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WebRTC and conference APIs for Node.js","maintainers":[{"name":"akirilyuk","email":"alexander.kiri.aa@googlemail.com"}],"readme":"# @node-webrtc-rust/sdk\n\nTypeScript SDK for **agentic voice workloads** and WebRTC in Node.js — backed by a Rust native engine ([NAPI-RS](https://napi.rs) + [webrtc-rs](https://github.com/webrtc-rs/webrtc)).\n\nBuild phone bots, browser voice assistants, and session workers where **Node runs your LLM and tools**, and **Rust owns media timing** (VAD, barge-in, STT/TTS vendors, outbound PCM at 20 ms cadence).\n\n```bash\nnpm install @node-webrtc-rust/sdk @node-webrtc-rust/signaling @node-webrtc-rust/helpers\n```\n\n## Exports\n\n| Import path                        | Purpose                                                                                         |\n| ---------------------------------- | ----------------------------------------------------------------------------------------------- |\n| **`@node-webrtc-rust/sdk/voice`**  | **VoiceAgent** — VAD, barge-in, STT/TTS vendors, speech events — [VOICE-API.md](./VOICE-API.md) |\n| `@node-webrtc-rust/sdk`            | W3C-style `RTCPeerConnection`, tracks, data channels                                            |\n| `@node-webrtc-rust/sdk/conference` | Conference room control plane (MCU mixing)                                                      |\n\n**WebRTC parity vs browser:** [`docs/webrtc-api-parity.md`](../../docs/webrtc-api-parity.md)\n\n---\n\n## Voice agent — build agentic workloads\n\nImport **`@node-webrtc-rust/sdk/voice`** when you need a conversational loop without reimplementing PCM timing or vendor HTTP/WebSocket clients in Node.\n\n**API reference:** [VOICE-API.md](./VOICE-API.md) (exports, speech events, data-channel bridge) · [VOICE-VAD-AND-BARGE-IN.md](./VOICE-VAD-AND-BARGE-IN.md) (tuning)\n\n### Problem this solves\n\n| Without VoiceAgent                          | With VoiceAgent                                                 |\n| ------------------------------------------- | --------------------------------------------------------------- |\n| Manual Opus decode + frame alignment        | Inbound loop via `RemoteAudioTrack.readSample()`                |\n| Roll your own VAD / interrupt handling      | Configurable VAD + atomic TTS flush on barge-in                 |\n| STT/TTS SDK calls in Node + glue to WebRTC  | Rust vendor adapters; Node gets text events + `sendTextToTTS()` |\n| Race between user speech and agent playback | Native buffer flush **before** `barge_in` reaches JS            |\n\n### Pipeline B (recommended)\n\nSTT final text **up** to Node → your LLM → TTS text **down** to Rust → outbound WebRTC track:\n\n```typescript\nimport { LocalAudioTrack, RTCPeerConnection } from '@node-webrtc-rust/sdk'\nimport { VoiceAgent } from '@node-webrtc-rust/sdk/voice'\n\n// 1. Connect WebRTC (browser, SIP gateway, or another Node peer)\nconst pc = new RTCPeerConnection({ iceServers: [...] })\nconst agentTrack = new LocalAudioTrack('agent', 'session-1')\nawait pc.addTrack(agentTrack)\n\n// Resolve tracks after negotiation (see examples/voice-agent/)\nlet userTrack: RemoteAudioTrack\npc.ontrack = (e) => { if (e.track.kind === 'audio') userTrack = e.track as RemoteAudioTrack }\n\n// 2. Configure one VoiceAgent per conversation (VAD preset = good defaults for phone bots)\nimport { VOICE_AGENT_VAD_PRESET } from '@node-webrtc-rust/sdk/voice'\n\nconst agent = new VoiceAgent({\n  vad: VOICE_AGENT_VAD_PRESET,\n  stt: { provider: 'deepgram', model: 'nova-2', language: 'en' },\n  tts: { provider: 'openai', model: 'tts-1', voice: 'alloy' },\n  events: { mode: 'both' },\n})\n\n// 3. Bind tracks — one attach() = one session\nawait agent.attach({ inboundTrack: userTrack, outboundTrack: agentTrack })\nawait agent.start()\n\n// 4. Agent loop — your business logic stays in TypeScript\nagent.on('user_speech_final', async (event) => {\n  const stream = await myLLM.chat(event.text!)\n  for await (const token of stream) {\n    await agent.sendTextToTTS(token)\n  }\n})\n\nagent.on('barge_in', () => {\n  cancelLLMStream()\n  // flushTts already ran in Rust if bargeIn.flushTts === true\n})\n```\n\n### Attach model\n\n`attach({ inboundTrack, outboundTrack })` binds **one peer connection session**:\n\n| Track           | Direction                         | Used for                        |\n| --------------- | --------------------------------- | ------------------------------- |\n| `inboundTrack`  | User → agent (`RemoteAudioTrack`) | VAD + STT (`readSample` loop)   |\n| `outboundTrack` | Agent → user (`LocalAudioTrack`)  | TTS PCM after `sendTextToTTS()` |\n\nMultiple concurrent callers = multiple `VoiceAgent` instances (one per WebRTC connection). Use [`@node-webrtc-rust/helpers`](../../helpers/README.md) `SessionPod` to run many sessions on one server process.\n\n### Server helpers (`@node-webrtc-rust/helpers`)\n\n| Helper                  | Use when                                                                            |\n| ----------------------- | ----------------------------------------------------------------------------------- |\n| `SessionPod`            | One signaling entry point, many concurrent sessions (your host assigns `sessionId`) |\n| `VoiceAgentSessionHost` | One signaling room; auto-spawns one agent per `client-*` peer                       |\n| `createKickFrame`       | Prime outbound RTP before TTS PCM                                                   |\n\nRunnable demo: [`examples/voice-agent-multi-session-pod`](../../examples/voice-agent-multi-session-pod/README.md).\n\n### Configuration reference\n\n```typescript\ninterface VoiceAgentConfig {\n  vad?: {\n    enabled?: boolean // default true\n    provider?: 'energy' | 'silero' // default energy; silero needs custom build + local ONNX Runtime — see VOICE-VAD-AND-BARGE-IN.md\n    threshold?: number // energy ~0.05–0.2; silero probability ~0.5\n    minSpeechDurationMs?: number\n    minSilenceDurationMs?: number\n    speechPadMs?: number // pre-roll ring capacity (ms), default 300\n    sampleRate?: 8000 | 16000\n    bargeIn?: {\n      enabled?: boolean // master switch (default true)\n      useVad?: boolean // auto barge on VAD SpeechStart (default true); false = flushTts() only\n      flushTts?: boolean // clear TTS buffer when barge-in runs (default true)\n      requireSttPartial?: boolean // while agent TTS plays: barge only after STT partial (default true)\n      minSttPartialTokens?: number // min whitespace tokens for semantic barge (default 2)\n      minSttPartialChars?: number // legacy alias for minSttPartialTokens (same meaning)\n      agentPlaybackGuardMs?: number // optional ms to ignore VAD barge after TTS starts (default 0)\n    }\n    gateStt?: boolean // only feed STT while gate is open (default false)\n    gateSttOpenOnPending?: boolean // also open gate during VAD pending speech (default true)\n    sttGateHoldMs?: number // keep feeding STT after speech end (default 1000)\n  }\n  stt?: {\n    provider: 'openai' | 'deepgram' | 'google' | 'assemblyai' | 'local-sherpa' | 'mock'\n    model?: string\n    modelPath?: string // local-sherpa: ONNX model directory\n    language?: string\n    apiKey?: string // or env var — see vendor table\n  }\n  tts?: {\n    provider: 'openai' | 'elevenlabs' | 'google' | 'cartesia' | 'local-sherpa' | 'mock'\n    model?: string\n    modelPath?: string // local-sherpa: Piper/VITS directory\n    voice?: string\n    apiKey?: string\n  }\n  events?: {\n    mode?: 'callback' | 'stream' | 'both' // default 'both'\n  }\n}\n```\n\n### STT/TTS vendors\n\nProviders are **mix-and-match** per session. **Official API docs:** [`examples/shared/VOICE_VENDOR_REFERENCE.md`](../../examples/shared/VOICE_VENDOR_REFERENCE.md)\n\n| Provider       | STT | TTS | Typical env var                                                         | API docs                                                                                                                       |\n| -------------- | --- | --- | ----------------------------------------------------------------------- | ------------------------------------------------------------------------------------------------------------------------------ |\n| `openai`       | ✓   | ✓   | `OPENAI_API_KEY`                                                        | [STT](https://platform.openai.com/docs/guides/speech-to-text) · [TTS](https://platform.openai.com/docs/guides/text-to-speech)  |\n| `deepgram`     | ✓   | —   | `DEEPGRAM_API_KEY`                                                      | [Live streaming](https://developers.deepgram.com/docs/live-streaming-audio)                                                    |\n| `elevenlabs`   | —   | ✓   | `ELEVENLABS_API_KEY`                                                    | [TTS API](https://elevenlabs.io/docs/api-reference/text-to-speech/convert)                                                     |\n| `cartesia`     | —   | ✓   | `CARTESIA_API_KEY`                                                      | [TTS bytes](https://docs.cartesia.ai/api-reference/tts/bytes)                                                                  |\n| `assemblyai`   | ✓   | —   | `ASSEMBLYAI_API_KEY`                                                    | [Streaming STT](https://www.assemblyai.com/docs/speech-to-text/streaming)                                                      |\n| `google`       | ✓   | ✓   | `GOOGLE_APPLICATION_CREDENTIALS`                                        | [STT](https://cloud.google.com/speech-to-text/docs) · [TTS](https://cloud.google.com/text-to-speech/docs)                      |\n| `local-sherpa` | ✓   | ✓   | `SHERPA_STT_MODEL_PATH`, `SHERPA_TTS_MODEL_PATH`, `SHERPA_STT_LANGUAGE` | [Sherpa-ONNX](https://k2-fsa.github.io/sherpa/onnx/) · [Models](https://github.com/k2-fsa/sherpa-onnx/releases/tag/asr-models) |\n| `mock`         | ✓   | ✓   | _(none — use for CI/local)_                                             | —                                                                                                                              |\n\nExample pairings when a vendor only supports one direction:\n\n```typescript\n// Deepgram listen + OpenAI speech\nstt: { provider: 'deepgram', model: 'nova-2', language: 'en' },\ntts: { provider: 'openai', model: 'tts-1', voice: 'alloy' },\n\n// AssemblyAI STT + ElevenLabs TTS\nstt: { provider: 'assemblyai', model: 'universal-streaming-english' },\ntts: { provider: 'elevenlabs', model: 'eleven_multilingual_v2', voice: '...' },\n```\n\nAPI keys via `apiKey` in config or env vars. Never logged or returned in speech events.\n\n#### Local Sherpa-ONNX (`local-sherpa`) — free on-device STT + TTS\n\n**Recommended when privacy and speech latency matter.** User speech is transcribed and agent replies synthesized on **your server** with Sherpa-ONNX — no cloud STT/TTS API keys or network round-trips.\n\nRequires downloaded **Zipformer transducer** (STT) and **Piper/VITS** (TTS) directories.\n\n| Variable                | Required      | Purpose                                                                  |\n| ----------------------- | ------------- | ------------------------------------------------------------------------ |\n| `SHERPA_STT_MODEL_PATH` | **Yes** (STT) | Path to extracted Zipformer STT directory                                |\n| `SHERPA_TTS_MODEL_PATH` | **Yes** (TTS) | Path to Piper/VITS directory (`tokens.txt`, `*.onnx`, `espeak-ng-data/`) |\n| `SHERPA_STT_LANGUAGE`   | No            | `stt.language` tag (inferred from path when omitted)                     |\n| `SHERPA_TTS_SPEAKER`    | No            | Piper speaker id for `tts.voice` (default `0`)                           |\n\nAutomatic downloads (from repo root):\n\n```bash\nnpm run download-stt --workspace=@node-webrtc-rust/example-voice-agent-local-sherpa\nnpm run download-tts --workspace=@node-webrtc-rust/example-voice-agent-local-sherpa\nexport SHERPA_STT_MODEL_PATH=\"$PWD/examples/voice-agent-local-sherpa/.models/sherpa-onnx-streaming-zipformer-en-kroko-2025-08-06\"\nexport SHERPA_TTS_MODEL_PATH=\"$PWD/examples/voice-agent-local-sherpa/.models/vits-piper-en_US-amy-low\"\n```\n\nSupported languages, bundle names, and unavailable locales (Hindi, Portuguese, Italian): [`examples/shared/VOICE_VENDOR_REFERENCE.md`](../../examples/shared/VOICE_VENDOR_REFERENCE.md#local-sherpa-onnx--multilingual-models). Full walkthrough: [`examples/voice-agent-local-sherpa/README.md`](../../examples/voice-agent-local-sherpa/README.md).\n\n```typescript\nstt: {\n  provider: 'local-sherpa',\n  language: process.env.SHERPA_STT_LANGUAGE ?? 'en',\n  modelPath: process.env.SHERPA_STT_MODEL_PATH,\n},\ntts: {\n  provider: 'local-sherpa',\n  modelPath: process.env.SHERPA_TTS_MODEL_PATH,\n  voice: process.env.SHERPA_TTS_SPEAKER ?? '0',\n},\n```\n\n### Speech events\n\nThree layers — **VAD activity**, **STT session / vendor feed**, and **text**:\n\n| Event                                 | Source                            | Use in your agent                              |\n| ------------------------------------- | --------------------------------- | ---------------------------------------------- |\n| `vad_triggered`                       | VAD `SpeechStart`                 | STT listen opened; logging / `[speech]` traces |\n| `user_stt_start` / `user_stt_end`     | STT session                       | Recognition session boundaries                 |\n| `stt_stream_start` / `stt_stream_end` | STT vendor PCM gate               | Vendor feed on / off per utterance             |\n| `user_stt_not_found`                  | C1 timeout (`sttListenTimeoutMs`) | VAD fired but no partial — no LLM turn         |\n| `user_speaking_start`                 | VAD `SpeechStart`                 | User began talking (inbound voice activity)    |\n| `user_speaking_end`                   | VAD + `sttGateHoldMs` (`gateStt`) | End-of-utterance hint (paired with final)      |\n| `user_speech_partial`                 | STT                               | Live captions, semantic barge-in               |\n| `user_speech_final`                   | STT (or C2 forced)                | **Primary LLM turn trigger**                   |\n| `agent_speaking_start` / `end`        | TTS playback                      | UI / state machine                             |\n| `barge_in`                            | Barge-in path (see below)         | Cancel agent TTS (after optional flush)        |\n| `error`                               | Any                               | Vendor or pipeline failure                     |\n\nLifecycle detail: [VOICE-VAD-AND-BARGE-IN.md § STT utterance lifecycle](./VOICE-VAD-AND-BARGE-IN.md#stt-utterance-lifecycle-vad--stt-events).\n\n### VAD and barge-in\n\n**Full guide:** [`VOICE-VAD-AND-BARGE-IN.md`](./VOICE-VAD-AND-BARGE-IN.md) — **energy vs Silero**, weight/comparison, use cases, defaults, when to tune.\n\n**Quick start:** use `VOICE_AGENT_VAD_PRESET` (or omit `vad` and only set `gateStt: true` if you stream STT). Defaults: 250 ms min speech, 500 ms min silence, **`requireSttPartial: true`** (barge during agent TTS only after STT recognizes words), `sttGateHoldMs` 1000. See [VOICE-VAD-AND-BARGE-IN.md](./VOICE-VAD-AND-BARGE-IN.md#semantic-barge-in-requiresttpartial-default-true) and [STT flow fine-tuning](./VOICE-VAD-AND-BARGE-IN.md#stt-flow-fine-tuning-gatestt).\n\n| Export                    | Use                                              |\n| ------------------------- | ------------------------------------------------ |\n| `VOICE_AGENT_VAD_PRESET`  | Recommended — `gateStt: true` + default barge-in |\n| `DEFAULT_VOICE_AGENT_VAD` | Rust defaults (`gateStt: false`)                 |\n\nBarge-in requires `vad.enabled` and `bargeIn.useVad` (both default `true`). With **`requireSttPartial: true`** (preset default), interrupt during **agent TTS** waits for a qualifying `user_speech_partial` before `barge_in` — coughs and tones alone do not cut playback. Set `requireSttPartial: false` for immediate energy-VAD barge (legacy). Set `useVad: false` and call `flushTts()` for manual interrupt only.\n\n**Callback delivery:**\n\n```typescript\nagent.on('user_speech_final', (event) => {\n  /* event.text */\n})\nagent.off('user_speech_final', handler)\n```\n\n**Stream delivery:**\n\n```typescript\nfor await (const event of agent.speechEvents()) {\n  switch (event.type) {\n    case 'user_speech_final':\n      await handleTurn(event.text!)\n      break\n    case 'barge_in':\n      await cancelTurn()\n      break\n  }\n}\n```\n\nSet `events.mode: 'both'` to use handlers and the iterator on the same session.\n\n### Browser client over DataChannel\n\nWhen the agent runs on Node and the user connects from a browser, mirror speech events and accept TTS requests on an `RTCDataChannel`:\n\n```typescript\nimport {\n  VOICE_CONTROL_CHANNEL_LABEL,\n  wireVoiceAgentToDataChannel,\n} from '@node-webrtc-rust/sdk/voice'\n\nconst dc = pc.createDataChannel(VOICE_CONTROL_CHANNEL_LABEL, { ordered: true })\ndc.onopen = () => {\n  wireVoiceAgentToDataChannel(agent, dc)\n}\n```\n\n- **Server → client:** `{ type: 'speech_event', event: 'user_speech_final', text: '…' }` (STT, VAD, `barge_in`, …)\n- **Client → server:** `{ type: 'speak', text: 'Hello' }` → `agent.sendTextToTTS(text)`\n\nFull browser + Node demo: [`examples/voice-agent-browser`](../../examples/voice-agent-browser/README.md).\n\n**Live vendors:** default is mock. Set `VOICE_VENDOR=openai|deepgram|elevenlabs|cartesia|assemblyai|google` and the vendor’s API keys, then run `npm run start:live:<vendor> --workspace=@node-webrtc-rust/example-voice-agent-browser`. See that README for env vars and pairing notes.\n\n### API summary\n\n| Method                                    | Description                                                          |\n| ----------------------------------------- | -------------------------------------------------------------------- |\n| `attach({ inboundTrack, outboundTrack })` | Bind one PC session                                                  |\n| `start()`                                 | Start VAD/STT pipeline and inbound PCM loop                          |\n| `stop()`                                  | Stop pipeline; ends `speechEvents()` iterator                        |\n| `sendTextToTTS(text)`                     | Synthesize via TTS vendor → outbound track                           |\n| `flushTts()`                              | Clear pending TTS (also used when `bargeIn.flushTts: false`)         |\n| `on(type, fn)` / `off`                    | Callback handlers                                                    |\n| `speechEvents()`                          | Async generator of all speech events                                 |\n| `wireVoiceAgentToDataChannel(agent, dc)`  | Forward speech events + handle `{ type: 'speak' }` on a data channel |\n\n### Examples and live vendor testing\n\nFrom the repo root (after `npm run setup`):\n\n| Command                                                                                                      | Purpose                                        |\n| ------------------------------------------------------------------------------------------------------------ | ---------------------------------------------- |\n| `npm run start:callback --workspace=@node-webrtc-rust/example-voice-agent`                                   | Mock vendors, `on()` handlers                  |\n| `npm run start:stream --workspace=...`                                                                       | Mock vendors, `speechEvents()`                 |\n| `npm run start:barge-in --workspace=...`                                                                     | VAD + `flushTts`                               |\n| `npm run start --workspace=@node-webrtc-rust/example-voice-agent-browser`                                    | Browser mic + DataChannel events + TTS control |\n| `OPENAI_API_KEY=sk-... npm run start:live:openai --workspace=@node-webrtc-rust/example-voice-agent-browser`  | Browser demo with live OpenAI STT/TTS          |\n| `npm run start:live:deepgram` / `elevenlabs` / … `--workspace=@node-webrtc-rust/example-voice-agent-browser` | Live browser demo per vendor (see README)      |\n| `OPENAI_API_KEY=sk-... npm run start:live:openai --workspace=...`                                            | Live OpenAI preset                             |\n| `npm run start:live:deepgram` / `elevenlabs` / …                                                             | Per-vendor manual tests                        |\n\nSee [`examples/voice-agent/README.md`](../../examples/voice-agent/README.md) for env vars and track-direction notes.\n\n**SDK live tests** (opt-in):\n\n```bash\nVOICE_LIVE_TEST=1 VOICE_LIVE_OPENAI=1 OPENAI_API_KEY=sk-... \\\n  npm run test --workspace=@node-webrtc-rust/sdk -- voice-live\n```\n\nUse **`mock`** providers for CI and learning the API. Live vendor HTTP/WebSocket runs in Rust `vendor-*` crates; default builds use stubs until optional `live` features are enabled.\n\n---\n\n## WebRTC core\n\nLow-level transport for attaching `VoiceAgent` to browsers, telephony gateways, or Node peers:\n\n```typescript\nimport { RTCPeerConnection, LocalAudioTrack, RemoteAudioTrack } from '@node-webrtc-rust/sdk'\nimport { SignalingServer, SignalingClient, autoNegotiate } from '@node-webrtc-rust/signaling'\n```\n\n**Recent APIs:** `addTransceiver`, `getTransceivers` / `getSenders` / `getReceivers`, `RTCRtpSender.replaceTrack`, `removeTrack`, `RemoteAudioTrack.readSample`, `getStats`, `setConfiguration`, `closeAsync()` (awaitable native cleanup; `close()` stays W3C-compatible).\n\nPeer connection quick start: [root README](../../README.md#webrtc-core-and-conference).\n\nPCM conventions (kick frame, 20 ms frames): [`@node-webrtc-rust/helpers/pcm`](../../packages/helpers/README.md#pcm-kick-frame) (also re-exported from [`examples/shared/pcm-streaming.ts`](../../examples/shared/pcm-streaming.ts)).\n\n---\n\n## Conference rooms\n\nMulti-participant MCU mixing — separate from single-session `VoiceAgent`, but useful for group voice apps.\n\nConference APIs live under **`@node-webrtc-rust/sdk/conference`**. TypeScript manages room lifecycle and mute policy; Rust mixes Opus/PCM in `crates/conference`.\n\n### Control plane vs data plane\n\n| Layer         | Package / crate                  | Responsibility                              |\n| ------------- | -------------------------------- | ------------------------------------------- |\n| Control plane | `sdk/conference`                 | Room lifecycle, mute/kick, signaling bridge |\n| Signaling     | `@node-webrtc-rust/signaling`    | WebSocket SDP/ICE relay                     |\n| Data plane    | `bindings` → `crates/conference` | Peer connections, mixer graph               |\n\n### Quick start\n\n```typescript\nimport { createServer } from 'http'\n\nimport { ConferenceServer } from '@node-webrtc-rust/sdk/conference'\nimport { SignalingServer } from '@node-webrtc-rust/signaling'\n\nconst PORT = 3000\nconst httpServer = createServer()\nconst signaling = new SignalingServer({ server: httpServer, path: '/ws' })\nawait signaling.listen(PORT)\n\nconst conference = new ConferenceServer()\nconference.attachSignaling({ url: `ws://127.0.0.1:${PORT}/ws` })\n\nconference.on('participant-joined', ({ roomId, participantId }) => {\n  console.log(`joined ${participantId} in ${roomId}`)\n})\n\nawait conference.createRoom('demo', {\n  maxParticipants: 16,\n  iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],\n})\n```\n\nSee [`examples/conference-room`](../../examples/conference-room/) for a browser demo.\n\n### Mute modes\n\n| Mode                  | API                                                      | Effect                                        |\n| --------------------- | -------------------------------------------------------- | --------------------------------------------- |\n| **Global mute**       | `muteParticipant(id, { scope: 'global' })`               | Target excluded from **all** listeners' mixes |\n| **Listener mute**     | `muteParticipant(id, { scope: 'listener', listenerId })` | Only `listenerId` stops hearing `id`          |\n| **Room-wide silence** | `setMixingEnabled(false)`                                | Mixer outputs silence; mute state preserved   |\n\n### Authentication (production)\n\nThe library does not enforce roles. Gate admin actions in your app:\n\n| Action                          | Recommended policy         |\n| ------------------------------- | -------------------------- |\n| Global mute / kick / mixing off | Admin / moderator only     |\n| Listener-scoped mute            | Owning client (preference) |\n\n### Events\n\n`ConferenceServer` emits: `room-created`, `room-destroyed`, `participant-joined`, `participant-left`, `participant-kicked`, `participant-muted`, `mixing-enabled-changed`, `error`.\n\n---\n\n## Debug logging\n\n```bash\nWEBRTC_DEBUG=1 node your-agent.js\n```\n\nTraces SDK, bindings, and Rust core on stderr with `[webrtc-debug]` prefix.\n\n---\n\n## License\n\nMIT\n","readmeFilename":"README.md"}