84 lines
3.6 KiB
PHP
84 lines
3.6 KiB
PHP
<?php
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use Blax\WebRtc\Authorization\DefaultRoomAuthorizer;
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use Blax\WebRtc\Events\NullCallEventListener;
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use Blax\WebRtc\Media\NullMediaEngine;
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return [
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/*
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|--------------------------------------------------------------------------
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| Signaling bind
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|--------------------------------------------------------------------------
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| Where the signaling server (on the shared ReactPHP kernel) listens. For
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| browser clients you will typically front this with the laravel-websockets
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| WS transport instead; this raw endpoint drives + tests the media engine.
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*/
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'host' => env('WEBRTC_HOST', '127.0.0.1'),
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'port' => (int) env('WEBRTC_PORT', 8090),
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// ReactPHP TLS context; leave both unset for plain TCP.
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'tls' => array_filter([
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'local_cert' => env('WEBRTC_TLS_CERT'),
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'local_pk' => env('WEBRTC_TLS_KEY'),
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]),
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/*
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|--------------------------------------------------------------------------
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| Media engine
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|--------------------------------------------------------------------------
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| The backend that terminates media (ICE/DTLS/SRTP/RTP/Opus), records, and
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| bridges to external realtime providers. NullMediaEngine = signaling only;
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| Str0mMediaEngine requires the blax_webrtc Rust extension (see rust/).
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*/
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'media_engine' => env('WEBRTC_MEDIA_ENGINE', NullMediaEngine::class),
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/*
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|--------------------------------------------------------------------------
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| Recording (store the full call audio)
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|--------------------------------------------------------------------------
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| When enabled, the browser streams its mic to the server as binary WS frames
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| and RelaySignalingHandler appends them per participant via a RecordingStore
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| (FileRecordingStore by default: <path>/<room>/<peer>.<format>). See README.
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*/
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'recording' => [
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'enabled' => (bool) env('WEBRTC_RECORDING', false),
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'path' => env('WEBRTC_RECORDING_PATH', storage_path('app/webrtc')),
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'format' => env('WEBRTC_RECORDING_FORMAT', 'webm'),
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],
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/*
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|--------------------------------------------------------------------------
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| Room authorization + call-event log
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|--------------------------------------------------------------------------
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| `authorizer` gates private-* / presence-* joins (bind your own that checks a
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| token / the authenticated user; the default DENIES them). `events` receives
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| call-lifecycle events so you can log the call (bind your own to persist).
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*/
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'authorizer' => env('WEBRTC_AUTHORIZER', DefaultRoomAuthorizer::class),
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'events' => env('WEBRTC_EVENTS', NullCallEventListener::class),
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/*
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|--------------------------------------------------------------------------
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| ICE servers (STUN/TURN) advertised to browsers
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|--------------------------------------------------------------------------
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*/
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'ice_servers' => [
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// ['urls' => 'stun:stun.l.google.com:19302'],
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// ['urls' => 'turn:turn.example.com:3478', 'username' => '...', 'credential' => '...'],
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],
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/*
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|--------------------------------------------------------------------------
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| External realtime bridge (provider hidden from the browser)
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|--------------------------------------------------------------------------
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| The media engine dials these; the browser only ever talks to us, so the AI
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| provider is invisible and the model is swappable server-side.
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*/
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'bridge' => [
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'openai' => [
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'model' => env('WEBRTC_OPENAI_MODEL', 'gpt-realtime'),
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'api_key' => env('OPENAI_API_KEY'),
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],
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],
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];
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