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# Laravel WebRTC

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WebRTC for Laravel on the shared [`reactphp-kernel`](https://github.com/blax-software/reactphp-kernel) backbone. A **WebSocket signaling relay** connects browsers for **peer-to-peer calls today** (no server-side media needed); when you need the server *in* the media path — recording, AI bridging, an SFU — a pluggable **media engine** (a Rust `str0m` core via `ext-php-rs`) takes over.

## Features

- 📞 **Browser-to-browser calls, working today** — two browsers `join` a room and the server relays their SDP/ICE so they connect **peer-to-peer**; the audio never touches the server, so no media engine is required
- 👥 **Rooms & mesh signaling** — peer discovery on join, `peer-joined` / `peer-left` notifications, targeted `relay`, and room `broadcast`
- 🧩 **On the shared kernel** — runs on `reactphp-kernel` over the [`laravel-ws`](https://github.com/blax-software/laravel-ws) WebSocket transport; one process, one loop, alongside your other realtime servers
- 🎙️ **Server-side media, when you need it** — a pluggable `MediaEngine` terminates media for **recording / intercept**, **AI-realtime bridging**, and **SFU** group calls
- 🤖 **AI realtime, provider hidden** — bridge a caller to an external model (e.g. OpenAI Realtime) server-side; the browser only ever talks to *you*, and the model is a config swap
- 🔌 **Pluggable engine** — `NullMediaEngine` (signaling-only) now; `Str0mMediaEngine` (Rust `str0m` via `ext-php-rs`, shipped in `rust/`) for real server-terminated media

## Installation

```bash
composer require blax-software/laravel-webrtc
php artisan vendor:publish --tag="webrtc-config"
```

It depends on [`blax-software/reactphp-kernel`](https://github.com/blax-software/reactphp-kernel) + [`blax-software/laravel-ws`](https://github.com/blax-software/laravel-ws), resolved from git.blax.at via the `repositories` entry in `composer.json`.

## Quick Start

Run the signaling relay (browsers connect here):

```bash
php artisan webrtc:serve   # ws://127.0.0.1:8090 by default
```

That's all a **peer-to-peer call** needs — the server only relays signaling; the browsers exchange audio directly:

```js
const ws = new WebSocket('ws://127.0.0.1:8090')
const pc = new RTCPeerConnection()
let peer

ws.onmessage = async ({ data }) => {
  const m = JSON.parse(data)
  if (m.type === 'welcome')     ws.send(JSON.stringify({ type: 'join', room: 'lobby' }))
  if (m.type === 'joined')      m.peers.forEach(p => (peer = p, call(p)))   // someone's already here → call them
  if (m.type === 'peer-joined') peer = m.peer                               // they'll send us an offer
  if (m.type === 'relay') {                                                 // SDP / ICE from the other browser
    peer = m.from
    if (m.data.sdp)       { await pc.setRemoteDescription(m.data); if (m.data.type === 'offer') answer() }
    if (m.data.candidate) await pc.addIceCandidate(m.data)
  }
}
pc.onicecandidate = e => e.candidate && ws.send(JSON.stringify({ type: 'relay', to: peer, data: e.candidate }))
pc.ontrack = e => audioEl.srcObject = e.streams[0]

const send = (data) => ws.send(JSON.stringify({ type: 'relay', to: peer, data }))
async function call()   { const o = await pc.createOffer();  await pc.setLocalDescription(o); send(o) }
async function answer() { const a = await pc.createAnswer(); await pc.setLocalDescription(a); send(a) }
```

### Relay protocol (JSON over WebSocket)

```jsonc
←  { "type": "welcome", "peer": "<yourId>" }                     // on connect
→  { "type": "join", "room": "lobby" }
←  { "type": "joined", "room": "lobby", "peers": ["<id>", ...] } // who's already here
←  { "type": "peer-joined", "room": "lobby", "peer": "<id>" }    // sent to the others
→  { "type": "relay", "to": "<peerId>", "data": { ...sdp | candidate... } }
←  { "type": "relay", "from": "<peerId>", "data": { ... } }      // delivered to the target
→  { "type": "broadcast", "data": { ... } }                      // to the rest of your room
→  { "type": "leave" }        ← { "type": "left" } / others get { "type": "peer-left", "peer" }
```

### Server-terminated media (recording / AI bridge / SFU)

When the server must be *in* the media path, configure a `MediaEngine`. `NullMediaEngine` (default) is signaling-only; `Str0mMediaEngine` is backed by a Rust `str0m` core via `ext-php-rs` (see [`rust/README.md`](rust/README.md)):

```dotenv
WEBRTC_MEDIA_ENGINE="Blax\WebRtc\Media\Str0mMediaEngine"
```

The engine-backed path uses `Signaling\SignalingHandler` (offer/ice/record/connect/bridge/close) + `Server\SignalingServer`; it's independent of the browser relay above and shares the same `RoomManager`.

## Configuration

`config/webrtc.php` covers the signaling bind (`host`/`port`/`tls`), the `media_engine` binding, `recording` disk/path, advertised `ice_servers`, and the external `bridge` (e.g. OpenAI model + key). Swapping the realtime model — or the whole provider — is a server-side config change the browser never sees.

## Status

**Working MVP.** Browser-to-browser (mesh) calls work today via the WebSocket signaling relay — no server-side media required. Rooms, peer discovery, targeted relay and broadcast are tested, including a real WebSocket round-trip. Server-terminated media (recording, AI bridge, SFU) is the pluggable `MediaEngine`; the real one (`Str0mMediaEngine` + `rust/`) is the next build.

## Architecture

```
reactphp-kernel                        shared backbone (loop, IPC, signals)
   ├─ laravel-ws                       WebSocket transport (RFC6455)
   └─ laravel-webrtc  (this)
         ├─ RelaySignalingHandler      browser P2P calls  (no media engine needed)
         └─ MediaEngine (optional)     server-terminated media
               └─ Str0mMediaEngine     Rust/str0m via ext-php-rs  (rust/)
```

## Testing

```bash
composer install
composer test
```

The relay + signaling routers and engines are unit-tested with fakes; a real WebSocket round-trip proves the relay over `laravel-ws`, and a raw-socket test drives the engine signaling server — no browser or external services required.

## License

MIT. See [LICENSE](LICENSE).

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