laravel-webrtc/config/webrtc.php

<?php

use Blax\WebRtc\Authorization\DefaultRoomAuthorizer;
use Blax\WebRtc\Events\NullCallEventListener;
use Blax\WebRtc\Media\NullMediaEngine;

return [
    /*
    |--------------------------------------------------------------------------
    | Signaling bind
    |--------------------------------------------------------------------------
    | Where the signaling server (on the shared ReactPHP kernel) listens. For
    | browser clients you will typically front this with the laravel-websockets
    | WS transport instead; this raw endpoint drives + tests the media engine.
    */
    'host' => 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: <path>/<room>/<peer>.<format>). 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),

    /*
    |--------------------------------------------------------------------------
    | 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)
    |--------------------------------------------------------------------------
    | 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.
    */
    'bridge' => [
        'openai' => [
            'model' => env('WEBRTC_OPENAI_MODEL', 'gpt-realtime'),
            'api_key' => env('OPENAI_API_KEY'),
        ],
    ],
];