laravel-webrtc/config/webrtc.php

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<?php
feat: full package — typed rooms (public/private/presence/open-presence) + call recording Rooms - Rooms\RoomType derived from the name prefix (like laravel-websockets channels): public, private-* (authorized), presence-* (authorized + member list), open-presence-* (public + member list). Room/RoomManager carry per-member info. - Contracts\RoomAuthorizer + secure DefaultRoomAuthorizer (denies private/presence until the host binds one). Presence rooms broadcast the member list + info. Recording (store the FULL call audio, in PHP, today) - Pure-P2P audio is DTLS-SRTP end-to-end, so the server can't capture it. Instead the browser MediaRecords its mic and streams chunks over the WS as BINARY frames (needs the new laravel-ws onBinary); RelaySignalingHandler appends them per participant via a RecordingStore. Recording\FileRecordingStore -> <room>/<peer>.webm. welcome/joined carry a record flag telling the client to stream. - Contracts\CallEventListener call-log seam (peer/room/recording lifecycle) so calls are auditable even when unrecorded. Wiring - RelaySignalingHandler(authorizer, recorder, events); webrtc:serve + provider + config/webrtc.php (recording enabled/path/format, authorizer, events). Tests 35/99 incl on-disk FileRecordingStore, presence member lists, auth gating, binary recording+finalize+events, RoomType, and the real WS round-trip. README documents room types + the browser MediaRecorder->WS recording flow. rel learn-atc #1060 #1055 #1057 #1051 Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-07-08 07:01:12 +00:00
use Blax\WebRtc\Authorization\DefaultRoomAuthorizer;
use Blax\WebRtc\Events\NullCallEventListener;
use Blax\WebRtc\Media\NullMediaEngine;
use Blax\WebRtc\Rooms\Stores\ArrayRoomStore;
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
| (peer-to-peer mesh); RustMediaEngine = BACKEND-HOSTED rooms on the Rust
| SFU sidecar (plug-and-play see the `sfu` block below).
*/
'media_engine' => env('WEBRTC_MEDIA_ENGINE', NullMediaEngine::class),
/*
|--------------------------------------------------------------------------
| Rust SFU sidecar (RustMediaEngine)
|--------------------------------------------------------------------------
| The `blax-webrtc-sfu` binary from this package's rust/ directory: one
| process per host that terminates ICE/DTLS/SRTP and forwards audio between
| room peers (no decoding), records Ogg/Opus per peer, and reports usage.
|
| Plug-and-play defaults: the first RustMediaEngine call DOWNLOADS the
| checksummed prebuilt binary for this platform (auto_install) and SPAWNS
| it detached, flock-guarded (auto_spawn). For supervised deployments run
| `php artisan webrtc:sidecar` as a service and disable auto_spawn.
|
| udp_port 0 = ephemeral (fine same-host); pin it (and publish it on the
| container) for production, and set public_ip to the address browsers
| can reach it is advertised in the ICE host candidate.
*/
'sfu' => [
'socket' => env('WEBRTC_SFU_SOCKET', storage_path('app/webrtc/sfu.sock')),
'udp_ip' => env('WEBRTC_SFU_UDP_IP', '0.0.0.0'),
'udp_port' => (int) env('WEBRTC_SFU_UDP_PORT', 0),
'public_ip' => env('WEBRTC_SFU_PUBLIC_IP'),
'auto_spawn' => (bool) env('WEBRTC_SFU_AUTO_SPAWN', true),
'auto_install' => (bool) env('WEBRTC_SFU_AUTO_INSTALL', true),
'spawn_timeout' => (float) env('WEBRTC_SFU_SPAWN_TIMEOUT', 10),
'log' => env('WEBRTC_SFU_LOG', storage_path('logs/webrtc-sfu.log')),
// Binary resolution: explicit path > local rust/target build > install.
'binary' => env('WEBRTC_SFU_BINARY'),
'install_path' => env('WEBRTC_SFU_INSTALL_PATH', storage_path('app/webrtc/bin')),
'version' => env('WEBRTC_SFU_VERSION'), // default: the version pinned by this package
'download_url' => env(
'WEBRTC_SFU_DOWNLOAD_URL',
'https://git.blax.at/blax-software/laravel-webrtc/releases/download/{version}/blax-webrtc-sfu-{platform}',
),
// sha256 verification is REQUIRED when set; null opts out explicitly.
'checksum_url' => env(
'WEBRTC_SFU_CHECKSUM_URL',
'https://git.blax.at/blax-software/laravel-webrtc/releases/download/{version}/blax-webrtc-sfu-{platform}.sha256',
),
],
/*
|--------------------------------------------------------------------------
feat: full package — typed rooms (public/private/presence/open-presence) + call recording Rooms - Rooms\RoomType derived from the name prefix (like laravel-websockets channels): public, private-* (authorized), presence-* (authorized + member list), open-presence-* (public + member list). Room/RoomManager carry per-member info. - Contracts\RoomAuthorizer + secure DefaultRoomAuthorizer (denies private/presence until the host binds one). Presence rooms broadcast the member list + info. Recording (store the FULL call audio, in PHP, today) - Pure-P2P audio is DTLS-SRTP end-to-end, so the server can't capture it. Instead the browser MediaRecords its mic and streams chunks over the WS as BINARY frames (needs the new laravel-ws onBinary); RelaySignalingHandler appends them per participant via a RecordingStore. Recording\FileRecordingStore -> <room>/<peer>.webm. welcome/joined carry a record flag telling the client to stream. - Contracts\CallEventListener call-log seam (peer/room/recording lifecycle) so calls are auditable even when unrecorded. Wiring - RelaySignalingHandler(authorizer, recorder, events); webrtc:serve + provider + config/webrtc.php (recording enabled/path/format, authorizer, events). Tests 35/99 incl on-disk FileRecordingStore, presence member lists, auth gating, binary recording+finalize+events, RoomType, and the real WS round-trip. README documents room types + the browser MediaRecorder->WS recording flow. rel learn-atc #1060 #1055 #1057 #1051 Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-07-08 07:01:12 +00:00
| Recording (store the full call audio)
|--------------------------------------------------------------------------
feat: full package — typed rooms (public/private/presence/open-presence) + call recording Rooms - Rooms\RoomType derived from the name prefix (like laravel-websockets channels): public, private-* (authorized), presence-* (authorized + member list), open-presence-* (public + member list). Room/RoomManager carry per-member info. - Contracts\RoomAuthorizer + secure DefaultRoomAuthorizer (denies private/presence until the host binds one). Presence rooms broadcast the member list + info. Recording (store the FULL call audio, in PHP, today) - Pure-P2P audio is DTLS-SRTP end-to-end, so the server can't capture it. Instead the browser MediaRecords its mic and streams chunks over the WS as BINARY frames (needs the new laravel-ws onBinary); RelaySignalingHandler appends them per participant via a RecordingStore. Recording\FileRecordingStore -> <room>/<peer>.webm. welcome/joined carry a record flag telling the client to stream. - Contracts\CallEventListener call-log seam (peer/room/recording lifecycle) so calls are auditable even when unrecorded. Wiring - RelaySignalingHandler(authorizer, recorder, events); webrtc:serve + provider + config/webrtc.php (recording enabled/path/format, authorizer, events). Tests 35/99 incl on-disk FileRecordingStore, presence member lists, auth gating, binary recording+finalize+events, RoomType, and the real WS round-trip. README documents room types + the browser MediaRecorder->WS recording flow. rel learn-atc #1060 #1055 #1057 #1051 Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-07-08 07:01:12 +00:00
| 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' => [
feat: full package — typed rooms (public/private/presence/open-presence) + call recording Rooms - Rooms\RoomType derived from the name prefix (like laravel-websockets channels): public, private-* (authorized), presence-* (authorized + member list), open-presence-* (public + member list). Room/RoomManager carry per-member info. - Contracts\RoomAuthorizer + secure DefaultRoomAuthorizer (denies private/presence until the host binds one). Presence rooms broadcast the member list + info. Recording (store the FULL call audio, in PHP, today) - Pure-P2P audio is DTLS-SRTP end-to-end, so the server can't capture it. Instead the browser MediaRecords its mic and streams chunks over the WS as BINARY frames (needs the new laravel-ws onBinary); RelaySignalingHandler appends them per participant via a RecordingStore. Recording\FileRecordingStore -> <room>/<peer>.webm. welcome/joined carry a record flag telling the client to stream. - Contracts\CallEventListener call-log seam (peer/room/recording lifecycle) so calls are auditable even when unrecorded. Wiring - RelaySignalingHandler(authorizer, recorder, events); webrtc:serve + provider + config/webrtc.php (recording enabled/path/format, authorizer, events). Tests 35/99 incl on-disk FileRecordingStore, presence member lists, auth gating, binary recording+finalize+events, RoomType, and the real WS round-trip. README documents room types + the browser MediaRecorder->WS recording flow. rel learn-atc #1060 #1055 #1057 #1051 Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-07-08 07:01:12 +00:00
'enabled' => (bool) env('WEBRTC_RECORDING', false),
'path' => env('WEBRTC_RECORDING_PATH', storage_path('app/webrtc')),
'format' => env('WEBRTC_RECORDING_FORMAT', 'webm'),
],
/*
|--------------------------------------------------------------------------
| Backend-defined rooms (the room registry)
|--------------------------------------------------------------------------
| The rooms clients are allowed to join. A join to anything NOT listed here is
| rejected clients can't invent rooms. Each room configures its transport:
| 'p2p' browsers mesh, the server only relays signaling (no server media)
| 'sfu' the server terminates media (the Rust SFU; enables recording/scale)
|
| Shapes (all equivalent granularity):
| 'just-talk' => 'p2p',
| 'lobby' => ['mode' => 'sfu', 'limit' => 50, 'type' => 'presence'],
| 'private-desk' => ['mode' => 'p2p'], // membership type inferred from prefix
|
| For DYNAMIC rooms (created/edited at runtime), point `room_registry` at your
| own Contracts\RoomRegistry (e.g. an Eloquent-backed one); the static `rooms`
| map below is the fallback used when no registry class is bound.
*/
'room_registry' => env('WEBRTC_ROOM_REGISTRY'),
'rooms' => [],
feat: full package — typed rooms (public/private/presence/open-presence) + call recording Rooms - Rooms\RoomType derived from the name prefix (like laravel-websockets channels): public, private-* (authorized), presence-* (authorized + member list), open-presence-* (public + member list). Room/RoomManager carry per-member info. - Contracts\RoomAuthorizer + secure DefaultRoomAuthorizer (denies private/presence until the host binds one). Presence rooms broadcast the member list + info. Recording (store the FULL call audio, in PHP, today) - Pure-P2P audio is DTLS-SRTP end-to-end, so the server can't capture it. Instead the browser MediaRecords its mic and streams chunks over the WS as BINARY frames (needs the new laravel-ws onBinary); RelaySignalingHandler appends them per participant via a RecordingStore. Recording\FileRecordingStore -> <room>/<peer>.webm. welcome/joined carry a record flag telling the client to stream. - Contracts\CallEventListener call-log seam (peer/room/recording lifecycle) so calls are auditable even when unrecorded. Wiring - RelaySignalingHandler(authorizer, recorder, events); webrtc:serve + provider + config/webrtc.php (recording enabled/path/format, authorizer, events). Tests 35/99 incl on-disk FileRecordingStore, presence member lists, auth gating, binary recording+finalize+events, RoomType, and the real WS round-trip. README documents room types + the browser MediaRecorder->WS recording flow. rel learn-atc #1060 #1055 #1057 #1051 Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-07-08 07:01:12 +00:00
/*
|--------------------------------------------------------------------------
| 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', 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,
],
],
];