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server-side helpers for voice agents, session pods, and PCM streaming","maintainers":[{"name":"akirilyuk","email":"alexander.kiri.aa@googlemail.com"}],"readme":"# @node-webrtc-rust/helpers\n\nReusable **server-side** TypeScript helpers for voice agent apps — the same abstractions used in the examples, published as a package so you can skip copy-pasting negotiation boilerplate.\n\n## Install\n\n````bash\nnpm install @node-webrtc-rust/helpers @node-webrtc-rust/sdk @node-webrtc-rust/signaling\n\n## Tests\n\nFrom the **repo root** (not inside `packages/helpers/`):\n\n```bash\nnpm run test:helpers\n````\n\nCI runs the same via [`scripts/ci/run-helpers-unit-tests.sh`](../../scripts/ci/run-helpers-unit-tests.sh) in the **Typecheck & lint** job (`build.yml` on PRs, `build-main.yml` on push to `main`). That script is **unit only** (`*.integration.test.ts` excluded — those need a native `.node` and run in the Test job via `npm run test:integration --workspace=@node-webrtc-rust/helpers`). Before push (lint + helpers vitest when those paths changed):\n\n```bash\nnpm run ci:pre-push\n```\n\n````\n\n## What's included\n\n| Export                                 | Purpose                                                               |\n| -------------------------------------- | --------------------------------------------------------------------- |\n| `SessionPod`                           | One Node process, one signaling entry point, many concurrent sessions |\n| `VoiceAgentSessionHost`                | One signaling room; spawns one `VoiceAgent` + PC per browser client   |\n| `startMultiClientVoiceServer`          | One room, many tabs — wraps signaling + host + `/api/capacity`        |\n| `VoiceAgentSessionHost.broadcastSpeak` | TTS `text` on every connected client in the room                      |\n| `VoiceSessionHandler`                  | Per-tab hooks: transport/agent lifecycle, STT/VAD, speak requests     |\n| `VoiceSessionBudget`                   | Process-wide cap (`VOICE_MAX_CONCURRENT_SESSIONS`)                    |\n| `VOICE_AGENT_SERVER_PEER_ID`           | Stable server peer id for signaling joins                             |\n| `createKickFrame`, PCM constants       | RTP prime / 20 ms frame conventions                                   |\n\n## Multi-session pod (recommended server pattern)\n\n```typescript\nimport { createServer } from 'http'\n\nimport { SignalingServer } from '@node-webrtc-rust/signaling'\nimport { SessionPod } from '@node-webrtc-rust/helpers'\n\nconst PORT = 3003\nconst httpServer = createServer()\nconst signaling = new SignalingServer({ server: httpServer, path: '/ws' })\nawait signaling.listen(PORT)\n\nconst pod = new SessionPod(signaling, {\n  signalingUrl: `ws://127.0.0.1:${PORT}/ws`,\n  iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],\n  voiceConfig: { stt: { provider: 'mock' }, tts: { provider: 'mock' } },\n})\n\n// Orchestrator assigns a session id per call\nawait pod.ensureSession('call-abc123')\n\n// Browser client joins signaling room `call-abc123` as `client-*`\n// → pod creates RTCPeerConnection + VoiceAgent\n// → on hangup: agent.stop(), PC close, optional idle session teardown\n````\n\nRunnable demo: [`examples/voice-agent-multi-session-pod`](../../examples/voice-agent-multi-session-pod/).\n\n## Single room, many clients\n\nIf you only need one signaling room (not full pod orchestration):\n\n```typescript\nimport { SignalingClient } from '@node-webrtc-rust/signaling'\nimport { VOICE_AGENT_SERVER_PEER_ID, VoiceAgentSessionHost } from '@node-webrtc-rust/helpers'\n\nconst signaling = new SignalingClient({\n  url: 'ws://127.0.0.1:3001/ws',\n  room: 'demo',\n  peerId: VOICE_AGENT_SERVER_PEER_ID,\n})\nawait signaling.connect()\n\nconst host = new VoiceAgentSessionHost(signaling, iceServers, { voiceConfig })\n// Each `client-*` peer → own VoiceAgent; disconnect cleans up automatically\n```\n\nSee [`examples/voice-agent-browser`](../../examples/voice-agent-browser/).\n\n## Session budget (deployment sizing)\n\n```bash\nexport VOICE_MAX_CONCURRENT_SESSIONS=8\n```\n\n`VoiceAgentSessionHost` and `SessionPod` share one process-wide budget. When full, new `client-*` peers are rejected (no WebRTC offer). Expose metrics via `GET /api/capacity` when using `startMultiClientVoiceServer`.\n\nDemo: [`examples/voice-agent-local-sherpa-multi-client`](../../examples/voice-agent-local-sherpa-multi-client/) (three tabs, one room, local Sherpa; `npm run start` frees port 3004 via [`scripts/free-port.sh`](../../scripts/free-port.sh) — see example README).\n\n## Multi-client room (three agents, shared Sherpa)\n\n```typescript\nimport { startMultiClientVoiceServer } from '@node-webrtc-rust/helpers'\n\nconst server = await startMultiClientVoiceServer({\n  port: 3004,\n  room: 'sherpa-multi',\n  voiceConfig: {\n    stt: { provider: 'local-sherpa', modelPath },\n    tts: { provider: 'local-sherpa', modelPath },\n  },\n  iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],\n  serveHttp: myStaticHandler,\n})\n// server.budget → { active, max, available, rejectedTotal }\n```\n\nEach browser tab = one `VoiceAgent`; Sherpa ONNX weights are pooled in the native layer.\n\n### Custom STT / TTS logic\n\nPass a `VoiceSessionHandler` (or set `hostOptions.voiceHandler` on the host). Use `ctx.speak(text)` to play TTS on that tab's outbound track.\n\nPeer lifecycle has two phases:\n\n1. **`onPeerTransportReady`** — `RTCPeerConnection` is `connected` and the voice-control DataChannel is open. Safe for billing/capacity; VoiceAgent may not be attached yet (do not call `ctx.speak` here in voice mode).\n2. **`onPeerConnected`** — voice: after `VoiceAgent.attach()` + `VoiceAgent.start()` (safe for `ctx.speak`); data-only: immediately after transport-ready.\n\nTeardown after transport-ready uses `onPeerDisconnected`. Only peers that never reached transport-ready get `onPeerSignalingLost`.\n\n```typescript\nimport { startMultiClientVoiceServer, type VoiceSessionHandler } from '@node-webrtc-rust/helpers'\n\nconst voiceHandler: VoiceSessionHandler = {\n  async onPeerTransportReady(ctx) {\n    // PC + control channel ready — optional billing/capacity hook\n    console.log('transport ready', ctx.peerId)\n  },\n  async onPeerConnected(ctx) {\n    // VoiceAgent started — safe to speak\n    await ctx.speak('ready')\n  },\n  async onSpeechEvent(ctx, event) {\n    if (event.type === 'user_speech_final' && event.text) {\n      await ctx.speak(`You said: ${event.text}`)\n    }\n  },\n  async onSpeakRequest(ctx, text) {\n    await ctx.speak(text)\n  },\n}\n\nawait startMultiClientVoiceServer({\n  port: 3004,\n  room: 'demo',\n  voiceConfig,\n  iceServers,\n  voiceHandler,\n})\n```\n\nRunnable template: edit **`examples/voice-agent-local-sherpa-multi-client/src/voice-handler.ts`** only — see that example's README.\n\n## PCM kick frame\n\n```typescript\nimport { createKickFrame, PCM_KICK_DURATION_MS } from '@node-webrtc-rust/helpers/pcm'\n\nawait localTrack.writeSample(createKickFrame(), PCM_KICK_DURATION_MS)\n```\n\n## Design notes\n\n- **One `VoiceAgent` per WebRTC connection** — intentional; no multiplexing inside the agent.\n- **Transport before agent** — billing/capacity can use `onPeerTransportReady`; app TTS belongs in `onPeerConnected`.\n- **Disconnect = cleanup** — `VoiceAgent.stop()` and awaitable `RTCPeerConnection.closeAsync()` run in `VoiceAgentSessionHost`. Session budget slots are released only after native peer cleanup completes (or a bounded timeout), so capacity is not reused while TURN/ICE sockets are still draining.\n- **Scale pods by CPU/RAM**, not one process per call — see multi-session pod example README.\n","readmeFilename":"README.md"}