Rooms / recording / realtime bridge no longer require the bundled ReactPHP
server, so a host (e.g. a fork-per-message Laravel WS) can consume the package
on its own transport:
- RoomStore contract + ArrayRoomStore; RoomManager takes a store so membership
can be Cache/Redis-backed and survive across workers
- Realtime\OpenAiRealtimeBridge (+ RealtimeTransport / TextalkRealtimeTransport /
RealtimeTurn): the OpenAI GA realtime WS mechanism, turn-based, unit-tested via
a scripted transport
- Support\Audio (PCM16 -> WAV); RecordingStore::discard()
- provider binds RoomStore/RoomManager/RecordingStore/OpenAiRealtimeBridge and
guards webrtc:serve behind class_exists(laravel-ws + reactphp-kernel)
- laravel-ws + reactphp-kernel + react/* -> require-dev; runtime needs only
illuminate + textalk/websocket
- 40 tests green
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
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>
The key insight: a 2-party / small-mesh call needs NO server-side media — the
server only relays SDP/ICE and the browsers connect peer-to-peer. So the webrtc
package now has a genuinely working call MVP with no Rust/media engine.
- Signaling\RelaySignalingHandler: a blax-software/laravel-ws MessageHandler that
relays between browsers — welcome/join/joined/peer-joined/relay/broadcast/
peer-left. Peers in the kernel ConnectionRegistry, rooms in RoomManager.
- webrtc:serve now runs the relay over a laravel-ws WebSocketServer (browser-usable
ws://). require blax-software/laravel-ws.
- The engine-backed path (SignalingHandler/SignalingServer + MediaEngine) stays
for server-terminated media (recording/AI-bridge/SFU, Str0m = #1051).
- Tests 26 / 60 incl a REAL WebSocket welcome round-trip (browser->laravel-ws->relay);
verified with laravel-ws + reactphp-kernel resolved from Forgejo. README leads
with the browser P2P quickstart + JS example.
rel learn-atc #1055#1051#1019
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- Rooms\Room + Rooms\RoomManager: track peers per room, auto-mesh a new joiner
with everyone already there, drop empty rooms
- SignalingHandler: join/leave message types (auto-connectPeers on join), and
close now also leaves the room; expose rooms()
- SignalingServer::address() for the bound listen address
- Tests: real offer->answer round-trip over a live TCP socket on the kernel loop,
RoomManager + join/leave/close coverage. Suite: 17 tests / 42 assertions
- Verified on the updated reactphp-kernel (logging + registry)
rel learn-atc #1055#1051#1019
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
WebRTC for Laravel on blax-software/reactphp-kernel. PHP owns signaling +
orchestration; a pluggable MediaEngine owns real-time media, with the production
engine backed by a Rust core (str0m via ext-php-rs) so PHP never runs per-20ms DSP.
- WebRtcServiceProvider + config/webrtc.php + webrtc:serve command
- Contracts\MediaEngine (offer/ice/record/connect/bridge/close)
- Media\NullMediaEngine (signaling-only) + Media\Str0mMediaEngine (Rust seam)
- Signaling\SignalingHandler (transport-agnostic) + Server\SignalingServer (on the kernel)
- rust/ crate stub (blax_webrtc: str0m + ext-php-rs) documenting the media plan
Enables: server-side record/intercept, provider-hidden AI-realtime bridging,
and party-to-party/group calls. rel learn-atc #1055#1051#1019
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Blax Software