env('WEBRTC_HOST', '127.0.0.1'), 'port' => (int) env('WEBRTC_PORT', 8090), // ReactPHP TLS context; leave both unset for plain TCP. 'tls' => array_filter([ 'local_cert' => env('WEBRTC_TLS_CERT'), 'local_pk' => env('WEBRTC_TLS_KEY'), ]), /* |-------------------------------------------------------------------------- | Media engine |-------------------------------------------------------------------------- | The backend that terminates media (ICE/DTLS/SRTP/RTP/Opus), records, and | bridges to external realtime providers. NullMediaEngine = signaling only; | Str0mMediaEngine requires the blax_webrtc Rust extension (see rust/). */ 'media_engine' => env('WEBRTC_MEDIA_ENGINE', NullMediaEngine::class), /* |-------------------------------------------------------------------------- | Recording (store the full call audio) |-------------------------------------------------------------------------- | When enabled, the browser streams its mic to the server as binary WS frames | and RelaySignalingHandler appends them per participant via a RecordingStore | (FileRecordingStore by default: //.). See README. */ 'recording' => [ 'enabled' => (bool) env('WEBRTC_RECORDING', false), 'path' => env('WEBRTC_RECORDING_PATH', storage_path('app/webrtc')), 'format' => env('WEBRTC_RECORDING_FORMAT', 'webm'), ], /* |-------------------------------------------------------------------------- | Room authorization + call-event log |-------------------------------------------------------------------------- | `authorizer` gates private-* / presence-* joins (bind your own that checks a | token / the authenticated user; the default DENIES them). `events` receives | call-lifecycle events so you can log the call (bind your own to persist). */ 'authorizer' => env('WEBRTC_AUTHORIZER', DefaultRoomAuthorizer::class), 'events' => env('WEBRTC_EVENTS', NullCallEventListener::class), /* |-------------------------------------------------------------------------- | Room membership store |-------------------------------------------------------------------------- | Where room membership + presence lives. The default ArrayRoomStore keeps it | in process memory (right for the long-lived bundled server). A fork-per- | message host (a classic Laravel WebSocket handling each frame in a fresh | worker) binds a Cache/Redis-backed RoomStore so state survives across | workers — that is how a host runs this relay on its existing WS. */ 'room_store' => env('WEBRTC_ROOM_STORE', \Blax\WebRtc\Rooms\Stores\ArrayRoomStore::class), /* |-------------------------------------------------------------------------- | ICE servers (STUN/TURN) advertised to browsers |-------------------------------------------------------------------------- */ 'ice_servers' => [ // ['urls' => 'stun:stun.l.google.com:19302'], // ['urls' => 'turn:turn.example.com:3478', 'username' => '...', 'credential' => '...'], ], /* |-------------------------------------------------------------------------- | External realtime bridge (provider hidden from the browser) |-------------------------------------------------------------------------- | OpenAiRealtimeBridge (or the media engine) dials these; the browser only ever | talks to us, so the AI provider is invisible and the model is swappable | server-side. `instructions` is the persona (the host injects its own — e.g. | an ATC prompt) and `voice` the spoken voice. */ 'bridge' => [ 'openai' => [ 'endpoint' => env('WEBRTC_OPENAI_ENDPOINT', 'wss://api.openai.com/v1/realtime'), 'model' => env('WEBRTC_OPENAI_MODEL', 'gpt-realtime'), 'api_key' => env('OPENAI_API_KEY'), 'voice' => env('WEBRTC_OPENAI_VOICE', 'alloy'), 'instructions' => null, ], ], ];