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Ekdahl"},"license":"MIT","homepage":"https://sigx.dev/lynx/modules/webrtc/overview/","keywords":["signalx","sigx","lynx","mobile","ios","android","webrtc","rtc","peerconnection","voip","realtime"],"repository":{"url":"git+https://github.com/signalxjs/lynx.git","type":"git","directory":"packages/lynx-webrtc"},"description":"W3C-shaped WebRTC for sigx-lynx: peer connections, audio tracks, data channels","maintainers":[{"name":"andtii","email":"andreas.ekdahl@omniaintranet.com"}],"readme":"# @sigx/lynx-webrtc\n\nW3C-shaped WebRTC for sigx-lynx: peer connections, microphone audio tracks, and\ndata channels, backed by libwebrtc ([webrtc-sdk](https://github.com/webrtc-sdk)\nbuilds). Code written against browser WebRTC ports nearly unchanged.\n\n> **Platforms:** Android + iOS.\n> Not available on the web runtime (the background thread has no\n> `RTCPeerConnection`) — gate with `isWebRTCAvailable()`.\n\n## 📚 Documentation\n\nFull guides, API reference and live examples → **[https://sigx.dev/lynx/modules/webrtc/overview/](https://sigx.dev/lynx/modules/webrtc/overview/)**\n\n## Install\n\n```sh\npnpm add @sigx/lynx-webrtc\nsigx prebuild\n```\n\nAutolinking adds the native SDK dependency (gradle dep / `WebRTC-SDK` pod),\nthe Android `RECORD_AUDIO` / `MODIFY_AUDIO_SETTINGS` permissions, the iOS\nmicrophone usage description, and the iOS `audio` background mode (so calls\nsurvive backgrounding). Note the native SDK adds roughly 5–10 MB per ABI on\nAndroid and ~10 MB on iOS.\n\n## Usage\n\n```ts\nimport { RTCPeerConnection, mediaDevices } from '@sigx/lynx-webrtc';\n\n// 1. Capture the mic (shows the OS permission prompt when needed).\nconst stream = await mediaDevices.getUserMedia({ audio: true });\nconst [micTrack] = stream.getAudioTracks();\n\n// 2. Wire up a peer connection.\nconst peer = new RTCPeerConnection({ iceServers: [{ urls: 'stun:stun.example.com' }] });\npeer.addTrack(micTrack, stream);\nconst events = peer.createDataChannel('events');\nevents.onmessage = e => console.log('peer says', e.data);\n\n// 3. Negotiate. For an HTTP (non-trickle) SDP exchange, wait for ICE\n//    gathering to complete so localDescription carries every candidate.\nconst offer = await peer.createOffer();\nawait peer.setLocalDescription(offer);\nawait new Promise<void>(resolve => {\n    if (peer.iceGatheringState === 'complete') return resolve();\n    peer.onicegatheringstatechange = () => {\n        if (peer.iceGatheringState === 'complete') resolve();\n    };\n});\nconst answerSdp = await postOfferSomewhere(peer.localDescription!.sdp);\nawait peer.setRemoteDescription({ type: 'answer', sdp: answerSdp });\n\n// 4. Remote audio plays automatically — no element to attach.\npeer.ontrack = e => console.log('remote track', e.track.id);\n\n// Mute / hang up.\nmicTrack.enabled = false;\npeer.close();\nmicTrack.stop();\n```\n\n## API\n\n### `mediaDevices.getUserMedia(constraints)`\n\nAudio only (`{ audio: true }`). Rejects with `NotAllowedError` when the\nmicrophone permission is denied and `NotSupportedError` for video constraints.\n\n### Errors\n\nFailures throw a `SigxError` (from `@sigx/lynx-core`) whose `name` is the\nDOMException name a browser would use — `InvalidStateError`,\n`InvalidAccessError`, `NotAllowedError`, `NotSupportedError`, or\n`OperationError` for a native failure with no W3C equivalent. The same string\nis the machine-readable `code`, so both of these work:\n\n```ts\nimport { isSigxError } from '@sigx/lynx-core';\n\ntry {\n    await peer.setRemoteDescription(answer);\n} catch (e) {\n    if ((e as Error).name === 'InvalidStateError') { /* ported browser code */ }\n    if (isSigxError(e) && e.code === 'InvalidStateError') { /* sigx code */ }\n}\n```\n\nMessages are scoped — `[@sigx/lynx-webrtc] <action> failed: <cause>` — and\ncarry the native cause; branch on `name`/`code`, never on the message. Bad\narguments still throw a plain `TypeError`, as the W3C algorithms specify.\n\n### `RTCPeerConnection`\n\n| Member | Notes |\n|---|---|\n| `new RTCPeerConnection(config?)` | `config.iceServers` supported (`urls` string or array, `username`, `credential`) |\n| `createOffer(options?)` / `createAnswer()` | resolves `{ type, sdp }` |\n| `setLocalDescription(desc?)` / `setRemoteDescription(desc)` | no-arg implicit form supported |\n| `addIceCandidate(candidate?)` | `null`/omitted = end-of-candidates |\n| `addTrack(track, ...streams)` → `RTCRtpSender` | minimal sender (`{ track }`) |\n| `removeTrack(sender)` | |\n| `createDataChannel(label, init?)` | synchronous, like W3C |\n| `close()` | also ends remote tracks and closes child channels; final — no further state events fire, including from calls still in flight |\n| `connectionState` / `iceConnectionState` / `iceGatheringState` / `signalingState` | sync getters |\n| `localDescription` / `remoteDescription` | `localDescription` is refreshed on every `icegatheringstatechange`, so it carries the full SDP once gathering completes |\n| `onconnectionstatechange` / `oniceconnectionstatechange` / `onicegatheringstatechange` / `onsignalingstatechange` | |\n| `onicecandidate` | `ev.candidate` is `null` at end-of-candidates |\n| `ontrack` | `ev.track`, `ev.streams` (no `receiver`/`transceiver`) |\n| `ondatachannel` | `ev.channel` |\n\n### `RTCDataChannel`\n\n`label`, `ordered`, `protocol`, `id` (SCTP stream id, `null` until open),\n`readyState`, `bufferedAmount` (write-through approximation),\n`binaryType: 'arraybuffer'`, `send(string | ArrayBuffer | view)`, `close()`,\n`onopen` / `onmessage` / `onclose` / `onerror`.\n\n### `MediaStreamTrack` / `MediaStream`\n\n`track.enabled = false` mutes without stopping capture; `track.stop()` releases\nthe capturer. `stream.getAudioTracks()` etc. are JS-side bookkeeping; the\nstream's `id` is what appears as the msid in SDP.\n\n### Extras (non-W3C)\n\n```ts\nimport { WebRTC, isWebRTCAvailable } from '@sigx/lynx-webrtc';\nimport type { PermissionResponse } from '@sigx/lynx-webrtc';\n\nawait WebRTC.setAudioOutput('earpiece');     // default route is 'speaker'\nconst { status } = await WebRTC.requestPermission();   // pre-prompt explainer flows\nawait WebRTC.getPermissionStatus();\nisWebRTCAvailable();                          // false on web / when not linked\n```\n\nBoth permission methods resolve the shared\n`PermissionResponse` — `{ status: 'granted' | 'denied' | 'undetermined' |\n'blocked', canAskAgain: boolean }` — re-exported from this barrel so your own\npermission state types without a second import from `@sigx/lynx-core`.\n\n## Deviations from W3C\n\n- Remote audio tracks render automatically through the platform audio device —\n  there is no `<audio>` element. Guard any element-attach code with a platform\n  check when sharing a store with web.\n- `localDescription` / `remoteDescription` return plain frozen `{ type, sdp }`\n  objects with `toJSON()`; there are no `RTCSessionDescription` /\n  `RTCIceCandidate` constructors (init dictionaries are accepted everywhere).\n- `addTrack` returns a minimal sender — no `replaceTrack` / `getParameters`.\n- No `getStats()`, transceivers, video tracks, or renderer views in v1.\n\n## Web\n\n**Not supported on web.** This package ships no `.web.ts` shim and\n`@sigx/lynx-web-host` has no `sigx.webrtc.*` handler, so the `WebRTC` module is\nnot registered in a `sigx run:web` build: `isWebRTCAvailable()` returns `false`,\nand `mediaDevices.getUserMedia()` / `new RTCPeerConnection()` throw the\n\"module not available\" error — gate on `isWebRTCAvailable()` before touching\neither. The browser's own `RTCPeerConnection` is a page API while the app's\nbackground code runs in a Web Worker, so a web implementation would have to\ndrive it over the `@sigx/lynx-web-host` bridge; nothing does that today.\n\n## Platform notes\n\n- **Call audio session:** while at least one peer connection is live, the\n  module holds the platform in call mode (Android `MODE_IN_COMMUNICATION`;\n  iOS `AVAudioSession` `.playAndRecord`/`.voiceChat` via WebRTC's\n  manual-audio mode) and routes output to the loudspeaker by default; the\n  previous state is restored when the last peer closes. On iOS the\n  configuration is re-asserted after route changes and interruptions.\n- **Backgrounding (iOS):** the manifest contributes the `audio`\n  `UIBackgroundModes` entry, so an active call keeps streaming when the app\n  is backgrounded.\n- **Bluetooth headsets:** Android SCO routing is not wired up yet — calls\n  use the built-in speaker or earpiece (iOS allows Bluetooth via\n  `.allowBluetooth`).\n- **`@sigx/lynx-audio` interplay:** avoid starting/stopping its recorders or\n  players during an active call; both packages manage the audio session.\n- **iOS simulator:** WebRTC's audio unit is unreliable there — SDP and data\n  channels work, but verify audio on a physical device.\n- **Android R8:** the manifest contributes `-keep class org.webrtc.** { *; }`\n  via `android.proguardRules`, which `sigx prebuild` writes into\n  `app/proguard-rules-generated.pro`. It is not optional: the\n  `io.github.webrtc-sdk` AAR ships no consumer proguard rules, and libwebrtc\n  resolves its Java surface from native by exact name — including\n  `org.webrtc.WebRtcClassLoader`, which has no Java callers at all. Without the\n  rule R8 deletes it and the library aborts the process inside `JNI_OnLoad` on\n  the first `getUserMedia`. Release builds only; debug is unminified.\n\n## License\n\nMIT\n","readmeFilename":"README.md"}