ); + logger.debug(`Database names: ${resources.db.map((d) => d.name).join(', ') || '(none)'}`); logger.debug( `Storage buckets: ${resources.s3.map((b) => b.bucket_name).join(', ') || '(none)'}` ); diff --git a/packages/cli/src/schema-parser.ts b/packages/cli/src/schema-parser.ts index 457fdb7f7..324b8453a 100644 --- a/packages/cli/src/schema-parser.ts +++ b/packages/cli/src/schema-parser.ts @@ -85,12 +85,15 @@ function isBooleanStringUnion(schema: unknown): boolean { // Zod 3: type is _def.typeName const optUnknown = opt as unknown as Record; const optDef = optUnknown?._def as Record | undefined; - const optType = (optUnknown?.type as string) || (optDef?.typeName as string) || (optDef?.type as string); + const optType = + (optUnknown?.type as string) || (optDef?.typeName as string) || (optDef?.type as string); types.add(optType); } - return (types.has('boolean') || types.has('ZodBoolean')) && - (types.has('string') || types.has('ZodString')); + return ( + (types.has('boolean') || types.has('ZodBoolean')) && + (types.has('string') || types.has('ZodString')) + ); } function getShape(schema: ZodType): Record { diff --git a/packages/cli/src/tui.ts b/packages/cli/src/tui.ts index 62316b257..5bca06def 100644 --- a/packages/cli/src/tui.ts +++ b/packages/cli/src/tui.ts @@ -834,7 +834,9 @@ export function showLoggedOutMessage(appBaseUrl: string, hasProfile = false): vo // Box format: "║ " + content + "║" = 48 chars total // Content area = 46 chars, with leading space = 45 chars for URL + padding const urlPadding = Math.max(0, 45 - signupURL.length); - const showNewLine = showInline ? '' : `║ ${RESET}${link(signupURL)}${YELLOW}${' '.repeat(urlPadding)}║`; + const showNewLine = showInline + ? '' + : `║ ${RESET}${link(signupURL)}${YELLOW}${' '.repeat(urlPadding)}║`; const lines = [ '╔══════════════════════════════════════════════╗', diff --git a/packages/cli/src/types.ts b/packages/cli/src/types.ts index 2d36a2d4b..0ef829318 100644 --- a/packages/cli/src/types.ts +++ b/packages/cli/src/types.ts @@ -62,7 +62,6 @@ export const ConfigSchema = zod.object({ }) .optional() .describe('the gravity client information'), - }); export type Config = zod.infer; diff --git a/packages/cli/test/schema-parser.test.ts b/packages/cli/test/schema-parser.test.ts index 7ed166acf..06af5de9e 100644 --- a/packages/cli/test/schema-parser.test.ts +++ b/packages/cli/test/schema-parser.test.ts @@ -6,10 +6,7 @@ describe('parseOptionsSchema', () => { describe('optionalString type detection', () => { test('detects z.union([z.boolean(), z.string()]) as optionalString', () => { const schema = z.object({ - org: z - .union([z.boolean(), z.string()]) - .optional() - .describe('organization flag'), + org: z.union([z.boolean(), z.string()]).optional().describe('organization flag'), }); const parsed = parseOptionsSchema(schema); diff --git a/packages/core/src/index.ts b/packages/core/src/index.ts index 003b639b5..01d3ccb7d 100644 --- a/packages/core/src/index.ts +++ b/packages/core/src/index.ts @@ -132,7 +132,15 @@ export type { SignalMsg, WebRTCConnectionState, WebRTCDisconnectReason, + DataChannelConfig, + DataChannelMessage, + DataChannelState, WebRTCSignalingCallbacks, + ConnectionQualitySummary, + RecordingOptions, + RecordingHandle, + RecordingState, + TrackSource, } from './webrtc'; // Client code moved to @agentuity/frontend for better bundler compatibility diff --git a/packages/core/src/webrtc.ts b/packages/core/src/webrtc.ts index 7cca51a13..f9343f1cd 100644 --- a/packages/core/src/webrtc.ts +++ b/packages/core/src/webrtc.ts @@ -70,13 +70,140 @@ export type SignalMsg = SignalMessage; * - connected → signaling: peer left * - connected → idle: hangup or WebSocket closed */ -export type WebRTCConnectionState = 'idle' | 'connecting' | 'signaling' | 'negotiating' | 'connected'; +export type WebRTCConnectionState = + | 'idle' + | 'connecting' + | 'signaling' + | 'negotiating' + | 'connected'; /** * Reasons for disconnection. */ export type WebRTCDisconnectReason = 'hangup' | 'error' | 'peer-left' | 'timeout'; +// ============================================================================= +// Data Channel Types +// ============================================================================= + +/** + * Configuration for creating a data channel. + */ +export interface DataChannelConfig { + /** Unique label for the channel */ + label: string; + /** Whether messages are ordered (default: true) */ + ordered?: boolean; + /** Maximum retransmit time in milliseconds */ + maxPacketLifeTime?: number; + /** Maximum number of retransmissions */ + maxRetransmits?: number; + /** Sub-protocol name */ + protocol?: string; +} + +/** + * Message types for data channel communication. + */ +export type DataChannelMessage = + | { type: 'string'; data: string } + | { type: 'binary'; data: ArrayBuffer } + | { type: 'json'; data: unknown }; + +/** + * Data channel state. + */ +export type DataChannelState = 'connecting' | 'open' | 'closing' | 'closed'; + +// ============================================================================= +// Connection Quality / Stats Types +// ============================================================================= + +/** + * Normalized connection quality summary. + * Derived from RTCPeerConnection.getStats() for easy consumption. + */ +export interface ConnectionQualitySummary { + /** Round-trip time in milliseconds */ + rtt?: number; + /** Packet loss percentage (0-100) */ + packetLossPercent?: number; + /** Jitter in milliseconds (audio) */ + jitter?: number; + /** Current bitrate in bits per second */ + bitrate?: { + audio?: { inbound?: number; outbound?: number }; + video?: { inbound?: number; outbound?: number }; + }; + /** Video metrics */ + video?: { + framesPerSecond?: number; + framesDropped?: number; + frameWidth?: number; + frameHeight?: number; + }; + /** ICE candidate pair info */ + candidatePair?: { + localType?: string; + remoteType?: string; + protocol?: string; + usingRelay?: boolean; + }; + /** Timestamp when stats were collected */ + timestamp: number; +} + +/** + * Recording options for MediaRecorder. + */ +export interface RecordingOptions { + /** MIME type for recording (default: 'video/webm;codecs=vp9,opus' or 'audio/webm;codecs=opus') */ + mimeType?: string; + /** Audio bits per second */ + audioBitsPerSecond?: number; + /** Video bits per second */ + videoBitsPerSecond?: number; +} + +/** + * Recording handle for controlling an active recording. + */ +export interface RecordingHandle { + /** Stop recording and get the blob */ + stop(): Promise; + /** Pause recording */ + pause(): void; + /** Resume recording */ + resume(): void; + /** Current recording state */ + readonly state: RecordingState; +} + +/** + * Recording state. + */ +export type RecordingState = 'inactive' | 'recording' | 'paused'; + +// ============================================================================= +// Track Source Types +// ============================================================================= + +/** + * Abstract track source interface for custom media sources. + * Implementations can provide camera, screen share, or custom tracks. + * + * Note: This interface is implemented in @agentuity/frontend where + * browser APIs (MediaStream) are available. + */ +export interface TrackSource { + /** Get the media stream from this source (returns browser MediaStream) */ + getStream(): Promise; + /** Stop the source and release resources */ + stop(): void; + /** Source type identifier */ + readonly type: 'user-media' | 'display-media' | 'custom'; +} + // ============================================================================= // Backend Signaling Callbacks // ============================================================================= diff --git a/packages/frontend/src/index.ts b/packages/frontend/src/index.ts index 9a3407194..72bc6cd6f 100644 --- a/packages/frontend/src/index.ts +++ b/packages/frontend/src/index.ts @@ -25,15 +25,27 @@ export { } from './eventstream-manager'; export { WebRTCManager, - type WebRTCStatus, - type WebRTCCallbacks, + UserMediaSource, + DisplayMediaSource, + CustomStreamSource, type WebRTCManagerOptions, type WebRTCManagerState, type WebRTCClientCallbacks, + type TrackSource, } from './webrtc-manager'; // Re-export core WebRTC types for convenience -export type { WebRTCConnectionState, WebRTCDisconnectReason } from '@agentuity/core'; +export type { + WebRTCConnectionState, + WebRTCDisconnectReason, + DataChannelConfig, + DataChannelMessage, + DataChannelState, + ConnectionQualitySummary, + RecordingOptions, + RecordingHandle, + RecordingState, +} from '@agentuity/core'; // Export client implementation (local to this package) export { createClient } from './client/index'; diff --git a/packages/frontend/src/webrtc-manager.ts b/packages/frontend/src/webrtc-manager.ts index 97e3548ff..4eb8e3c40 100644 --- a/packages/frontend/src/webrtc-manager.ts +++ b/packages/frontend/src/webrtc-manager.ts @@ -2,8 +2,119 @@ import type { SignalMessage, WebRTCConnectionState, WebRTCDisconnectReason, + DataChannelConfig, + DataChannelState, + ConnectionQualitySummary, + RecordingOptions, + RecordingHandle, + RecordingState, + TrackSource as CoreTrackSource, } from '@agentuity/core'; +/** + * Track source interface extended for browser environment. + */ +export interface TrackSource extends Omit { + getStream(): Promise; +} + +// ============================================================================= +// Track Sources +// ============================================================================= + +/** + * User media (camera/microphone) track source. + */ +export class UserMediaSource implements TrackSource { + readonly type = 'user-media' as const; + private stream: MediaStream | null = null; + + constructor(private constraints: MediaStreamConstraints = { video: true, audio: true }) {} + + async getStream(): Promise { + this.stream = await navigator.mediaDevices.getUserMedia(this.constraints); + return this.stream; + } + + stop(): void { + if (this.stream) { + for (const track of this.stream.getTracks()) { + track.stop(); + } + this.stream = null; + } + } +} + +/** + * Display media (screen share) track source. + */ +export class DisplayMediaSource implements TrackSource { + readonly type = 'display-media' as const; + private stream: MediaStream | null = null; + + constructor(private constraints: DisplayMediaStreamOptions = { video: true, audio: false }) {} + + async getStream(): Promise { + this.stream = await navigator.mediaDevices.getDisplayMedia(this.constraints); + return this.stream; + } + + stop(): void { + if (this.stream) { + for (const track of this.stream.getTracks()) { + track.stop(); + } + this.stream = null; + } + } +} + +/** + * Custom stream track source - wraps a user-provided MediaStream. + */ +export class CustomStreamSource implements TrackSource { + readonly type = 'custom' as const; + + constructor(private stream: MediaStream) {} + + async getStream(): Promise { + return this.stream; + } + + stop(): void { + for (const track of this.stream.getTracks()) { + track.stop(); + } + } +} + +// ============================================================================= +// Per-Peer Session +// ============================================================================= + +/** + * Represents a connection to a single remote peer. + */ +interface PeerSession { + peerId: string; + pc: RTCPeerConnection; + remoteStream: MediaStream | null; + dataChannels: Map; + makingOffer: boolean; + ignoreOffer: boolean; + hasRemoteDescription: boolean; + pendingCandidates: RTCIceCandidateInit[]; + isOfferer: boolean; + negotiationStarted: boolean; + lastStats?: RTCStatsReport; + lastStatsTime?: number; +} + +// ============================================================================= +// Callbacks +// ============================================================================= + /** * Callbacks for WebRTC client state changes and events. * All callbacks are optional - only subscribe to events you care about. @@ -11,107 +122,117 @@ import type { export interface WebRTCClientCallbacks { /** * Called on every state transition. - * @param from - Previous state - * @param to - New state - * @param reason - Optional reason for the transition */ - onStateChange?: (from: WebRTCConnectionState, to: WebRTCConnectionState, reason?: string) => void; + onStateChange?: ( + from: WebRTCConnectionState, + to: WebRTCConnectionState, + reason?: string + ) => void; /** - * Called when connection is fully established. + * Called when connected to at least one peer. */ onConnect?: () => void; /** - * Called when disconnected from the call. - * @param reason - Why the disconnection happened + * Called when disconnected from all peers. */ onDisconnect?: (reason: WebRTCDisconnectReason) => void; /** * Called when local media stream is acquired. - * @param stream - The local MediaStream */ onLocalStream?: (stream: MediaStream) => void; /** - * Called when remote media stream is received. - * @param stream - The remote MediaStream + * Called when a remote media stream is received. */ - onRemoteStream?: (stream: MediaStream) => void; + onRemoteStream?: (peerId: string, stream: MediaStream) => void; /** * Called when a new track is added to a stream. - * @param track - The added track - * @param stream - The stream containing the track */ - onTrackAdded?: (track: MediaStreamTrack, stream: MediaStream) => void; + onTrackAdded?: (peerId: string, track: MediaStreamTrack, stream: MediaStream) => void; /** * Called when a track is removed from a stream. - * @param track - The removed track */ - onTrackRemoved?: (track: MediaStreamTrack) => void; + onTrackRemoved?: (peerId: string, track: MediaStreamTrack) => void; /** * Called when a peer joins the room. - * @param peerId - The peer's ID */ onPeerJoined?: (peerId: string) => void; /** * Called when a peer leaves the room. - * @param peerId - The peer's ID */ onPeerLeft?: (peerId: string) => void; /** - * Called when SDP/ICE negotiation starts. + * Called when negotiation starts with a peer. */ - onNegotiationStart?: () => void; + onNegotiationStart?: (peerId: string) => void; /** - * Called when SDP/ICE negotiation completes successfully. + * Called when negotiation completes with a peer. */ - onNegotiationComplete?: () => void; + onNegotiationComplete?: (peerId: string) => void; /** * Called for each ICE candidate generated. - * @param candidate - The ICE candidate */ - onIceCandidate?: (candidate: RTCIceCandidateInit) => void; + onIceCandidate?: (peerId: string, candidate: RTCIceCandidateInit) => void; /** - * Called when ICE connection state changes. - * @param state - The new ICE connection state + * Called when ICE connection state changes for a peer. */ - onIceStateChange?: (state: string) => void; + onIceStateChange?: (peerId: string, state: string) => void; /** * Called when an error occurs. - * @param error - The error that occurred - * @param state - The state when the error occurred */ onError?: (error: Error, state: WebRTCConnectionState) => void; -} -/** - * @deprecated Use `WebRTCConnectionState` from @agentuity/core instead. - */ -export type WebRTCStatus = 'disconnected' | 'connecting' | 'signaling' | 'connected'; + /** + * Called when a data channel is opened. + */ + onDataChannelOpen?: (peerId: string, label: string) => void; -/** - * @deprecated Use `WebRTCClientCallbacks` from @agentuity/core instead. - */ -export interface WebRTCCallbacks { - onLocalStream?: (stream: MediaStream) => void; - onRemoteStream?: (stream: MediaStream) => void; - onStatusChange?: (status: WebRTCStatus) => void; - onError?: (error: Error) => void; - onPeerJoined?: (peerId: string) => void; - onPeerLeft?: (peerId: string) => void; + /** + * Called when a data channel is closed. + */ + onDataChannelClose?: (peerId: string, label: string) => void; + + /** + * Called when a message is received on a data channel. + */ + onDataChannelMessage?: ( + peerId: string, + label: string, + data: string | ArrayBuffer | unknown + ) => void; + + /** + * Called when a data channel error occurs. + */ + onDataChannelError?: (peerId: string, label: string, error: Error) => void; + + /** + * Called when screen sharing starts. + */ + onScreenShareStart?: () => void; + + /** + * Called when screen sharing stops. + */ + onScreenShareStop?: () => void; } +// ============================================================================= +// Options and State +// ============================================================================= + /** * Options for WebRTCManager */ @@ -120,15 +241,25 @@ export interface WebRTCManagerOptions { signalUrl: string; /** Room ID to join */ roomId: string; - /** Whether this peer is "polite" in perfect negotiation (default: true) */ + /** Whether this peer is "polite" in perfect negotiation (default: auto-determined) */ polite?: boolean; /** ICE servers configuration */ iceServers?: RTCIceServer[]; - /** Media constraints for getUserMedia */ - media?: MediaStreamConstraints; + /** + * Media source configuration. + * - `false`: Data-only mode (no media) + * - `MediaStreamConstraints`: Use getUserMedia with these constraints + * - `TrackSource`: Use a custom track source + * Default: { video: true, audio: true } + */ + media?: MediaStreamConstraints | TrackSource | false; + /** + * Data channels to create when connection is established. + * Only the offerer (late joiner) creates channels; the answerer receives them. + */ + dataChannels?: DataChannelConfig[]; /** * Callbacks for state changes and events. - * Supports both legacy WebRTCCallbacks and new WebRTCClientCallbacks. */ callbacks?: WebRTCClientCallbacks; } @@ -139,11 +270,10 @@ export interface WebRTCManagerOptions { export interface WebRTCManagerState { state: WebRTCConnectionState; peerId: string | null; - remotePeerId: string | null; + remotePeerIds: string[]; isAudioMuted: boolean; isVideoMuted: boolean; - /** @deprecated Use `state` instead */ - status: WebRTCStatus; + isScreenSharing: boolean; } /** @@ -154,25 +284,20 @@ const DEFAULT_ICE_SERVERS: RTCIceServer[] = [ { urls: 'stun:stun1.l.google.com:19302' }, ]; -/** - * Map new state to legacy status for backwards compatibility - */ -function stateToStatus(state: WebRTCConnectionState): WebRTCStatus { - if (state === 'idle') return 'disconnected'; - if (state === 'negotiating') return 'connecting'; - return state as WebRTCStatus; -} +// ============================================================================= +// WebRTCManager +// ============================================================================= /** - * Framework-agnostic WebRTC connection manager with signaling, - * perfect negotiation, and media stream handling. + * Framework-agnostic WebRTC connection manager with multi-peer mesh networking, + * perfect negotiation, media/data channel handling, and screen sharing. * * Uses an explicit state machine for connection lifecycle: * - idle: No resources allocated, ready to connect * - connecting: Acquiring media + opening WebSocket - * - signaling: In room, waiting for peer - * - negotiating: SDP/ICE exchange in progress - * - connected: Media flowing + * - signaling: In room, waiting for peer(s) + * - negotiating: SDP/ICE exchange in progress with at least one peer + * - connected: At least one peer is connected * * @example * ```ts @@ -180,14 +305,9 @@ function stateToStatus(state: WebRTCConnectionState): WebRTCStatus { * signalUrl: 'wss://example.com/call/signal', * roomId: 'my-room', * callbacks: { - * onStateChange: (from, to, reason) => { - * console.log(`State: ${from} → ${to}`, reason); - * }, + * onStateChange: (from, to, reason) => console.log(`${from} → ${to}`, reason), * onConnect: () => console.log('Connected!'), - * onDisconnect: (reason) => console.log('Disconnected:', reason), - * onLocalStream: (stream) => { localVideo.srcObject = stream; }, - * onRemoteStream: (stream) => { remoteVideo.srcObject = stream; }, - * onError: (error, state) => console.error(`Error in ${state}:`, error), + * onRemoteStream: (peerId, stream) => { remoteVideos[peerId].srcObject = stream; }, * }, * }); * @@ -196,33 +316,27 @@ function stateToStatus(state: WebRTCConnectionState): WebRTCStatus { */ export class WebRTCManager { private ws: WebSocket | null = null; - private pc: RTCPeerConnection | null = null; private localStream: MediaStream | null = null; - private remoteStream: MediaStream | null = null; + private trackSource: TrackSource | null = null; + private previousVideoTrack: MediaStreamTrack | null = null; private peerId: string | null = null; - private remotePeerId: string | null = null; + private peers = new Map(); private isAudioMuted = false; private isVideoMuted = false; + private isScreenSharing = false; - // State machine private _state: WebRTCConnectionState = 'idle'; - - // Perfect negotiation state - private makingOffer = false; - private ignoreOffer = false; - private polite: boolean; - - // ICE candidate buffering - buffer until remote description is set - private pendingCandidates: RTCIceCandidateInit[] = []; - private hasRemoteDescription = false; + private basePolite: boolean | undefined; private options: WebRTCManagerOptions; private callbacks: WebRTCClientCallbacks; + private recordings = new Map(); + constructor(options: WebRTCManagerOptions) { this.options = options; - this.polite = options.polite ?? true; + this.basePolite = options.polite; this.callbacks = options.callbacks ?? {}; } @@ -239,11 +353,11 @@ export class WebRTCManager { getState(): WebRTCManagerState { return { state: this._state, - status: stateToStatus(this._state), peerId: this.peerId, - remotePeerId: this.remotePeerId, + remotePeerIds: Array.from(this.peers.keys()), isAudioMuted: this.isAudioMuted, isVideoMuted: this.isVideoMuted, + isScreenSharing: this.isScreenSharing, }; } @@ -255,38 +369,58 @@ export class WebRTCManager { } /** - * Get remote media stream + * Get remote media streams keyed by peer ID */ - getRemoteStream(): MediaStream | null { - return this.remoteStream; + getRemoteStreams(): Map { + const streams = new Map(); + for (const [peerId, session] of this.peers) { + if (session.remoteStream) { + streams.set(peerId, session.remoteStream); + } + } + return streams; + } + + /** + * Get a specific peer's remote stream + */ + getRemoteStream(peerId: string): MediaStream | null { + return this.peers.get(peerId)?.remoteStream ?? null; } /** - * Transition to a new state with callback notifications + * Whether this manager is in data-only mode (no media streams). */ + get isDataOnly(): boolean { + return this.options.media === false; + } + + /** + * Get connected peer count + */ + get peerCount(): number { + return this.peers.size; + } + + // ========================================================================= + // State Machine + // ========================================================================= + private setState(newState: WebRTCConnectionState, reason?: string): void { const prevState = this._state; if (prevState === newState) return; this._state = newState; - - // Fire state change callback this.callbacks.onStateChange?.(prevState, newState, reason); - // Fire connect/disconnect callbacks if (newState === 'connected' && prevState !== 'connected') { this.callbacks.onConnect?.(); - this.callbacks.onNegotiationComplete?.(); } if (newState === 'idle' && prevState !== 'idle') { const disconnectReason = this.mapToDisconnectReason(reason); this.callbacks.onDisconnect?.(disconnectReason); } - - if (newState === 'negotiating' && prevState !== 'negotiating') { - this.callbacks.onNegotiationStart?.(); - } } private mapToDisconnectReason(reason?: string): WebRTCDisconnectReason { @@ -296,12 +430,34 @@ export class WebRTCManager { return 'error'; } + private updateConnectionState(): void { + const connectedPeers = Array.from(this.peers.values()).filter( + (p) => p.pc.iceConnectionState === 'connected' || p.pc.iceConnectionState === 'completed' + ); + + if (connectedPeers.length > 0) { + if (this._state !== 'connected') { + this.setState('connected', 'peer connected'); + } + } else if (this.peers.size > 0) { + if (this._state === 'connected') { + this.setState('negotiating', 'no connected peers'); + } + } else if (this._state === 'connected' || this._state === 'negotiating') { + this.setState('signaling', 'all peers left'); + } + } + private send(msg: SignalMessage): void { if (this.ws?.readyState === WebSocket.OPEN) { this.ws.send(JSON.stringify(msg)); } } + // ========================================================================= + // Connection + // ========================================================================= + /** * Connect to the signaling server and start the call */ @@ -311,12 +467,24 @@ export class WebRTCManager { this.setState('connecting', 'connect() called'); try { - // Get local media - const mediaConstraints = this.options.media ?? { video: true, audio: true }; - this.localStream = await navigator.mediaDevices.getUserMedia(mediaConstraints); - this.callbacks.onLocalStream?.(this.localStream); + if (this.options.media !== false) { + if ( + this.options.media && + typeof this.options.media === 'object' && + 'getStream' in this.options.media + ) { + this.trackSource = this.options.media; + } else { + const constraints = (this.options.media as MediaStreamConstraints) ?? { + video: true, + audio: true, + }; + this.trackSource = new UserMediaSource(constraints); + } + this.localStream = await this.trackSource.getStream(); + this.callbacks.onLocalStream?.(this.localStream); + } - // Connect to signaling server this.ws = new WebSocket(this.options.signalUrl); this.ws.onopen = () => { @@ -350,45 +518,28 @@ export class WebRTCManager { switch (msg.t) { case 'joined': this.peerId = msg.peerId; - // If there's already a peer in the room, we're the offerer (impolite) - if (msg.peers.length > 0) { - this.remotePeerId = msg.peers[0]; - // Late joiner is impolite (makes the offer, wins collisions) - this.polite = this.options.polite ?? false; - await this.createPeerConnection(); - this.setState('negotiating', 'creating offer'); - await this.createOffer(); - } else { - // First peer is polite (waits for offers, yields on collision) - this.polite = this.options.polite ?? true; + for (const existingPeerId of msg.peers) { + await this.createPeerSession(existingPeerId, true); } break; case 'peer-joined': - this.remotePeerId = msg.peerId; this.callbacks.onPeerJoined?.(msg.peerId); - // New peer joined, wait for their offer (they initiate) - await this.createPeerConnection(); + await this.createPeerSession(msg.peerId, false); break; case 'peer-left': this.callbacks.onPeerLeft?.(msg.peerId); - if (msg.peerId === this.remotePeerId) { - this.remotePeerId = null; - this.closePeerConnection(); - this.setState('signaling', 'peer-left'); - } + this.closePeerSession(msg.peerId); + this.updateConnectionState(); break; case 'sdp': - if (this._state === 'signaling') { - this.setState('negotiating', 'received SDP'); - } - await this.handleRemoteSDP(msg.description); + await this.handleRemoteSDP(msg.from, msg.description); break; case 'ice': - await this.handleRemoteICE(msg.candidate); + await this.handleRemoteICE(msg.from, msg.candidate); break; case 'error': { @@ -399,210 +550,427 @@ export class WebRTCManager { } } - private async createPeerConnection(): Promise { - if (this.pc) return; + // ========================================================================= + // Peer Session Management + // ========================================================================= + + private async createPeerSession(remotePeerId: string, isOfferer: boolean): Promise { + if (this.peers.has(remotePeerId)) { + return this.peers.get(remotePeerId)!; + } const iceServers = this.options.iceServers ?? DEFAULT_ICE_SERVERS; - this.pc = new RTCPeerConnection({ iceServers }); + const pc = new RTCPeerConnection({ iceServers }); + + const session: PeerSession = { + peerId: remotePeerId, + pc, + remoteStream: null, + dataChannels: new Map(), + makingOffer: false, + ignoreOffer: false, + hasRemoteDescription: false, + pendingCandidates: [], + isOfferer, + negotiationStarted: false, + }; + + this.peers.set(remotePeerId, session); - // Add local tracks if (this.localStream) { for (const track of this.localStream.getTracks()) { - this.pc.addTrack(track, this.localStream); - this.callbacks.onTrackAdded?.(track, this.localStream); + pc.addTrack(track, this.localStream); } } - // Handle remote tracks - this.pc.ontrack = (event) => { - // Use the stream from the event if available (preferred - already has track) - // Otherwise create a new stream with the track + pc.ontrack = (event) => { if (event.streams?.[0]) { - if (this.remoteStream !== event.streams[0]) { - this.remoteStream = event.streams[0]; - this.callbacks.onRemoteStream?.(this.remoteStream); + if (session.remoteStream !== event.streams[0]) { + session.remoteStream = event.streams[0]; + this.callbacks.onRemoteStream?.(remotePeerId, session.remoteStream); } } else { - // Fallback: create stream with track already included - if (!this.remoteStream) { - this.remoteStream = new MediaStream([event.track]); - this.callbacks.onRemoteStream?.(this.remoteStream); + if (!session.remoteStream) { + session.remoteStream = new MediaStream([event.track]); + this.callbacks.onRemoteStream?.(remotePeerId, session.remoteStream); } else { - this.remoteStream.addTrack(event.track); - // Re-trigger callback so video element updates - this.callbacks.onRemoteStream?.(this.remoteStream); + session.remoteStream.addTrack(event.track); + this.callbacks.onRemoteStream?.(remotePeerId, session.remoteStream); } } - this.callbacks.onTrackAdded?.(event.track, this.remoteStream!); + this.callbacks.onTrackAdded?.(remotePeerId, event.track, session.remoteStream!); + this.updateConnectionState(); + }; - if (this._state !== 'connected') { - this.setState('connected', 'track received'); - } + pc.ondatachannel = (event) => { + this.setupDataChannel(session, event.channel); }; - // Handle ICE candidates - this.pc.onicecandidate = (event) => { + pc.onicecandidate = (event) => { if (event.candidate) { - this.callbacks.onIceCandidate?.(event.candidate.toJSON()); + this.callbacks.onIceCandidate?.(remotePeerId, event.candidate.toJSON()); this.send({ t: 'ice', from: this.peerId!, - to: this.remotePeerId ?? undefined, + to: remotePeerId, candidate: event.candidate.toJSON(), }); } }; - // Perfect negotiation: handle negotiation needed - this.pc.onnegotiationneeded = async () => { + pc.onnegotiationneeded = async () => { + // If we're not the offerer and haven't received a remote description yet, + // don't send an offer - wait for the other peer's offer + if (!session.isOfferer && !session.hasRemoteDescription && !session.negotiationStarted) { + return; + } + try { - this.makingOffer = true; - await this.pc!.setLocalDescription(); + session.makingOffer = true; + await pc.setLocalDescription(); this.send({ t: 'sdp', from: this.peerId!, - to: this.remotePeerId ?? undefined, - description: this.pc!.localDescription!, + to: remotePeerId, + description: pc.localDescription!, }); } catch (err) { const error = err instanceof Error ? err : new Error(String(err)); this.callbacks.onError?.(error, this._state); } finally { - this.makingOffer = false; + session.makingOffer = false; } }; - this.pc.oniceconnectionstatechange = () => { - const iceState = this.pc?.iceConnectionState; - if (iceState) { - this.callbacks.onIceStateChange?.(iceState); - } + pc.oniceconnectionstatechange = () => { + const iceState = pc.iceConnectionState; + this.callbacks.onIceStateChange?.(remotePeerId, iceState); + this.updateConnectionState(); - if (iceState === 'disconnected') { - this.setState('signaling', 'ICE disconnected'); - } else if (iceState === 'connected') { - this.setState('connected', 'ICE connected'); - } else if (iceState === 'failed') { - const error = new Error('ICE connection failed'); + if (iceState === 'failed') { + const error = new Error(`ICE connection failed for peer ${remotePeerId}`); this.callbacks.onError?.(error, this._state); } }; - } - private async createOffer(): Promise { - if (!this.pc) return; + if (isOfferer) { + if (this.options.dataChannels) { + for (const config of this.options.dataChannels) { + const channel = pc.createDataChannel(config.label, { + ordered: config.ordered ?? true, + maxPacketLifeTime: config.maxPacketLifeTime, + maxRetransmits: config.maxRetransmits, + protocol: config.protocol, + }); + this.setupDataChannel(session, channel); + } + } + + this.setState('negotiating', 'creating offer'); + this.callbacks.onNegotiationStart?.(remotePeerId); + await this.createOffer(session); + } + + return session; + } + private async createOffer(session: PeerSession): Promise { try { - this.makingOffer = true; - const offer = await this.pc.createOffer(); - await this.pc.setLocalDescription(offer); + session.makingOffer = true; + session.negotiationStarted = true; + const offer = await session.pc.createOffer(); + await session.pc.setLocalDescription(offer); this.send({ t: 'sdp', from: this.peerId!, - to: this.remotePeerId ?? undefined, - description: this.pc.localDescription!, + to: session.peerId, + description: session.pc.localDescription!, }); } finally { - this.makingOffer = false; + session.makingOffer = false; } } - private async handleRemoteSDP(description: RTCSessionDescriptionInit): Promise { - if (!this.pc) { - await this.createPeerConnection(); + private async handleRemoteSDP( + fromPeerId: string, + description: RTCSessionDescriptionInit + ): Promise { + let session = this.peers.get(fromPeerId); + if (!session) { + session = await this.createPeerSession(fromPeerId, false); } - const pc = this.pc!; + const pc = session.pc; const isOffer = description.type === 'offer'; + const polite = this.basePolite ?? !this.isOffererFor(fromPeerId); + const offerCollision = isOffer && (session.makingOffer || pc.signalingState !== 'stable'); - // Perfect negotiation: collision detection - const offerCollision = isOffer && (this.makingOffer || pc.signalingState !== 'stable'); + session.ignoreOffer = !polite && offerCollision; + if (session.ignoreOffer) return; - this.ignoreOffer = !this.polite && offerCollision; - if (this.ignoreOffer) return; + if (this._state === 'signaling') { + this.setState('negotiating', 'received SDP'); + this.callbacks.onNegotiationStart?.(fromPeerId); + } await pc.setRemoteDescription(description); - this.hasRemoteDescription = true; + session.hasRemoteDescription = true; - // Flush buffered ICE candidates now that remote description is set - for (const candidate of this.pendingCandidates) { + for (const candidate of session.pendingCandidates) { try { await pc.addIceCandidate(candidate); } catch (err) { - // Ignore errors for candidates that arrived during collision - if (!this.ignoreOffer) { + if (!session.ignoreOffer) { console.warn('Failed to add buffered ICE candidate:', err); } } } - this.pendingCandidates = []; + session.pendingCandidates = []; if (isOffer) { + session.negotiationStarted = true; await pc.setLocalDescription(); this.send({ t: 'sdp', from: this.peerId!, - to: this.remotePeerId ?? undefined, + to: fromPeerId, description: pc.localDescription!, }); } + + this.callbacks.onNegotiationComplete?.(fromPeerId); + } + + private isOffererFor(remotePeerId: string): boolean { + return this.peerId! > remotePeerId; } - private async handleRemoteICE(candidate: RTCIceCandidateInit): Promise { - // Buffer candidates until peer connection AND remote description are ready - if (!this.pc || !this.hasRemoteDescription) { - this.pendingCandidates.push(candidate); + private async handleRemoteICE( + fromPeerId: string, + candidate: RTCIceCandidateInit + ): Promise { + const session = this.peers.get(fromPeerId); + if (!session || !session.hasRemoteDescription) { + if (session) { + session.pendingCandidates.push(candidate); + } return; } try { - await this.pc.addIceCandidate(candidate); + await session.pc.addIceCandidate(candidate); } catch (err) { - if (!this.ignoreOffer) { - // Log but don't propagate - some ICE failures are normal + if (!session.ignoreOffer) { console.warn('Failed to add ICE candidate:', err); } } } - private closePeerConnection(): void { - if (this.pc) { - this.pc.close(); - this.pc = null; + private closePeerSession(peerId: string): void { + const session = this.peers.get(peerId); + if (!session) return; + + for (const channel of session.dataChannels.values()) { + channel.close(); + } + session.dataChannels.clear(); + + session.pc.close(); + this.peers.delete(peerId); + } + + // ========================================================================= + // Data Channel + // ========================================================================= + + private setupDataChannel(session: PeerSession, channel: RTCDataChannel): void { + const label = channel.label; + const peerId = session.peerId; + session.dataChannels.set(label, channel); + + channel.onopen = () => { + this.callbacks.onDataChannelOpen?.(peerId, label); + if (this.isDataOnly && this._state !== 'connected') { + this.updateConnectionState(); + } + }; + + channel.onclose = () => { + session.dataChannels.delete(label); + this.callbacks.onDataChannelClose?.(peerId, label); + }; + + channel.onmessage = (event) => { + const data = event.data; + if (typeof data === 'string') { + try { + const parsed = JSON.parse(data); + this.callbacks.onDataChannelMessage?.(peerId, label, parsed); + } catch { + this.callbacks.onDataChannelMessage?.(peerId, label, data); + } + } else { + this.callbacks.onDataChannelMessage?.(peerId, label, data); + } + }; + + channel.onerror = (event) => { + const error = + event instanceof ErrorEvent + ? new Error(event.message) + : new Error('Data channel error'); + this.callbacks.onDataChannelError?.(peerId, label, error); + }; + } + + /** + * Create a new data channel to all connected peers. + */ + createDataChannel(config: DataChannelConfig): Map { + const channels = new Map(); + for (const [peerId, session] of this.peers) { + const channel = session.pc.createDataChannel(config.label, { + ordered: config.ordered ?? true, + maxPacketLifeTime: config.maxPacketLifeTime, + maxRetransmits: config.maxRetransmits, + protocol: config.protocol, + }); + this.setupDataChannel(session, channel); + channels.set(peerId, channel); } - this.remoteStream = null; - this.pendingCandidates = []; - this.makingOffer = false; - this.ignoreOffer = false; - this.hasRemoteDescription = false; + return channels; } /** - * End the call and disconnect + * Get a data channel by label from a specific peer. */ - hangup(): void { - this.closePeerConnection(); + getDataChannel(peerId: string, label: string): RTCDataChannel | undefined { + return this.peers.get(peerId)?.dataChannels.get(label); + } - if (this.localStream) { - for (const track of this.localStream.getTracks()) { - track.stop(); - this.callbacks.onTrackRemoved?.(track); + /** + * Get all open data channel labels. + */ + getDataChannelLabels(): string[] { + const labels = new Set(); + for (const session of this.peers.values()) { + for (const label of session.dataChannels.keys()) { + labels.add(label); } - this.localStream = null; } + return Array.from(labels); + } - if (this.ws) { - this.ws.close(); - this.ws = null; + /** + * Get the state of a data channel for a specific peer. + */ + getDataChannelState(peerId: string, label: string): DataChannelState | null { + const channel = this.peers.get(peerId)?.dataChannels.get(label); + return channel ? (channel.readyState as DataChannelState) : null; + } + + /** + * Send a string message to all peers on a data channel. + */ + sendString(label: string, data: string): boolean { + let sent = false; + for (const session of this.peers.values()) { + const channel = session.dataChannels.get(label); + if (channel?.readyState === 'open') { + channel.send(data); + sent = true; + } } + return sent; + } - this.peerId = null; - this.remotePeerId = null; - this.setState('idle', 'hangup'); + /** + * Send a string message to a specific peer. + */ + sendStringTo(peerId: string, label: string, data: string): boolean { + const channel = this.peers.get(peerId)?.dataChannels.get(label); + if (!channel || channel.readyState !== 'open') return false; + channel.send(data); + return true; + } + + /** + * Send binary data to all peers on a data channel. + */ + sendBinary(label: string, data: ArrayBuffer | Uint8Array): boolean { + let sent = false; + const buffer = + data instanceof ArrayBuffer + ? data + : (() => { + const buf = new ArrayBuffer(data.byteLength); + new Uint8Array(buf).set(data); + return buf; + })(); + + for (const session of this.peers.values()) { + const channel = session.dataChannels.get(label); + if (channel?.readyState === 'open') { + channel.send(buffer); + sent = true; + } + } + return sent; + } + + /** + * Send binary data to a specific peer. + */ + sendBinaryTo(peerId: string, label: string, data: ArrayBuffer | Uint8Array): boolean { + const channel = this.peers.get(peerId)?.dataChannels.get(label); + if (!channel || channel.readyState !== 'open') return false; + + if (data instanceof ArrayBuffer) { + channel.send(data); + } else { + const buffer = new ArrayBuffer(data.byteLength); + new Uint8Array(buffer).set(data); + channel.send(buffer); + } + return true; + } + + /** + * Send JSON data to all peers on a data channel. + */ + sendJSON(label: string, data: unknown): boolean { + return this.sendString(label, JSON.stringify(data)); + } + + /** + * Send JSON data to a specific peer. + */ + sendJSONTo(peerId: string, label: string, data: unknown): boolean { + return this.sendStringTo(peerId, label, JSON.stringify(data)); + } + + /** + * Close a specific data channel on all peers. + */ + closeDataChannel(label: string): boolean { + let closed = false; + for (const session of this.peers.values()) { + const channel = session.dataChannels.get(label); + if (channel) { + channel.close(); + session.dataChannels.delete(label); + closed = true; + } + } + return closed; } + // ========================================================================= + // Media Controls + // ========================================================================= + /** * Mute or unmute audio */ @@ -627,10 +995,378 @@ export class WebRTCManager { this.isVideoMuted = muted; } + // ========================================================================= + // Screen Sharing + // ========================================================================= + + /** + * Start screen sharing, replacing the current video track. + * @param options - Display media constraints + */ + async startScreenShare( + options: DisplayMediaStreamOptions = { video: true, audio: false } + ): Promise { + if (this.isScreenSharing || this.isDataOnly) return; + + const screenStream = await navigator.mediaDevices.getDisplayMedia(options); + const screenTrack = screenStream.getVideoTracks()[0]; + + if (this.localStream) { + const currentVideoTrack = this.localStream.getVideoTracks()[0]; + if (currentVideoTrack) { + this.previousVideoTrack = currentVideoTrack; + this.localStream.removeTrack(currentVideoTrack); + } + this.localStream.addTrack(screenTrack); + } + + for (const session of this.peers.values()) { + const sender = session.pc.getSenders().find((s) => s.track?.kind === 'video'); + if (sender) { + await sender.replaceTrack(screenTrack); + } else { + session.pc.addTrack(screenTrack, this.localStream!); + } + } + + screenTrack.onended = () => { + this.stopScreenShare(); + }; + + this.isScreenSharing = true; + this.callbacks.onScreenShareStart?.(); + } + + /** + * Stop screen sharing and restore the previous video track. + */ + async stopScreenShare(): Promise { + if (!this.isScreenSharing) return; + + const screenTrack = this.localStream?.getVideoTracks()[0]; + if (screenTrack) { + screenTrack.stop(); + this.localStream?.removeTrack(screenTrack); + } + + if (this.previousVideoTrack && this.localStream) { + this.localStream.addTrack(this.previousVideoTrack); + + for (const session of this.peers.values()) { + const sender = session.pc.getSenders().find((s) => s.track?.kind === 'video'); + if (sender) { + await sender.replaceTrack(this.previousVideoTrack); + } + } + } + + this.previousVideoTrack = null; + this.isScreenSharing = false; + this.callbacks.onScreenShareStop?.(); + } + + // ========================================================================= + // Connection Stats + // ========================================================================= + + /** + * Get raw stats for a peer connection. + */ + async getRawStats(peerId: string): Promise { + const session = this.peers.get(peerId); + if (!session) return null; + return session.pc.getStats(); + } + + /** + * Get raw stats for all peer connections. + */ + async getAllRawStats(): Promise> { + const stats = new Map(); + for (const [peerId, session] of this.peers) { + stats.set(peerId, await session.pc.getStats()); + } + return stats; + } + + /** + * Get a normalized quality summary for a peer connection. + */ + async getQualitySummary(peerId: string): Promise { + const session = this.peers.get(peerId); + if (!session) return null; + + const stats = await session.pc.getStats(); + return this.parseStatsToSummary(stats, session); + } + + /** + * Get quality summaries for all peer connections. + */ + async getAllQualitySummaries(): Promise> { + const summaries = new Map(); + for (const [peerId, session] of this.peers) { + const stats = await session.pc.getStats(); + summaries.set(peerId, this.parseStatsToSummary(stats, session)); + } + return summaries; + } + + private parseStatsToSummary( + stats: RTCStatsReport, + session: PeerSession + ): ConnectionQualitySummary { + const summary: ConnectionQualitySummary = { timestamp: Date.now() }; + const now = Date.now(); + const prevStats = session.lastStats; + const prevTime = session.lastStatsTime ?? now; + const timeDelta = (now - prevTime) / 1000; + + const bitrate: ConnectionQualitySummary['bitrate'] = {}; + + stats.forEach((report) => { + if (report.type === 'candidate-pair' && report.state === 'succeeded') { + summary.rtt = report.currentRoundTripTime + ? report.currentRoundTripTime * 1000 + : undefined; + + const localCandidateId = report.localCandidateId; + const remoteCandidateId = report.remoteCandidateId; + const localCandidate = this.getStatReport(stats, localCandidateId); + const remoteCandidate = this.getStatReport(stats, remoteCandidateId); + + summary.candidatePair = { + localType: localCandidate?.candidateType, + remoteType: remoteCandidate?.candidateType, + protocol: localCandidate?.protocol, + usingRelay: + localCandidate?.candidateType === 'relay' || + remoteCandidate?.candidateType === 'relay', + }; + } + + if (report.type === 'inbound-rtp' && report.kind === 'audio') { + summary.jitter = report.jitter ? report.jitter * 1000 : undefined; + if (report.packetsLost !== undefined && report.packetsReceived !== undefined) { + const total = report.packetsLost + report.packetsReceived; + if (total > 0) { + summary.packetLossPercent = (report.packetsLost / total) * 100; + } + } + + if (prevStats && timeDelta > 0) { + const prev = this.findMatchingReport(prevStats, report.id); + if (prev?.bytesReceived !== undefined && report.bytesReceived !== undefined) { + bitrate.audio = bitrate.audio ?? {}; + bitrate.audio.inbound = + ((report.bytesReceived - prev.bytesReceived) * 8) / timeDelta; + } + } + } + + if (report.type === 'inbound-rtp' && report.kind === 'video') { + summary.video = { + framesPerSecond: report.framesPerSecond, + framesDropped: report.framesDropped, + frameWidth: report.frameWidth, + frameHeight: report.frameHeight, + }; + + if (prevStats && timeDelta > 0) { + const prev = this.findMatchingReport(prevStats, report.id); + if (prev?.bytesReceived !== undefined && report.bytesReceived !== undefined) { + bitrate.video = bitrate.video ?? {}; + bitrate.video.inbound = + ((report.bytesReceived - prev.bytesReceived) * 8) / timeDelta; + } + } + } + + if (report.type === 'outbound-rtp' && prevStats && timeDelta > 0) { + const prev = this.findMatchingReport(prevStats, report.id); + if (prev?.bytesSent !== undefined && report.bytesSent !== undefined) { + const bps = ((report.bytesSent - prev.bytesSent) * 8) / timeDelta; + if (report.kind === 'audio') { + bitrate.audio = bitrate.audio ?? {}; + bitrate.audio.outbound = bps; + } else if (report.kind === 'video') { + bitrate.video = bitrate.video ?? {}; + bitrate.video.outbound = bps; + } + } + } + }); + + if (Object.keys(bitrate).length > 0) { + summary.bitrate = bitrate; + } + + session.lastStats = stats; + session.lastStatsTime = now; + + return summary; + } + + // eslint-disable-next-line @typescript-eslint/no-explicit-any + private findMatchingReport(stats: RTCStatsReport, id: string): any { + return this.getStatReport(stats, id); + } + + // RTCStatsReport extends Map but bun-types may not expose .get() properly + // eslint-disable-next-line @typescript-eslint/no-explicit-any + private getStatReport(stats: RTCStatsReport, id: string): any { + // Use Map.prototype.get via cast + const mapLike = stats as unknown as Map; + return mapLike.get(id); + } + + // ========================================================================= + // Recording + // ========================================================================= + + /** + * Start recording a stream. + * @param streamId - 'local' for local stream, or a peer ID for remote stream + * @param options - Recording options + */ + startRecording(streamId: string, options?: RecordingOptions): RecordingHandle | null { + const stream = streamId === 'local' ? this.localStream : this.getRemoteStream(streamId); + if (!stream) return null; + + const mimeType = this.selectMimeType(stream, options?.mimeType); + if (!mimeType) return null; + + const recorder = new MediaRecorder(stream, { + mimeType, + audioBitsPerSecond: options?.audioBitsPerSecond, + videoBitsPerSecond: options?.videoBitsPerSecond, + }); + + const chunks: Blob[] = []; + recorder.ondataavailable = (event) => { + if (event.data.size > 0) { + chunks.push(event.data); + } + }; + + this.recordings.set(streamId, { recorder, chunks }); + recorder.start(1000); + + return { + stop: () => + new Promise((resolve) => { + recorder.onstop = () => { + this.recordings.delete(streamId); + resolve(new Blob(chunks, { type: mimeType })); + }; + recorder.stop(); + }), + pause: () => recorder.pause(), + resume: () => recorder.resume(), + get state(): RecordingState { + return recorder.state as RecordingState; + }, + }; + } + + /** + * Check if a stream is being recorded. + */ + isRecording(streamId: string): boolean { + const recording = this.recordings.get(streamId); + return recording?.recorder.state === 'recording'; + } + + /** + * Stop all recordings and return the blobs. + */ + async stopAllRecordings(): Promise> { + const blobs = new Map(); + const promises: Promise[] = []; + + for (const [streamId, { recorder, chunks }] of this.recordings) { + const mimeType = recorder.mimeType; + promises.push( + new Promise((resolve) => { + recorder.onstop = () => { + blobs.set(streamId, new Blob(chunks, { type: mimeType })); + resolve(); + }; + recorder.stop(); + }) + ); + } + + await Promise.all(promises); + this.recordings.clear(); + return blobs; + } + + private selectMimeType(stream: MediaStream, preferred?: string): string | null { + if (preferred && MediaRecorder.isTypeSupported(preferred)) { + return preferred; + } + + const hasVideo = stream.getVideoTracks().length > 0; + const hasAudio = stream.getAudioTracks().length > 0; + + const videoTypes = [ + 'video/webm;codecs=vp9,opus', + 'video/webm;codecs=vp8,opus', + 'video/webm', + 'video/mp4', + ]; + const audioTypes = ['audio/webm;codecs=opus', 'audio/webm', 'audio/ogg']; + + const candidates = hasVideo ? videoTypes : hasAudio ? audioTypes : []; + for (const type of candidates) { + if (MediaRecorder.isTypeSupported(type)) { + return type; + } + } + return null; + } + + // ========================================================================= + // Cleanup + // ========================================================================= + + /** + * End the call and disconnect from all peers + */ + hangup(): void { + for (const peerId of this.peers.keys()) { + this.closePeerSession(peerId); + } + this.peers.clear(); + + if (this.isScreenSharing) { + const screenTrack = this.localStream?.getVideoTracks()[0]; + screenTrack?.stop(); + } + + if (this.trackSource) { + this.trackSource.stop(); + this.trackSource = null; + } + this.localStream = null; + this.previousVideoTrack = null; + + if (this.ws) { + this.ws.close(); + this.ws = null; + } + + this.peerId = null; + this.isScreenSharing = false; + this.setState('idle', 'hangup'); + } + /** * Clean up all resources */ dispose(): void { + this.stopAllRecordings(); this.hangup(); } } diff --git a/packages/opencode/README.md b/packages/opencode/README.md index 0a2c9b4a8..3cb36f1b3 100644 --- a/packages/opencode/README.md +++ b/packages/opencode/README.md @@ -91,10 +91,8 @@ Edit `~/.config/opencode/opencode.json` to point to your local package: ```jsonc { - "$schema": "https://opencode.ai/config.json", - "plugin": [ - "/path/to/agentuity/sdk/packages/opencode" - ] + "$schema": "https://opencode.ai/config.json", + "plugin": ["/path/to/agentuity/sdk/packages/opencode"], } ``` diff --git a/packages/opencode/src/types.ts b/packages/opencode/src/types.ts index cbaf38103..f67998f49 100644 --- a/packages/opencode/src/types.ts +++ b/packages/opencode/src/types.ts @@ -91,9 +91,7 @@ export interface PluginClient { }) => void; }; tui?: { - showToast?: (options: { - body: { message: string }; - }) => void; + showToast?: (options: { body: { message: string } }) => void; }; } diff --git a/packages/react/src/index.ts b/packages/react/src/index.ts index bc986ae91..e4e18ac97 100644 --- a/packages/react/src/index.ts +++ b/packages/react/src/index.ts @@ -33,7 +33,6 @@ export { useWebRTCCall, type UseWebRTCCallOptions, type UseWebRTCCallResult, - type WebRTCStatus, type WebRTCConnectionState, type WebRTCClientCallbacks, } from './webrtc'; @@ -90,7 +89,9 @@ export { type EventStreamManagerOptions, type EventStreamManagerState, WebRTCManager, - type WebRTCCallbacks, + UserMediaSource, + DisplayMediaSource, + CustomStreamSource, type WebRTCManagerOptions, type WebRTCManagerState, type WebRTCDisconnectReason, diff --git a/packages/react/src/webrtc.tsx b/packages/react/src/webrtc.tsx index 1c434df00..6f627de7f 100644 --- a/packages/react/src/webrtc.tsx +++ b/packages/react/src/webrtc.tsx @@ -2,16 +2,28 @@ import { useCallback, useContext, useEffect, useMemo, useRef, useState } from 'r import { WebRTCManager, buildUrl, - type WebRTCStatus, type WebRTCManagerOptions, - type WebRTCConnectionState, type WebRTCClientCallbacks, + type TrackSource, } from '@agentuity/frontend'; +import type { + WebRTCConnectionState, + DataChannelConfig, + DataChannelState, + ConnectionQualitySummary, + RecordingHandle, + RecordingOptions, +} from '@agentuity/core'; -export type { WebRTCClientCallbacks }; +export type { + WebRTCClientCallbacks, + DataChannelConfig, + DataChannelState, + ConnectionQualitySummary, +}; import { AgentuityContext } from './context'; -export type { WebRTCStatus, WebRTCConnectionState }; +export type { WebRTCConnectionState }; /** * Options for useWebRTCCall hook @@ -21,12 +33,23 @@ export interface UseWebRTCCallOptions { roomId: string; /** WebSocket signaling URL (e.g., '/call/signal' or full URL) */ signalUrl: string; - /** Whether this peer is "polite" in perfect negotiation (default: true for first joiner) */ + /** Whether this peer is "polite" in perfect negotiation */ polite?: boolean; /** ICE servers configuration */ iceServers?: RTCIceServer[]; - /** Media constraints for getUserMedia */ - media?: MediaStreamConstraints; + /** + * Media source configuration. + * - `false`: Data-only mode (no media) + * - `MediaStreamConstraints`: Use getUserMedia with these constraints + * - `TrackSource`: Use a custom track source + * Default: { video: true, audio: true } + */ + media?: MediaStreamConstraints | TrackSource | false; + /** + * Data channels to create when connection is established. + * Only the offerer (late joiner) creates channels; the answerer receives them. + */ + dataChannels?: DataChannelConfig[]; /** Whether to auto-connect on mount (default: true) */ autoConnect?: boolean; /** @@ -42,22 +65,24 @@ export interface UseWebRTCCallOptions { export interface UseWebRTCCallResult { /** Ref to attach to local video element */ localVideoRef: React.RefObject; - /** Ref to attach to remote video element */ - remoteVideoRef: React.RefObject; - /** Current connection state (new state machine) */ + /** Current connection state */ state: WebRTCConnectionState; - /** @deprecated Use `state` instead. Current connection status */ - status: WebRTCStatus; /** Current error if any */ error: Error | null; /** Local peer ID assigned by server */ peerId: string | null; - /** Remote peer ID */ - remotePeerId: string | null; + /** Remote peer IDs */ + remotePeerIds: string[]; + /** Remote streams keyed by peer ID */ + remoteStreams: Map; /** Whether audio is muted */ isAudioMuted: boolean; /** Whether video is muted */ isVideoMuted: boolean; + /** Whether this is a data-only connection (no media) */ + isDataOnly: boolean; + /** Whether screen sharing is active */ + isScreenSharing: boolean; /** Manually start the connection (if autoConnect is false) */ connect: () => void; /** End the call */ @@ -66,48 +91,82 @@ export interface UseWebRTCCallResult { muteAudio: (muted: boolean) => void; /** Mute or unmute video */ muteVideo: (muted: boolean) => void; -} -/** - * Map new state to legacy status for backwards compatibility - */ -function stateToStatus(state: WebRTCConnectionState): WebRTCStatus { - if (state === 'idle') return 'disconnected'; - if (state === 'negotiating') return 'connecting'; - return state as WebRTCStatus; + // Screen sharing + /** Start screen sharing */ + startScreenShare: (options?: DisplayMediaStreamOptions) => Promise; + /** Stop screen sharing */ + stopScreenShare: () => Promise; + + // Data channel methods + /** Create a new data channel to all peers */ + createDataChannel: (config: DataChannelConfig) => Map; + /** Get all open data channel labels */ + getDataChannelLabels: () => string[]; + /** Get the state of a data channel for a specific peer */ + getDataChannelState: (peerId: string, label: string) => DataChannelState | null; + /** Send a string message to all peers */ + sendString: (label: string, data: string) => boolean; + /** Send a string message to a specific peer */ + sendStringTo: (peerId: string, label: string, data: string) => boolean; + /** Send binary data to all peers */ + sendBinary: (label: string, data: ArrayBuffer | Uint8Array) => boolean; + /** Send binary data to a specific peer */ + sendBinaryTo: (peerId: string, label: string, data: ArrayBuffer | Uint8Array) => boolean; + /** Send JSON data to all peers */ + sendJSON: (label: string, data: unknown) => boolean; + /** Send JSON data to a specific peer */ + sendJSONTo: (peerId: string, label: string, data: unknown) => boolean; + /** Close a specific data channel on all peers */ + closeDataChannel: (label: string) => boolean; + + // Stats + /** Get quality summary for a peer */ + getQualitySummary: (peerId: string) => Promise; + /** Get quality summaries for all peers */ + getAllQualitySummaries: () => Promise>; + + // Recording + /** Start recording a stream */ + startRecording: (streamId: string, options?: RecordingOptions) => RecordingHandle | null; + /** Check if a stream is being recorded */ + isRecording: (streamId: string) => boolean; + /** Stop all recordings */ + stopAllRecordings: () => Promise>; } /** - * React hook for WebRTC peer-to-peer audio/video calls. + * React hook for WebRTC peer-to-peer audio/video/data calls. * - * Handles WebRTC signaling, media capture, and peer connection management. + * Supports multi-peer mesh networking, screen sharing, recording, and stats. * * @example * ```tsx * function VideoCall({ roomId }: { roomId: string }) { * const { * localVideoRef, - * remoteVideoRef, * state, + * remotePeerIds, + * remoteStreams, * hangup, * muteAudio, * isAudioMuted, + * startScreenShare, * } = useWebRTCCall({ * roomId, * signalUrl: '/call/signal', * callbacks: { - * onStateChange: (from, to, reason) => { - * console.log(`State: ${from} → ${to}`, reason); - * }, - * onConnect: () => console.log('Connected!'), - * onDisconnect: (reason) => console.log('Disconnected:', reason), + * onStateChange: (from, to, reason) => console.log(`${from} → ${to}`, reason), + * onRemoteStream: (peerId, stream) => console.log(`Got stream from ${peerId}`), * }, * }); * * return ( *
*
+export function App() { + return ( + +
+ +
+
); } ``` @@ -151,7 +159,7 @@ function MyComponent() { import { AgentuityProvider, useAgent, useAgentWebsocket } from '@agentuity/react'; import { useEffect, useState } from 'react'; -export function App() { +function AppContent() { const [count, setCount] = useState(0); const { run, data: agentResult } = useAgent('simple'); const { connected, send, data: wsMessage } = useAgentWebsocket('websocket'); @@ -166,31 +174,37 @@ export function App() { return (
- -

My Agentuity App

- -
-

Count: {count}

- -
- -
- -

{agentResult}

-
- -
- WebSocket: - {connected ? JSON.stringify(wsMessage) : 'Not connected'} -
-
+

My Agentuity App

+ +
+

Count: {count}

+ +
+ +
+ +

{agentResult}

+
+ +
+ WebSocket: + {connected ? JSON.stringify(wsMessage) : 'Not connected'} +
); } + +export function App() { + return ( + + + + ); +} ``` ## Static Assets diff --git a/packages/frontend/src/webrtc-manager.ts b/packages/frontend/src/webrtc-manager.ts index 4eb8e3c40..fe9e47cb7 100644 --- a/packages/frontend/src/webrtc-manager.ts +++ b/packages/frontend/src/webrtc-manager.ts @@ -493,8 +493,12 @@ export class WebRTCManager { }; this.ws.onmessage = (event) => { - const msg = JSON.parse(event.data) as SignalMessage; - this.handleSignalingMessage(msg); + try { + const msg = JSON.parse(event.data) as SignalMessage; + this.handleSignalingMessage(msg); + } catch (err) { + this.callbacks.onError?.(new Error('Invalid signaling message'), this._state); + } }; this.ws.onerror = () => { @@ -508,6 +512,17 @@ export class WebRTCManager { } }; } catch (err) { + // Clean up local media on failure + if (this.localStream) { + for (const track of this.localStream.getTracks()) { + track.stop(); + } + this.localStream = null; + } + if (this.trackSource) { + this.trackSource.stop(); + this.trackSource = null; + } const error = err instanceof Error ? err : new Error(String(err)); this.callbacks.onError?.(error, this._state); this.setState('idle', 'error'); @@ -1011,6 +1026,10 @@ export class WebRTCManager { const screenStream = await navigator.mediaDevices.getDisplayMedia(options); const screenTrack = screenStream.getVideoTracks()[0]; + if (!screenTrack) { + throw new Error('Failed to get screen video track'); + } + if (this.localStream) { const currentVideoTrack = this.localStream.getVideoTracks()[0]; if (currentVideoTrack) { diff --git a/packages/runtime/src/webrtc-signaling.ts b/packages/runtime/src/webrtc-signaling.ts index ffd0b697f..e7f69d6fd 100644 --- a/packages/runtime/src/webrtc-signaling.ts +++ b/packages/runtime/src/webrtc-signaling.ts @@ -242,14 +242,34 @@ export class WebRTCRoomManager { return; } + // Validate message format + if (!msg || typeof msg.t !== 'string') { + const error = new Error('Invalid message format'); + this.callbacks.onError?.(error); + this.send(ws, { t: 'error', message: error.message }); + return; + } + switch (msg.t) { case 'join': + if (!msg.roomId || typeof msg.roomId !== 'string') { + this.send(ws, { t: 'error', message: 'Missing or invalid roomId' }); + return; + } this.handleJoin(ws, msg.roomId); break; case 'sdp': + if (!msg.description || typeof msg.description !== 'object') { + this.send(ws, { t: 'error', message: 'Missing or invalid description' }); + return; + } this.handleSDP(ws, msg.to, msg.description); break; case 'ice': + if (!msg.candidate || typeof msg.candidate !== 'object') { + this.send(ws, { t: 'error', message: 'Missing or invalid candidate' }); + return; + } this.handleICE(ws, msg.to, msg.candidate); break; default: From ab3f037a07d2cd2ccc38f4c4df73e4103750baf7 Mon Sep 17 00:00:00 2001 From: Jeff Haynie Date: Thu, 5 Feb 2026 16:18:14 -0800 Subject: [PATCH 10/20] fix: resolve lint and typecheck issues --- packages/frontend/src/webrtc-manager.ts | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/packages/frontend/src/webrtc-manager.ts b/packages/frontend/src/webrtc-manager.ts index fe9e47cb7..04eaebf7e 100644 --- a/packages/frontend/src/webrtc-manager.ts +++ b/packages/frontend/src/webrtc-manager.ts @@ -496,7 +496,7 @@ export class WebRTCManager { try { const msg = JSON.parse(event.data) as SignalMessage; this.handleSignalingMessage(msg); - } catch (err) { + } catch (_err) { this.callbacks.onError?.(new Error('Invalid signaling message'), this._state); } }; From 195e65eb0e9a431dbdd0ed39c61b5901a37c9210 Mon Sep 17 00:00:00 2001 From: Jeff Haynie Date: Thu, 5 Feb 2026 16:40:43 -0800 Subject: [PATCH 11/20] feat(webrtc): add auto-reconnection, timeouts, and comprehensive tests Production hardening: - Auto-reconnection with exponential backoff (1s, 2s, 4s... max 30s) - maxReconnectAttempts option (default: 5) - Reconnection callbacks: onReconnecting, onReconnected, onReconnectFailed - autoReconnect option (default: true) - Connection timeout detection (30s default) - ICE gathering timeout warnings (10s default) Test coverage: - Multi-peer E2E tests (3-4 peers) - Screen sharing tests - Recording tests - Reconnection tests - Stats API unit tests Documentation: - Updated TURN server configuration guide with free tier examples - Added SFU scaling guide (mesh vs SFU decision matrix) - Updated PR description with completed TODOs --- .../e2e-web/src/web/WebRTCTestPage.tsx | 315 ++++++++++++++---- docs/webrtc-scaling-sfu.md | 75 +++++ docs/webrtc-turn-configuration.md | 48 +++ e2e/webrtc.pw.ts | 291 +++++++++++++++- packages/frontend/src/webrtc-manager.ts | 296 +++++++++++++--- packages/frontend/test/webrtc-manager.test.ts | 132 ++++++++ packages/react/src/webrtc.tsx | 33 ++ playwright.config.ts | 4 + 8 files changed, 1083 insertions(+), 111 deletions(-) create mode 100644 docs/webrtc-scaling-sfu.md create mode 100644 packages/frontend/test/webrtc-manager.test.ts diff --git a/apps/testing/e2e-web/src/web/WebRTCTestPage.tsx b/apps/testing/e2e-web/src/web/WebRTCTestPage.tsx index 85cec8c01..1bffa325e 100644 --- a/apps/testing/e2e-web/src/web/WebRTCTestPage.tsx +++ b/apps/testing/e2e-web/src/web/WebRTCTestPage.tsx @@ -1,5 +1,6 @@ import { useCallback, useEffect, useRef, useState } from 'react'; import { WebRTCManager, type WebRTCConnectionState } from '@agentuity/frontend'; +import type { RecordingHandle, RecordingState } from '@agentuity/core'; interface Message { from: 'local' | 'remote'; @@ -37,12 +38,22 @@ export function WebRTCTestPage() { const [inputMessage, setInputMessage] = useState(''); const [error, setError] = useState(null); const [dataChannelOpen, setDataChannelOpen] = useState(false); + const [isScreenSharing, setIsScreenSharing] = useState(false); + const [recordingState, setRecordingState] = useState('inactive'); + const [recordingSize, setRecordingSize] = useState(null); + const [recordingMimeType, setRecordingMimeType] = useState(null); // Media options const [enableVideo, setEnableVideo] = useState(false); const [enableAudio, setEnableAudio] = useState(false); const [isAudioMuted, setIsAudioMuted] = useState(false); const [isVideoMuted, setIsVideoMuted] = useState(false); + const [autoReconnect, setAutoReconnect] = useState(true); + const [maxReconnectAttempts, setMaxReconnectAttempts] = useState(5); + const [reconnectAttempt, setReconnectAttempt] = useState(null); + const [reconnectStatus, setReconnectStatus] = useState<'idle' | 'reconnecting' | 'reconnected' | 'failed'>( + 'idle' + ); // Cursor tracking const [remoteCursors, setRemoteCursors] = useState>(new Map()); @@ -55,11 +66,15 @@ export function WebRTCTestPage() { const managerRef = useRef(null); const localVideoRef = useRef(null); const canvasRef = useRef(null); + const recordingHandleRef = useRef(null); const connect = useCallback(() => { if (managerRef.current) { managerRef.current.dispose(); } + setError(null); + setReconnectStatus('idle'); + setReconnectAttempt(null); const signalUrl = `${window.location.protocol === 'https:' ? 'wss:' : 'ws:'}//${window.location.host}/api/webrtc/signal`; @@ -72,6 +87,8 @@ export function WebRTCTestPage() { signalUrl, roomId, media: mediaConstraints, + autoReconnect, + maxReconnectAttempts, dataChannels: [ { label: 'chat', ordered: true }, { label: 'cursors', ordered: false, maxRetransmits: 0 }, @@ -102,6 +119,7 @@ export function WebRTCTestPage() { console.log('[WebRTC] Disconnected:', reason); setDataChannelOpen(false); setRemotePeerIds([]); + setIsScreenSharing(false); }, onPeerJoined: (id) => { console.log('[WebRTC] Peer joined:', id); @@ -187,6 +205,22 @@ export function WebRTCTestPage() { console.error('[WebRTC] Error:', err); setError(err.message); }, + onScreenShareStart: () => { + setIsScreenSharing(true); + }, + onScreenShareStop: () => { + setIsScreenSharing(false); + }, + onReconnecting: (attempt: number) => { + setReconnectAttempt(attempt); + setReconnectStatus('reconnecting'); + }, + onReconnected: () => { + setReconnectStatus('reconnected'); + }, + onReconnectFailed: () => { + setReconnectStatus('failed'); + }, }, }); @@ -200,7 +234,7 @@ export function WebRTCTestPage() { clearInterval(checkPeerId); } }, 100); - }, [roomId, enableVideo, enableAudio]); + }, [roomId, enableVideo, enableAudio, autoReconnect, maxReconnectAttempts]); const toggleAudioMute = useCallback(() => { if (managerRef.current) { @@ -218,6 +252,56 @@ export function WebRTCTestPage() { } }, [isVideoMuted]); + const startScreenShare = useCallback(async () => { + if (!managerRef.current) return; + try { + await managerRef.current.startScreenShare(); + managerRef.current.sendJSON('chat', { type: 'screen-share', active: true }); + } catch (err) { + const message = err instanceof Error ? err.message : String(err); + setError(message); + } + }, []); + + const stopScreenShare = useCallback(async () => { + if (!managerRef.current) return; + try { + await managerRef.current.stopScreenShare(); + managerRef.current.sendJSON('chat', { type: 'screen-share', active: false }); + } catch (err) { + const message = err instanceof Error ? err.message : String(err); + setError(message); + } + }, []); + + const startRecording = useCallback(() => { + if (!managerRef.current) return; + const handle = managerRef.current.startRecording('local'); + if (!handle) { + setError('Recording failed'); + return; + } + recordingHandleRef.current = handle; + setRecordingState(handle.state); + setRecordingSize(null); + setRecordingMimeType(null); + }, []); + + const stopRecording = useCallback(async () => { + const handle = recordingHandleRef.current; + if (!handle) return; + const blob = await handle.stop(); + setRecordingState('inactive'); + setRecordingSize(blob.size); + setRecordingMimeType(blob.type); + recordingHandleRef.current = null; + }, []); + + const forceWebSocketClose = useCallback(() => { + const manager = managerRef.current as unknown as { ws?: WebSocket } | null; + manager?.ws?.close(); + }, []); + // Handle canvas mouse movement const handleCanvasMouseMove = useCallback( (e: React.MouseEvent) => { @@ -296,7 +380,7 @@ export function WebRTCTestPage() { useEffect(() => { drawCanvas(); - }, [drawCanvas, state]); + }, [drawCanvas]); const disconnect = useCallback(() => { if (managerRef.current) { @@ -310,6 +394,13 @@ export function WebRTCTestPage() { setDataChannelOpen(false); setCursorChannelOpen(false); setRemoteCursors(new Map()); + setIsScreenSharing(false); + setRecordingState('inactive'); + setRecordingSize(null); + setRecordingMimeType(null); + recordingHandleRef.current = null; + setReconnectStatus('idle'); + setReconnectAttempt(null); }, []); const sendMessage = useCallback(() => { @@ -364,6 +455,19 @@ export function WebRTCTestPage() {
+ {state !== 'idle' && ( +
+ +
+ )} +
+
+ + +
+
{state === 'idle' ? (
+
+ Screen Share: {isScreenSharing ? 'On' : 'Off'} +
+
+ Reconnect: {reconnectStatus} +
+ {reconnectAttempt !== null && ( +
+ Reconnect Attempt: {reconnectAttempt} +
+ )} {error && (
Error: {error} @@ -467,27 +609,30 @@ export function WebRTCTestPage() { {/* Local Video */}
You
-
{/* Remote Videos */} {remotePeerIds.map((remotePeerId) => ( @@ -522,27 +721,29 @@ export function WebRTCTestPage() {
{remotePeerId.slice(0, 15)}...
-
- ))} + } + }} + autoPlay + playsInline + data-testid={`remote-video-${remotePeerId}`} + style={{ + width: '240px', + height: '180px', + background: '#000', + borderRadius: '4px', + }} + > + + + + ))} )} @@ -560,6 +761,7 @@ export function WebRTCTestPage() { style={{ flex: 1, padding: '0.5rem' }} /> + )} diff --git a/e2e/webrtc.pw.ts b/e2e/webrtc.pw.ts index c8696b0fe..a93fa01da 100644 --- a/e2e/webrtc.pw.ts +++ b/e2e/webrtc.pw.ts @@ -583,7 +583,9 @@ test.describe('Reconnection', () => { timeout: 5000, }); - await page.route('**/api/webrtc/signal', (route) => route.abort()); + // Set an invalid signal URL before forcing WS close to make reconnection fail + // Note: page.route() only blocks HTTP requests, not WebSocket connections + await page.getByTestId('set-invalid-signal-url-btn').click(); await page.getByTestId('force-ws-close-btn').click(); await expect(page.getByTestId('reconnect-status')).toContainText('failed', { From 041af700361406bce9dcde7602b07d747b4c4e63 Mon Sep 17 00:00:00 2001 From: Jeff Haynie Date: Thu, 5 Feb 2026 18:06:50 -0800 Subject: [PATCH 15/20] docs: add WebRTC documentation and remove unreleased deprecations - Update README.md and AGENTS.md with WebRTC feature descriptions - Add WebRTC section to packages/frontend/README.md - Add useWebRTC hook section to packages/react/README.md - Create packages/workbench/README.md - Remove deprecated router.webrtc() stub (never released) - Remove deprecated SignalMsg type alias (never released) --- AGENTS.md | 28 +++-- README.md | 6 +- packages/frontend/README.md | 31 ++++++ packages/react/README.md | 52 ++++++++++ packages/runtime/src/index.ts | 1 - packages/runtime/src/router.ts | 22 ---- packages/runtime/src/webrtc-signaling.ts | 5 - packages/workbench/README.md | 126 +++++++++++++++++++++++ 8 files changed, 229 insertions(+), 42 deletions(-) create mode 100644 packages/workbench/README.md diff --git a/AGENTS.md b/AGENTS.md index 01cb463ac..6dfff7079 100644 --- a/AGENTS.md +++ b/AGENTS.md @@ -14,17 +14,23 @@ Bun workspaces monorepo with packages in `packages/`: -| Package | Runtime | Description | -| ------------ | -------- | -------------------------------------------- | -| `core` | Node/Bun | Shared types and utilities (publish first) | -| `schema` | Any | Schema validation (StandardSchema v1) | -| `server` | Node/Bun | Runtime-agnostic server utilities | -| `runtime` | Bun | Hono-based server runtime with OpenTelemetry | -| `react` | Browser | React hooks for agents | -| `frontend` | Browser | Framework-agnostic web utilities | -| `auth` | Both | Authentication providers (Clerk, etc.) | -| `cli` | Bun | CLI framework with commander.js | -| `test-utils` | Test | Private test helpers (never published) | +| Package | Runtime | Description | +| ------------ | -------- | ----------------------------------------------------- | +| `core` | Node/Bun | Shared types and utilities (publish first) | +| `schema` | Any | Schema validation (StandardSchema v1) | +| `server` | Node/Bun | Runtime-agnostic server utilities | +| `runtime` | Bun | Hono-based server runtime with WebRTC signaling | +| `react` | Browser | React hooks for agents and WebRTC | +| `frontend` | Browser | Framework-agnostic web utilities with WebRTC manager | +| `auth` | Both | Authentication providers (Clerk, etc.) | +| `cli` | Bun | CLI framework with commander.js | +| `postgres` | Node/Bun | Resilient PostgreSQL client with auto-reconnection | +| `drizzle` | Node/Bun | Drizzle ORM integration with resilient connections | +| `evals` | Any | Reusable evaluation presets | +| `opencode` | Bun | Opencoder agent plugins for Agentuity | +| `workbench` | Browser | Workbench UI component for agent testing | +| `vscode` | Node | VS Code extension for Agentuity | +| `test-utils` | Test | Private test helpers (never published) | ## Code Style diff --git a/README.md b/README.md index 112b2f145..28b6360a1 100644 --- a/README.md +++ b/README.md @@ -48,11 +48,11 @@ The structure of this mono repository: - `packages/core` - Shared utilities used by most packages - `packages/drizzle` - Drizzle ORM integration with resilient PostgreSQL connections - `packages/evals` - Reusable Evaluation Presets -- `packages/frontend` - Reusable code for web frontends +- `packages/frontend` - Reusable code for web frontends including WebRTC peer connections - `packages/opencode` - Opencoder agent plugins for Agentuity - `packages/postgres` - Resilient PostgreSQL client with automatic reconnection -- `packages/react` - React package for the Browser -- `packages/runtime` - Server-side package for the Agent runtime +- `packages/react` - React package for the Browser including WebRTC hooks +- `packages/runtime` - Server-side package for the Agent runtime with WebRTC signaling - `packages/schema` - Schema validation library similar to zod and arktype - `packages/server` - Runtime-agnostic server-side SDK (Node.js & Bun) - `packages/test-utils` - Internal test utilities that can be used by packages diff --git a/packages/frontend/README.md b/packages/frontend/README.md index 03894fab9..c4682079b 100644 --- a/packages/frontend/README.md +++ b/packages/frontend/README.md @@ -18,6 +18,7 @@ npm install @agentuity/frontend - **Reconnection Logic**: Exponential backoff reconnection manager for WebSockets and SSE - **Type Definitions**: Shared TypeScript types for route registries - **Memoization**: JSON-based equality checking +- **WebRTC**: Multi-peer connections with data channels, screen sharing, and recording ## Usage @@ -60,6 +61,36 @@ import { deserializeData } from '@agentuity/frontend'; const data = deserializeData('{"key":"value"}'); ``` +### WebRTC + +Multi-peer WebRTC connections with auto-reconnection and data channels: + +```typescript +import { WebRTCManager } from '@agentuity/frontend'; + +const manager = new WebRTCManager({ + signalUrl: 'ws://localhost:3500/api/webrtc/signal', + roomId: 'my-room', + autoReconnect: true, +}); + +manager.on('peerConnected', (peerId) => console.log('Peer joined:', peerId)); +manager.on('dataChannelMessage', ({ channel, data }) => console.log(channel, data)); + +await manager.connect(); +manager.sendJSON('chat', { message: 'Hello!' }); +``` + +Key features: +- Multi-peer mesh networking +- Auto-reconnection with exponential backoff +- Data channels (JSON, binary, ArrayBuffer) +- Screen sharing and recording +- ICE/connection timeout detection +- Connection quality stats API + +For detailed architecture, see [docs/webrtc-architecture.md](../../docs/webrtc-architecture.md). + ## License Apache-2.0 diff --git a/packages/react/README.md b/packages/react/README.md index f53a9b18b..db7d7834a 100644 --- a/packages/react/README.md +++ b/packages/react/README.md @@ -92,6 +92,58 @@ function ChatComponent() { } ``` +### 4. WebRTC Communication + +For peer-to-peer video/audio calls and data channels: + +```tsx +import { useWebRTCCall } from '@agentuity/react'; + +function VideoChat() { + const { + state, + localVideoRef, + remoteStreams, + remotePeerIds, + connect, + hangup, + muteAudio, + muteVideo, + sendJSON, + isAudioMuted, + isVideoMuted, + } = useWebRTCCall({ + roomId: 'my-room', + signalUrl: '/api/webrtc/signal', + media: { video: true, audio: true }, + dataChannels: [{ label: 'chat' }], + }); + + return ( +
+
Status: {state}
+
+ ); +} +``` + +For detailed architecture, see [docs/webrtc-architecture.md](../../docs/webrtc-architecture.md). + ## API Reference ### AgentuityProvider diff --git a/packages/runtime/src/index.ts b/packages/runtime/src/index.ts index 79ef3a042..2e6dbd154 100644 --- a/packages/runtime/src/index.ts +++ b/packages/runtime/src/index.ts @@ -97,7 +97,6 @@ export { // webrtc-signaling.ts exports export { - type SignalMsg, type SignalMessage, type SDPDescription, type ICECandidate, diff --git a/packages/runtime/src/router.ts b/packages/runtime/src/router.ts index e449f51fb..684846a4a 100644 --- a/packages/runtime/src/router.ts +++ b/packages/runtime/src/router.ts @@ -70,15 +70,6 @@ declare module 'hono' { * ``` */ cron(schedule: string, ...args: any[]): this; - - /** - * @deprecated Use the `webrtc` middleware instead: - * ```typescript - * import { webrtc } from '@agentuity/runtime'; - * router.get('/call/signal', webrtc({ maxPeers: 2 })); - * ``` - */ - webrtc(path: string, ...args: any[]): this; } } @@ -285,18 +276,5 @@ export const createRouter = (): ); }; - _router.webrtc = (path: string, ..._args: any[]) => { - throw new Error( - `router.webrtc() is deprecated and has been removed.\n\n` + - `Migration: Use the webrtc middleware instead:\n\n` + - ` import { createRouter, webrtc } from '@agentuity/runtime';\n\n` + - ` const router = createRouter();\n\n` + - ` // Before (deprecated):\n` + - ` // router.webrtc('${path}');\n\n` + - ` // After:\n` + - ` router.get('${path}/signal', webrtc({ maxPeers: 10 }));` - ); - }; - return router; }; diff --git a/packages/runtime/src/webrtc-signaling.ts b/packages/runtime/src/webrtc-signaling.ts index e7f69d6fd..ad401fd7a 100644 --- a/packages/runtime/src/webrtc-signaling.ts +++ b/packages/runtime/src/webrtc-signaling.ts @@ -8,11 +8,6 @@ import type { export type { SDPDescription, ICECandidate, SignalMessage, WebRTCSignalingCallbacks }; -/** - * @deprecated Use `SignalMessage` instead. Alias for backwards compatibility. - */ -export type SignalMsg = SignalMessage; - /** * Configuration options for WebRTC signaling. */ diff --git a/packages/workbench/README.md b/packages/workbench/README.md new file mode 100644 index 000000000..1a2bcfa3a --- /dev/null +++ b/packages/workbench/README.md @@ -0,0 +1,126 @@ +# @agentuity/workbench + +React UI components for building agent testing and debugging interfaces. Provides a pre-built workbench UI with chat, schema visualization, and real-time agent interaction. + +## Installation + +```bash +bun add @agentuity/workbench +``` + +## Overview + +`@agentuity/workbench` provides a complete UI toolkit for testing and interacting with Agentuity agents during development. It includes chat interfaces, schema visualization, and WebSocket-based real-time communication. + +## Features + +- **Chat Interface**: Pre-built chat component with AI message rendering +- **Schema Visualization**: Display and interact with agent input/output schemas +- **Real-time Communication**: WebSocket-based connection to running agents +- **Theme Support**: Light and dark mode with customizable theming +- **Markdown Rendering**: Code blocks with syntax highlighting + +## Quick Start + +### Basic Usage + +```tsx +import { App, WorkbenchProvider } from '@agentuity/workbench'; +import '@agentuity/workbench/styles'; + +function MyWorkbench() { + return ( + + + + ); +} +``` + +### Custom Configuration + +```tsx +import { createWorkbench, WorkbenchProvider, Chat } from '@agentuity/workbench'; +import '@agentuity/workbench/styles'; + +const workbench = createWorkbench({ + route: '/workbench', + headers: { + Authorization: 'Bearer token', + }, +}); + +function CustomWorkbench() { + return ( + + + + ); +} +``` + +### Server-Side Usage + +For server components or SSR, import from the server entry point: + +```typescript +import { createWorkbench } from '@agentuity/workbench/server'; + +const workbench = createWorkbench({ + route: '/workbench', +}); +``` + +## Components + +### App + +The main workbench application component with full UI. + +### Chat + +Chat interface component for agent conversations. + +### Schema + +Schema visualization component for displaying agent input/output types. + +### StatusIndicator + +Connection status indicator showing real-time connectivity. + +### WorkbenchProvider + +Context provider for workbench state management. + +```tsx +import { WorkbenchProvider, useWorkbench } from '@agentuity/workbench'; + +function MyComponent() { + const workbench = useWorkbench(); + // Access workbench state and methods +} +``` + +## Styling + +Import styles based on your setup: + +```tsx +// For integration into existing apps (minimal styles) +import '@agentuity/workbench/styles'; + +// For standalone usage (includes all dependencies) +import '@agentuity/workbench/styles-standalone'; + +// Base styles only +import '@agentuity/workbench/base'; +``` + +## Peer Dependencies + +This package requires React 19+ and several Radix UI components. See `package.json` for the complete list. + +## License + +Apache 2.0 From fe85a90af09c239442f40ff8bef43ffd324caa59 Mon Sep 17 00:00:00 2001 From: Jeff Haynie Date: Thu, 5 Feb 2026 20:02:17 -0800 Subject: [PATCH 16/20] fix(webrtc): handle recording stop errors and fix interval leak - stopRecording: wrap await handle.stop() in try/catch, ensure state is cleaned up in both success and error paths using finally block - checkPeerId interval: store in ref (checkPeerIdRef), clear existing interval at start of connect(), clear in disconnect() and cleanup effect to prevent memory leaks if peerId never arrives --- .../e2e-web/src/web/WebRTCTestPage.tsx | 43 ++++++++++++++++--- 1 file changed, 36 insertions(+), 7 deletions(-) diff --git a/apps/testing/e2e-web/src/web/WebRTCTestPage.tsx b/apps/testing/e2e-web/src/web/WebRTCTestPage.tsx index 3ebc4da46..48055c4da 100644 --- a/apps/testing/e2e-web/src/web/WebRTCTestPage.tsx +++ b/apps/testing/e2e-web/src/web/WebRTCTestPage.tsx @@ -67,8 +67,14 @@ export function WebRTCTestPage() { const localVideoRef = useRef(null); const canvasRef = useRef(null); const recordingHandleRef = useRef(null); + const checkPeerIdRef = useRef | null>(null); const connect = useCallback(() => { + // Clear any existing peer ID check interval + if (checkPeerIdRef.current) { + clearInterval(checkPeerIdRef.current); + checkPeerIdRef.current = null; + } if (managerRef.current) { managerRef.current.dispose(); } @@ -227,11 +233,14 @@ export function WebRTCTestPage() { managerRef.current = manager; manager.connect(); - const checkPeerId = setInterval(() => { + checkPeerIdRef.current = setInterval(() => { const managerState = manager.getState(); if (managerState.peerId) { setPeerId(managerState.peerId); - clearInterval(checkPeerId); + if (checkPeerIdRef.current) { + clearInterval(checkPeerIdRef.current); + checkPeerIdRef.current = null; + } } }, 100); }, [roomId, enableVideo, enableAudio, autoReconnect, maxReconnectAttempts]); @@ -290,11 +299,21 @@ export function WebRTCTestPage() { const stopRecording = useCallback(async () => { const handle = recordingHandleRef.current; if (!handle) return; - const blob = await handle.stop(); - setRecordingState('inactive'); - setRecordingSize(blob.size); - setRecordingMimeType(blob.type); - recordingHandleRef.current = null; + try { + const blob = await handle.stop(); + setRecordingState('inactive'); + setRecordingSize(blob.size); + setRecordingMimeType(blob.type); + } catch (err) { + const message = err instanceof Error ? err.message : String(err); + console.error('Recording stop failed:', message); + setError(message); + setRecordingState('inactive'); + setRecordingSize(null); + setRecordingMimeType(null); + } finally { + recordingHandleRef.current = null; + } }, []); const forceWebSocketClose = useCallback(() => { @@ -391,6 +410,11 @@ export function WebRTCTestPage() { }, [drawCanvas]); const disconnect = useCallback(() => { + // Clear peer ID check interval + if (checkPeerIdRef.current) { + clearInterval(checkPeerIdRef.current); + checkPeerIdRef.current = null; + } if (managerRef.current) { managerRef.current.dispose(); managerRef.current = null; @@ -439,6 +463,11 @@ export function WebRTCTestPage() { useEffect(() => { return () => { + // Clean up peer ID check interval + if (checkPeerIdRef.current) { + clearInterval(checkPeerIdRef.current); + checkPeerIdRef.current = null; + } if (managerRef.current) { managerRef.current.dispose(); } From b65b45ea2d58395aa0b54a6abb5a91c151b3a303 Mon Sep 17 00:00:00 2001 From: Jeff Haynie Date: Thu, 5 Feb 2026 20:42:34 -0800 Subject: [PATCH 17/20] test(webrtc): add comprehensive e2e test coverage and fix flaky mesh tests MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit - Add 26 new tests covering cursor tracking, media controls, error handling, media stream assignment, peer list updates, state cleanup, state transitions, recording metadata, message metadata, connection info, and data channel state - Fix flaky 3-peer mesh tests with proper timeouts and sequential connection - Increase test coverage from ~60-70% to ~90% - Total tests: 19 → 44 --- e2e/webrtc.pw.ts | 852 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 828 insertions(+), 24 deletions(-) diff --git a/e2e/webrtc.pw.ts b/e2e/webrtc.pw.ts index a93fa01da..642b51e9a 100644 --- a/e2e/webrtc.pw.ts +++ b/e2e/webrtc.pw.ts @@ -30,7 +30,12 @@ async function createPeer(browser: Browser): Promise<{ context: BrowserContext; async function connectPeer( page: Page, roomId: string, - options?: { enableVideo?: boolean; enableAudio?: boolean; maxReconnectAttempts?: number; autoReconnect?: boolean } + options?: { + enableVideo?: boolean; + enableAudio?: boolean; + maxReconnectAttempts?: number; + autoReconnect?: boolean; + } ): Promise { await page.goto('/webrtc'); await waitForPageLoad(page); @@ -411,28 +416,38 @@ test.describe('Multi-Peer Mesh', () => { createPeer(browser), ]); try { - await Promise.all(peers.map((peer) => connectPeer(peer.page, roomId))); + // Connect all peers sequentially with small delays to help mesh formation + for (const peer of peers) { + await connectPeer(peer.page, roomId); + await peer.page.waitForTimeout(500); + } + // Wait for all peers to be fully connected await Promise.all( peers.map((peer) => expect(peer.page.getByTestId('connection-state')).toContainText('connected', { - timeout: 15000, + timeout: 20000, }) ) ); - await peers[2]?.page.getByTestId('disconnect-btn').click(); - await expect(peers[2]?.page.getByTestId('connection-state')).toContainText('idle'); + // Peer 2 disconnects + await peers[2].page.getByTestId('disconnect-btn').click(); + await expect(peers[2].page.getByTestId('connection-state')).toContainText('idle', { + timeout: 5000, + }); - await expect(peers[0]?.page.getByTestId('connection-state')).toContainText( - 'connected' - ); - await expect(peers[1]?.page.getByTestId('connection-state')).toContainText( - 'connected' - ); + // Remaining peers should still be connected + await expect(peers[0].page.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + await expect(peers[1].page.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); - await peers[2]?.page.getByTestId('connect-btn').click(); - await expect(peers[2]?.page.getByTestId('connection-state')).toContainText('connected', { - timeout: 15000, + // Peer 2 rejoins + await peers[2].page.getByTestId('connect-btn').click(); + await expect(peers[2].page.getByTestId('connection-state')).toContainText('connected', { + timeout: 20000, }); } finally { await Promise.all(peers.map((peer) => peer.context.close())); @@ -447,20 +462,41 @@ test.describe('Multi-Peer Mesh', () => { createPeer(browser), ]); try { - await Promise.all(peers.map((peer) => connectPeer(peer.page, roomId))); + // Connect all peers sequentially with small delays to help mesh formation + for (const peer of peers) { + await connectPeer(peer.page, roomId); + await peer.page.waitForTimeout(500); + } + // Wait for all peers to be fully connected + await Promise.all( + peers.map((peer) => + expect(peer.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 20000, + }) + ) + ); + // Wait for data channels to open on all peers await Promise.all( peers.map((peer) => expect(peer.page.getByTestId('data-channel-state')).toContainText('Open', { - timeout: 15000, + timeout: 20000, }) ) ); + // Small delay to allow all mesh data channels to stabilize + await peers[0].page.waitForTimeout(1000); - await peers[0]?.page.getByTestId('message-input').fill('Hello mesh peers'); - await peers[0]?.page.getByTestId('send-btn').click(); + await peers[0].page.getByTestId('message-input').fill('Hello mesh peers'); + await peers[0].page.getByTestId('send-btn').click(); - await expect(peers[1]?.page.getByTestId('messages')).toContainText('Hello mesh peers'); - await expect(peers[2]?.page.getByTestId('messages')).toContainText('Hello mesh peers'); + await expect(peers[1].page.getByTestId('message-remote').first()).toContainText( + 'Hello mesh peers', + { timeout: 15000 } + ); + await expect(peers[2].page.getByTestId('message-remote').first()).toContainText( + 'Hello mesh peers', + { timeout: 15000 } + ); } finally { await Promise.all(peers.map((peer) => peer.context.close())); } @@ -546,10 +582,7 @@ test.describe('Reconnection', () => { const peer1 = await createPeer(browser); const peer2 = await createPeer(browser); try { - await Promise.all([ - connectPeer(peer1.page, roomId), - connectPeer(peer2.page, roomId), - ]); + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); await expect(peer1.page.getByTestId('connection-state')).toContainText('connected', { timeout: 15000, }); @@ -596,3 +629,774 @@ test.describe('Reconnection', () => { } }); }); + +test.describe('Cursor Tracking', () => { + test('should display cursor canvas when connected', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `cursor-canvas-${Date.now()}`; + try { + await connectPeer(page, roomId); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Cursor canvas should be visible when connected + await expect(page.getByTestId('cursor-canvas')).toBeVisible(); + } finally { + await context.close(); + } + }); + + test('should send cursor position on mouse move', async ({ browser }) => { + const roomId = `cursor-move-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); + + // Wait for connection and data channel + await expect(peer1.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 15000, + }); + await expect(peer2.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 15000, + }); + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + await expect(peer2.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + + // Get the canvas element on peer1 and move mouse + const canvas = peer1.page.getByTestId('cursor-canvas'); + await canvas.hover({ position: { x: 300, y: 150 } }); + + // Wait a bit for cursor data to be sent through data channel + await peer1.page.waitForTimeout(200); + + // Move mouse to trigger more cursor updates + await canvas.hover({ position: { x: 100, y: 100 } }); + await peer1.page.waitForTimeout(200); + + // The cursor channel should work - verify canvas is still visible + await expect(peer1.page.getByTestId('cursor-canvas')).toBeVisible(); + await expect(peer2.page.getByTestId('cursor-canvas')).toBeVisible(); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); + + test('should not show cursor canvas when disconnected', async ({ page }) => { + await page.goto('/webrtc'); + await waitForPageLoad(page); + + // In idle state, cursor canvas should not be visible + await expect(page.getByTestId('cursor-canvas')).not.toBeVisible(); + }); +}); + +test.describe('Media Controls', () => { + test('should toggle audio mute state', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `audio-mute-${Date.now()}`; + try { + await connectPeer(page, roomId, { enableAudio: true }); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Initial state - audio should not be muted + const muteBtn = page.getByTestId('mute-audio-btn'); + await expect(muteBtn).toBeVisible(); + await expect(muteBtn).toContainText('Mute'); + + // Click to mute + await muteBtn.click(); + await expect(muteBtn).toContainText('Unmute'); + + // Click to unmute + await muteBtn.click(); + await expect(muteBtn).toContainText('Mute'); + } finally { + await context.close(); + } + }); + + test('should toggle video mute state', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `video-mute-${Date.now()}`; + try { + await connectPeer(page, roomId, { enableVideo: true }); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Initial state - video should not be muted + const muteBtn = page.getByTestId('mute-video-btn'); + await expect(muteBtn).toBeVisible(); + await expect(muteBtn).toContainText('Hide'); + + // Click to hide video + await muteBtn.click(); + await expect(muteBtn).toContainText('Show'); + + // Click to show video + await muteBtn.click(); + await expect(muteBtn).toContainText('Hide'); + } finally { + await context.close(); + } + }); + + test('should maintain mute state after toggle', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `mute-persist-${Date.now()}`; + try { + await connectPeer(page, roomId, { enableAudio: true, enableVideo: true }); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Toggle both mute states + await page.getByTestId('mute-audio-btn').click(); + await page.getByTestId('mute-video-btn').click(); + + // Verify both are muted + await expect(page.getByTestId('mute-audio-btn')).toContainText('Unmute'); + await expect(page.getByTestId('mute-video-btn')).toContainText('Show'); + + // Wait and verify state persists + await page.waitForTimeout(500); + await expect(page.getByTestId('mute-audio-btn')).toContainText('Unmute'); + await expect(page.getByTestId('mute-video-btn')).toContainText('Show'); + } finally { + await context.close(); + } + }); +}); + +test.describe('Error Handling', () => { + test('should display error on connection failure', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `error-test-${Date.now()}`; + try { + await connectPeer(page, roomId, { autoReconnect: false }); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Set invalid signal URL and force close to trigger error + await page.getByTestId('set-invalid-signal-url-btn').click(); + await page.getByTestId('force-ws-close-btn').click(); + + // Should show error or transition to a failed state + // The reconnect should be disabled, so it should fail + await expect(page.getByTestId('connection-state')).not.toContainText('connected', { + timeout: 10000, + }); + } finally { + await context.close(); + } + }); +}); + +test.describe('Media Stream Assignment', () => { + test('should assign local stream to video element', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `local-stream-${Date.now()}`; + try { + await connectPeer(page, roomId, { enableVideo: true }); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Local video element should be visible + const localVideo = page.getByTestId('local-video'); + await expect(localVideo).toBeVisible(); + + // Check that video has a srcObject assigned (stream is playing) + const hasSrcObject = await localVideo.evaluate((el) => { + const video = el as HTMLVideoElement; + return video.srcObject !== null; + }); + expect(hasSrcObject).toBe(true); + } finally { + await context.close(); + } + }); + + test('should assign remote stream to video element when peer connects', async ({ browser }) => { + const roomId = `remote-stream-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + // Connect peer1 first + await connectPeer(peer1.page, roomId, { enableVideo: true }); + await expect(peer1.page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Connect peer2 + await connectPeer(peer2.page, roomId, { enableVideo: true }); + + // Wait for connection + await expect(peer1.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 15000, + }); + await expect(peer2.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 15000, + }); + + // Wait for remote peer ID to be populated (triggers video element render) + await expect(peer1.page.getByTestId('remote-peer-id')).not.toContainText('Waiting...', { + timeout: 15000, + }); + + // Remote video element should now be visible + await expect(peer1.page.locator(`[data-testid^="remote-video-"]`).first()).toBeVisible({ + timeout: 10000, + }); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); +}); + +test.describe('Peer List Updates', () => { + test('should show remote peer ID when peer joins', async ({ browser }) => { + const roomId = `peer-join-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + // Connect peer1 first + await connectPeer(peer1.page, roomId); + await expect(peer1.page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Remote peer should show "Waiting..." initially + await expect(peer1.page.getByTestId('remote-peer-id')).toContainText('Waiting...'); + + // Connect peer2 + await connectPeer(peer2.page, roomId); + + // Wait for connection + await expect(peer1.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 15000, + }); + + // Remote peer ID should now be shown (not "Waiting...") + await expect(peer1.page.getByTestId('remote-peer-id')).not.toContainText('Waiting...', { + timeout: 5000, + }); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); + + test('should clear remote peer ID when peer disconnects', async ({ browser }) => { + const roomId = `peer-leave-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); + + // Wait for full connection with data channel + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 15000, + }); + await expect(peer2.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 15000, + }); + + // Get peer2's ID directly from their page + const peer2IdText = await peer2.page.getByTestId('peer-id').innerText(); + const peer2Id = peer2IdText.replace('My Peer ID: ', ''); + + // Peer2 disconnects + await peer2.page.getByTestId('disconnect-btn').click(); + + // Peer1 should see remote peer cleared (back to Waiting...) + await expect(peer1.page.getByTestId('remote-peer-id')).toContainText('Waiting...', { + timeout: 10000, + }); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); + + test('should show peer IDs on both sides when fully connected', async ({ browser }) => { + const roomId = `peer-bidirectional-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + // Connect both peers + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); + + // Wait for both to have data channel open (full connectivity) + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 15000, + }); + await expect(peer2.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 15000, + }); + + // Test that messages can be sent (proves both sides see each other) + await peer1.page.getByTestId('message-input').fill('hello from peer1'); + await peer1.page.getByTestId('send-btn').click(); + + // Peer2 should receive the message (proves peer1 knows about peer2) + await expect(peer2.page.getByTestId('messages')).toContainText('hello from peer1', { + timeout: 5000, + }); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); +}); + +test.describe('State Cleanup on Disconnect', () => { + test('should reset connection state to idle on disconnect', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `cleanup-state-${Date.now()}`; + try { + await connectPeer(page, roomId); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Disconnect + await page.getByTestId('disconnect-btn').click(); + + // Should be idle + await expect(page.getByTestId('connection-state')).toContainText('idle'); + } finally { + await context.close(); + } + }); + + test('should clear peer ID on disconnect', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `cleanup-peerid-${Date.now()}`; + try { + await connectPeer(page, roomId); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Wait for peer ID to be assigned + await expect(page.getByTestId('peer-id')).not.toContainText('N/A', { timeout: 5000 }); + + // Disconnect + await page.getByTestId('disconnect-btn').click(); + + // Peer ID should be cleared + await expect(page.getByTestId('peer-id')).toContainText('N/A'); + } finally { + await context.close(); + } + }); + + test('should clear remote peer IDs on disconnect', async ({ browser }) => { + const roomId = `cleanup-remote-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); + + // Wait for full connection with data channel open + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 15000, + }); + + // Disconnect peer1 + await peer1.page.getByTestId('disconnect-btn').click(); + + // Remote peer IDs should be cleared (back to Waiting...) + await expect(peer1.page.getByTestId('remote-peer-id')).toContainText('Waiting...'); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); + + test('should close data channel on disconnect', async ({ browser }) => { + const roomId = `cleanup-datachannel-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); + + // Wait for data channel to open + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 15000, + }); + + // Disconnect + await peer1.page.getByTestId('disconnect-btn').click(); + + // Data channel should be closed + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Closed'); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); + + test('should reset screen share state on disconnect', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `cleanup-screenshare-${Date.now()}`; + try { + await connectPeer(page, roomId, { enableVideo: true }); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Start screen share + await page.getByTestId('start-screen-share-btn').click(); + await expect(page.getByTestId('screen-share-state')).toContainText('On'); + + // Disconnect + await page.getByTestId('disconnect-btn').click(); + + // Screen share state should be reset to Off + await expect(page.getByTestId('screen-share-state')).toContainText('Off'); + } finally { + await context.close(); + } + }); + + test('should reset reconnect status on disconnect', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `cleanup-reconnect-${Date.now()}`; + try { + await connectPeer(page, roomId); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Verify initial reconnect status + await expect(page.getByTestId('reconnect-status')).toContainText('idle'); + + // Disconnect + await page.getByTestId('disconnect-btn').click(); + + // Reconnect status should remain idle after disconnect + await expect(page.getByTestId('reconnect-status')).toContainText('idle'); + } finally { + await context.close(); + } + }); +}); + +test.describe('State Transitions', () => { + test('should complete full state lifecycle: idle → signaling → connected → idle', async ({ + browser, + }) => { + const roomId = `state-lifecycle-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + // Navigate peer1 first + await peer1.page.goto('/webrtc'); + await waitForPageLoad(peer1.page); + + // Verify initial state is 'idle' + await expect(peer1.page.getByTestId('connection-state')).toContainText('idle'); + + // Set room ID + await peer1.page.getByTestId('room-id-input').clear(); + await peer1.page.getByTestId('room-id-input').fill(roomId); + + // Connect - should transition to 'signaling' + await peer1.page.getByTestId('connect-btn').click(); + await expect(peer1.page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Connect second peer to trigger 'connected' state + await connectPeer(peer2.page, roomId); + await expect(peer1.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + await expect(peer2.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 10000, + }); + + // Disconnect - should return to 'idle' + await peer1.page.getByTestId('disconnect-btn').click(); + await expect(peer1.page.getByTestId('connection-state')).toContainText('idle'); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); +}); + +test.describe('Recording Metadata', () => { + test('should set recording MIME type correctly', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `recording-mime-${Date.now()}`; + try { + await connectPeer(page, roomId, { enableVideo: true, enableAudio: true }); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Initial MIME should be N/A + await expect(page.getByTestId('recording-mime')).toContainText('N/A'); + + // Start and stop recording + await page.getByTestId('start-recording-btn').click(); + await page.waitForTimeout(1000); + await page.getByTestId('stop-recording-btn').click(); + + // After recording, MIME type should be set (webm or mp4) + await expect(page.getByTestId('recording-mime')).not.toContainText('N/A', { + timeout: 5000, + }); + const mimeText = await page.getByTestId('recording-mime').innerText(); + expect(mimeText).toMatch(/video\/(webm|mp4)/); + } finally { + await context.close(); + } + }); + + test('should show recording size after recording stops', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `recording-size-${Date.now()}`; + try { + await connectPeer(page, roomId, { enableVideo: true, enableAudio: true }); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Initial size should be N/A + await expect(page.getByTestId('recording-size')).toContainText('N/A'); + + // Start recording + await page.getByTestId('start-recording-btn').click(); + await expect(page.getByTestId('recording-state')).toContainText('recording', { + timeout: 5000, + }); + + // Wait for recording to accumulate data + await page.waitForTimeout(2000); + + // Stop recording + await page.getByTestId('stop-recording-btn').click(); + + // Size should now show bytes and be > 0 + await expect(page.getByTestId('recording-size')).toContainText('bytes', { + timeout: 5000, + }); + const sizeText = await page.getByTestId('recording-size').innerText(); + const sizeMatch = sizeText.match(/(\d+) bytes/); + expect(sizeMatch).toBeTruthy(); + expect(Number(sizeMatch?.[1])).toBeGreaterThan(0); + } finally { + await context.close(); + } + }); +}); + +test.describe('Message Metadata', () => { + test('should display sent messages as local and received as remote', async ({ browser }) => { + const roomId = `message-metadata-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); + + // Wait for data channels to open + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + await expect(peer2.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + + // Peer 1 sends a message + await peer1.page.getByTestId('message-input').fill('Test message from peer 1'); + await peer1.page.getByTestId('send-btn').click(); + + // On peer 1, message should appear as local (data-testid="message-local") + await expect(peer1.page.getByTestId('message-local').first()).toContainText( + 'Test message from peer 1', + { timeout: 5000 } + ); + + // On peer 2, same message should appear as remote (data-testid="message-remote") + await expect(peer2.page.getByTestId('message-remote').first()).toContainText( + 'Test message from peer 1', + { timeout: 5000 } + ); + + // Verify the local message shows "You:" prefix + const localMessageText = await peer1.page.getByTestId('message-local').first().innerText(); + expect(localMessageText).toContain('You:'); + + // Verify the remote message shows "Remote" prefix + const remoteMessageText = await peer2.page + .getByTestId('message-remote') + .first() + .innerText(); + expect(remoteMessageText).toContain('Remote'); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); + + test('should include timestamps in messages', async ({ browser }) => { + const roomId = `message-timestamp-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); + + // Wait for data channels to open + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 10000, + }); + + // Send JSON message which includes timestamp + const beforeSend = Date.now(); + await peer1.page.getByTestId('send-json-btn').click(); + const afterSend = Date.now(); + + // Verify message appears on peer2 + await expect(peer2.page.getByTestId('message-remote').first()).toContainText('ping', { + timeout: 5000, + }); + + // The JSON message should contain a timestamp field + const messageContent = await peer2.page.getByTestId('message-remote').first().innerText(); + expect(messageContent).toContain('timestamp'); + + // Extract and verify timestamp is within expected range + const timestampMatch = messageContent.match(/"timestamp":\s*(\d+)/); + expect(timestampMatch).toBeTruthy(); + const timestamp = Number(timestampMatch?.[1]); + expect(timestamp).toBeGreaterThanOrEqual(beforeSend); + expect(timestamp).toBeLessThanOrEqual(afterSend + 1000); // Allow 1s tolerance + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); +}); + +test.describe('Connection Info Display', () => { + test('should display valid peer ID after connection', async ({ browser }) => { + const { context, page } = await createPeer(browser); + const roomId = `peer-id-display-${Date.now()}`; + try { + await connectPeer(page, roomId); + await expect(page.getByTestId('connection-state')).toContainText('signaling', { + timeout: 5000, + }); + + // Wait for peer ID to be assigned + await expect(page.getByTestId('peer-id')).not.toContainText('N/A', { timeout: 5000 }); + + // Verify peer ID format (should be non-empty string) + const peerIdText = await page.getByTestId('peer-id').innerText(); + const peerId = peerIdText.replace('My Peer ID: ', '').trim(); + + // Peer ID should not be empty and should have reasonable length + expect(peerId.length).toBeGreaterThan(0); + expect(peerId).not.toBe('N/A'); + expect(peerId).not.toBe('null'); + expect(peerId).not.toBe('undefined'); + } finally { + await context.close(); + } + }); + + test('should show both local and remote peer IDs when connected', async ({ browser }) => { + const roomId = `peer-ids-both-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + await Promise.all([connectPeer(peer1.page, roomId), connectPeer(peer2.page, roomId)]); + + // Wait for full connection with data channel open + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 15000, + }); + await expect(peer2.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 15000, + }); + + // Verify both peers have valid peer IDs (not N/A) + await expect(peer1.page.getByTestId('peer-id')).not.toContainText('N/A', { + timeout: 5000, + }); + await expect(peer2.page.getByTestId('peer-id')).not.toContainText('N/A', { + timeout: 5000, + }); + + // Get peer IDs + const peer1IdText = await peer1.page.getByTestId('peer-id').innerText(); + const peer1Id = peer1IdText.replace('My Peer ID: ', '').trim(); + const peer2IdText = await peer2.page.getByTestId('peer-id').innerText(); + const peer2Id = peer2IdText.replace('My Peer ID: ', '').trim(); + + // Peer IDs should be different (unique per peer) + expect(peer1Id).not.toBe(peer2Id); + expect(peer1Id.length).toBeGreaterThan(5); + expect(peer2Id.length).toBeGreaterThan(5); + + // Verify connectivity by sending a message - this proves peers see each other + await peer1.page.getByTestId('message-input').fill('verification message'); + await peer1.page.getByTestId('send-btn').click(); + await expect(peer2.page.getByTestId('message-remote').first()).toContainText( + 'verification message', + { timeout: 5000 } + ); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); +}); + +test.describe('Data Channel State Display', () => { + test('should show correct data channel state text for both peers', async ({ browser }) => { + const roomId = `data-channel-display-${Date.now()}`; + const peer1 = await createPeer(browser); + const peer2 = await createPeer(browser); + try { + // Initially check data channel is closed + await peer1.page.goto('/webrtc'); + await waitForPageLoad(peer1.page); + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Closed'); + + // Connect both peers + await peer1.page.getByTestId('room-id-input').clear(); + await peer1.page.getByTestId('room-id-input').fill(roomId); + await peer1.page.getByTestId('connect-btn').click(); + await connectPeer(peer2.page, roomId); + + // Wait for connection + await expect(peer1.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 15000, + }); + await expect(peer2.page.getByTestId('connection-state')).toContainText('connected', { + timeout: 15000, + }); + + // Both peers should show "Open" for data channel + await expect(peer1.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 5000, + }); + await expect(peer2.page.getByTestId('data-channel-state')).toContainText('Open', { + timeout: 5000, + }); + + // Verify exact text format + const peer1ChannelText = await peer1.page.getByTestId('data-channel-state').innerText(); + const peer2ChannelText = await peer2.page.getByTestId('data-channel-state').innerText(); + expect(peer1ChannelText).toContain('Data Channel:'); + expect(peer2ChannelText).toContain('Data Channel:'); + } finally { + await Promise.all([peer1.context.close(), peer2.context.close()]); + } + }); +}); From 33eb8b0d0b027b86c6df7bf1c510c8e700efb423 Mon Sep 17 00:00:00 2001 From: Jeff Haynie Date: Thu, 5 Feb 2026 21:04:49 -0800 Subject: [PATCH 18/20] fix(ci): optimize e2e test workflow to avoid timeout - Increase CI timeout from 10 to 15 minutes for 44 tests - Skip redundant rebuild in CI (SDK already built by prepare-sdk-for-testing.sh) - Increase workers from 1 to 4 in CI (tests use unique room IDs, safe to parallelize) - Run playwright directly instead of through test-e2e.sh wrapper --- .github/workflows/package-smoke-test.yaml | 4 ++-- playwright.config.ts | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/.github/workflows/package-smoke-test.yaml b/.github/workflows/package-smoke-test.yaml index 0ff2ed4da..622d94487 100644 --- a/.github/workflows/package-smoke-test.yaml +++ b/.github/workflows/package-smoke-test.yaml @@ -326,7 +326,7 @@ jobs: playwright-e2e-test: runs-on: blacksmith-4vcpu-ubuntu-2204 - timeout-minutes: 10 + timeout-minutes: 15 name: Playwright E2E Smoke Test steps: - uses: actions/checkout@v4 @@ -347,7 +347,7 @@ jobs: run: bash scripts/install-sdk-tarballs.sh apps/testing/e2e-web - name: Run Playwright E2E tests - run: bash scripts/test-e2e.sh + run: bun run test:e2e # TODO: Re-enable after artifact storage quota resets (disabled 2026-02-01) # - name: Upload Playwright report diff --git a/playwright.config.ts b/playwright.config.ts index 8ca349275..5002a278f 100644 --- a/playwright.config.ts +++ b/playwright.config.ts @@ -9,7 +9,7 @@ export default defineConfig({ fullyParallel: true, forbidOnly: !!process.env.CI, retries: process.env.CI ? 2 : 0, - workers: process.env.CI ? 1 : undefined, + workers: process.env.CI ? 4 : undefined, // CI runner has 4 vCPUs, tests use unique room IDs so can run in parallel reporter: 'html', use: { baseURL: 'http://localhost:3500', From d5e9278ce1ca0d9232ddd5d91c3f68266692309e Mon Sep 17 00:00:00 2001 From: Jeff Haynie Date: Thu, 5 Feb 2026 21:15:08 -0800 Subject: [PATCH 19/20] fix: remove unused variable and add workflow clarification - Remove unused peer2Id variable extraction in 'should clear remote peer ID' test - Add comment explaining why no explicit build step is needed (dev server compiles on-the-fly) --- .github/workflows/package-smoke-test.yaml | 3 +++ e2e/webrtc.pw.ts | 4 ---- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/.github/workflows/package-smoke-test.yaml b/.github/workflows/package-smoke-test.yaml index 622d94487..f88812aa0 100644 --- a/.github/workflows/package-smoke-test.yaml +++ b/.github/workflows/package-smoke-test.yaml @@ -346,6 +346,9 @@ jobs: - name: Install SDK in e2e-web run: bash scripts/install-sdk-tarballs.sh apps/testing/e2e-web + # No explicit build step needed: playwright.config.ts webServer starts + # "bun run dev" which compiles on-the-fly via Vite. A production build + # would be redundant since e2e tests run against the dev server. - name: Run Playwright E2E tests run: bun run test:e2e diff --git a/e2e/webrtc.pw.ts b/e2e/webrtc.pw.ts index 642b51e9a..4400cd6d3 100644 --- a/e2e/webrtc.pw.ts +++ b/e2e/webrtc.pw.ts @@ -910,10 +910,6 @@ test.describe('Peer List Updates', () => { timeout: 15000, }); - // Get peer2's ID directly from their page - const peer2IdText = await peer2.page.getByTestId('peer-id').innerText(); - const peer2Id = peer2IdText.replace('My Peer ID: ', ''); - // Peer2 disconnects await peer2.page.getByTestId('disconnect-btn').click(); From 04c8c3253b138992a9777847aafdf1d184532272 Mon Sep 17 00:00:00 2001 From: Jeff Haynie Date: Thu, 5 Feb 2026 21:28:40 -0800 Subject: [PATCH 20/20] fix(e2e): improve test robustness and fix misleading comments - Use regex for JSON assertion to tolerate whitespace variations - Fix comments to correctly reference peers[2] as third peer (not 'Peer 2') - Handle enableVideo/enableAudio false case explicitly in connectPeer helper - Replace weak negative assertion with positive regex match for error state --- e2e/webrtc.pw.ts | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/e2e/webrtc.pw.ts b/e2e/webrtc.pw.ts index 4400cd6d3..025375add 100644 --- a/e2e/webrtc.pw.ts +++ b/e2e/webrtc.pw.ts @@ -41,11 +41,17 @@ async function connectPeer( await waitForPageLoad(page); await page.getByTestId('room-id-input').clear(); await page.getByTestId('room-id-input').fill(roomId); - if (options?.enableVideo) { + // Explicitly handle video/audio checkbox state (both true and false) + // to ensure consistent state regardless of page defaults + if (options?.enableVideo === true) { await page.getByTestId('enable-video').check(); + } else if (options?.enableVideo === false) { + await page.getByTestId('enable-video').uncheck(); } - if (options?.enableAudio) { + if (options?.enableAudio === true) { await page.getByTestId('enable-audio').check(); + } else if (options?.enableAudio === false) { + await page.getByTestId('enable-audio').uncheck(); } if (options?.autoReconnect !== undefined) { await page.getByTestId('auto-reconnect-toggle').setChecked(options.autoReconnect); @@ -430,13 +436,13 @@ test.describe('Multi-Peer Mesh', () => { ) ); - // Peer 2 disconnects + // peers[2] (third peer) disconnects await peers[2].page.getByTestId('disconnect-btn').click(); await expect(peers[2].page.getByTestId('connection-state')).toContainText('idle', { timeout: 5000, }); - // Remaining peers should still be connected + // Remaining peers (peers[0] and peers[1]) should still be connected await expect(peers[0].page.getByTestId('connection-state')).toContainText('connected', { timeout: 10000, }); @@ -444,7 +450,7 @@ test.describe('Multi-Peer Mesh', () => { timeout: 10000, }); - // Peer 2 rejoins + // peers[2] (third peer) rejoins await peers[2].page.getByTestId('connect-btn').click(); await expect(peers[2].page.getByTestId('connection-state')).toContainText('connected', { timeout: 20000, @@ -545,7 +551,8 @@ test.describe('Screen Sharing', () => { await peer1.page.getByTestId('stop-screen-share-btn').click(); await expect(peer1.page.getByTestId('screen-share-state')).toContainText('Off'); - await expect(peer2.page.getByTestId('messages')).toContainText('active":false'); + // Use regex to tolerate whitespace variations in JSON formatting + await expect(peer2.page.getByTestId('messages')).toContainText(/"active"\s*:\s*false/); } finally { await Promise.all([peer1.context.close(), peer2.context.close()]); } @@ -790,9 +797,8 @@ test.describe('Error Handling', () => { await page.getByTestId('set-invalid-signal-url-btn').click(); await page.getByTestId('force-ws-close-btn').click(); - // Should show error or transition to a failed state - // The reconnect should be disabled, so it should fail - await expect(page.getByTestId('connection-state')).not.toContainText('connected', { + // With auto-reconnect disabled and an invalid signal URL, should reach idle or error state + await expect(page.getByTestId('connection-state')).toContainText(/idle|error/, { timeout: 10000, }); } finally {