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Gimpao"},"license":"MIT","homepage":"https://github.com/kitenebie/SoftPhone-FreePBX#readme","keywords":["react","sip","softphone","webrtc","freepbx","asterisk","voip","video-call","audio-call"],"repository":{"type":"git","url":"git+https://github.com/kitenebie/SoftPhone-FreePBX.git"},"description":"A React WebRTC SIP softphone component for FreePBX/Asterisk with audio/video calls, auto-recording, connection monitoring, and API upload support","maintainers":[{"name":"juv","email":"juvybantal@gmail.com"}],"readme":"# juv-ksip-softphone\n\nA professional, draggable, resizable, transparent React WebRTC SIP softphone component for FreePBX / Asterisk. Supports audio and video calls, incoming call handling, codec selection, and a global `ksipcall` API for triggering calls from anywhere in your app.\n\n## WEB DOCUMENTATION\n**Visit Documentation:** [php-ksip-telnet & juv-ksip-softphone Documentation](https://kitenebie.github.io/FreePBX-Console/#sp-intro)\n\n**Author:** Kenneth H. Gimpao\n---\n**Published:**  April 21, 2026\n\n---\n\n## What is this?\n\n`juv-ksip-softphone` is a **React-based WebRTC SIP softphone** designed to be dropped into any existing web application as a transparent floating overlay. It connects to a **FreePBX / Asterisk PBX server** using the SIP protocol over WebSocket (`ws://` or `wss://`) and enables real-time **audio and video calling** directly from the browser — no plugins, no downloads required.\n\n### What is it for?\n\nThis component is built for developers who need to add **VoIP calling capabilities** to their existing React web applications without rebuilding their UI from scratch. Common use cases include:\n\n- **Call center dashboards** — agents can make and receive calls directly from their browser-based CRM or ticketing system\n- **Customer support portals** — embed a softphone into a helpdesk app so support staff can call customers with one click\n- **Dispatch and operations systems** — field coordinators can communicate via audio/video without switching applications\n- **Healthcare platforms** — doctors and staff can conduct audio/video consultations from within a patient management system\n- **Any web app that needs calling** — simply drop `<Softphone />` into your app and it floats transparently on top\n\n### How does it work?\n\n1. **SIP over WebSocket** — The component connects to your FreePBX/Asterisk server via a WebSocket connection (`ws://` or `wss://`). FreePBX listens on port `8088` (ws) or `8089` (wss) for WebSocket SIP traffic.\n\n2. **WebRTC for media** — Once a call is established, the browser uses WebRTC (built into all modern browsers) to handle the actual audio and video streams. This means no additional software is needed — the browser handles microphone, camera, and speaker access natively.\n\n3. **SIP registration** — On connect, the component registers your extension with the PBX just like a physical desk phone would. Once registered, you can make outgoing calls and receive incoming calls.\n\n4. **Transparent overlay** — The softphone renders as a `position: fixed` transparent layer on top of your existing app. Your app's UI is completely unaffected — the softphone simply floats above it.\n\n5. **Global `ksipcall` API** — A global `window.ksipcall` object is exposed so you can trigger calls from anywhere in your codebase — even from plain JavaScript, Vue, Angular, or any other framework running on the same page.\n\n```\nBrowser  ──WebSocket──▶  FreePBX/Asterisk  ──SIP──▶  Other Phone/Extension\n           (SIP/WS)           (PBX)                    (IP Phone, Softphone)\n\nBrowser  ◀──WebRTC──▶  FreePBX/Asterisk  ◀──RTP──▶  Other Phone/Extension\n           (Audio/Video)      (Media)                  (Audio/Video Stream)\n```\n\n---\n\n## Screenshots\n\n<p align=\"center\">\n  <img src=\"https://raw.githubusercontent.com/kitenebie/SoftPhone-FreePBX/main/src/assets/image_1.png\" width=\"48%\" alt=\"Softphone Settings Panel\" />\n  <img src=\"https://raw.githubusercontent.com/kitenebie/SoftPhone-FreePBX/main/src/assets/image_2.png\" width=\"48%\" alt=\"Incoming Call\" />\n</p>\n<p align=\"center\">\n  <img src=\"https://raw.githubusercontent.com/kitenebie/SoftPhone-FreePBX/main/src/assets/image_3.png\" width=\"48%\" alt=\"Video Call\" />\n  <img src=\"https://raw.githubusercontent.com/kitenebie/SoftPhone-FreePBX/main/src/assets/image_4.png\" width=\"48%\" alt=\"Dialer Panel\" />\n</p>\n<p align=\"center\">\n  <img src=\"https://raw.githubusercontent.com/kitenebie/SoftPhone-FreePBX/main/src/assets/image_5.png\" width=\"48%\" alt=\"FullScreen Panel\" />\n</p>\n\n---\n\n## Features\n\n- 🎙 Audio & video calls via WebRTC + SIP (FreePBX / Asterisk)\n- 📞 Incoming call notifications with accept/reject\n- 🪟 Draggable + resizable floating video panel (expandable to fullscreen)\n- 🔢 Draggable dialpad panel\n- 💾 Auto-saves config to `localStorage` and auto-connects on reload\n- 🔄 Auto-reconnects on WebSocket disconnect\n- 🌐 Transparent background — overlays any existing app\n- 🎛 Floating messenger-style FAB nav (settings, dialer, opacity)\n- 📡 Global `ksipcall` API — trigger calls from plain JS or any framework\n- 🔒 ws:// / wss:// protocol selector\n- 🎚 Audio & video codec selection\n- ⚙ Fully configurable via props and in-app settings panel\n- 🔔 Ringtones for incoming and end call\n\n---\n\n## Installation\n\n```bash\nnpm install juv-ksip-softphone\n```\n\n---\n\n## Quick Start\n\n```jsx\nimport { Softphone } from 'juv-ksip-softphone';\nimport 'juv-ksip-softphone/styles';\n\nfunction App() {\n  return (\n    <>\n      <YourExistingApp />\n      <Softphone />\n    </>\n  );\n}\n```\n\nThe softphone renders as a transparent overlay with a floating phone button (top-right). Click it to open the nav menu.\n\n---\n\n## Props\n\n### SIP Connection Props\n\n| Prop | Type | Default | Description |\n|---|---|---|---|\n| `server` | `string` | `\"\"` | FreePBX server IP or hostname |\n| `wsProtocol` | `\"ws\"` \\| `\"wss\"` | `\"ws\"` | WebSocket protocol. Use `wss` for HTTPS pages |\n| `wsPort` | `string` | `\"8088\"` | WebSocket port (`8088` for ws, `8089` for wss) |\n| `extension` | `string` | `\"\"` | SIP extension number |\n| `password` | `string` | `\"\"` | SIP extension password |\n| `displayName` | `string` | `\"\"` | Caller ID display name (optional) |\n\n### Recording Props\n\n| Prop | Type | Default | Description |\n|---|---|---|---|\n| `autoRecord` | `boolean` | `false` | Enable automatic recording of all calls |\n| `recordingDir` | `string` | `\"video/recordings/Ksip\"` | Directory path for saved recordings |\n| `uploadApiUrl` | `string` | `\"\"` | API endpoint URL for uploading recordings (optional) |\n\n### Settings Configuration Props\n\n| Prop | Type | Default | Description |\n|---|---|---|---|\n| `settingConfigToggles` | `object` | all `true` | Controls which toggles are visible in settings UI |\n| `settingConfigTogglesActiveState` | `object` | varies | Sets initial active state for toggles |\n| `settingConfigCodecs` | `object` | all visible | Controls codec visibility and available options |\n\n### UI Props\n\n| Prop | Type | Default | Description |\n|---|---|---|---|\n| `enabledBubble` | `boolean` | `true` | Show or hide the entire softphone bubble |\n| `showDialer` | `boolean` | `true` | Show the dialer button in the FAB nav |\n| `showSetting` | `boolean` | `true` | Show the settings button in the FAB nav |\n| `showOpacity` | `boolean` | `true` | Show the opacity button in the FAB nav |\n| `answerwithVideoCall` | `boolean` | `false` | Auto-answer incoming calls with video. Forces `ShowIncomingCallAudio` to `false` |\n| `ShowIncomingCallVideoBtn` | `boolean` | `true` | Show the video answer button on incoming calls |\n| `ShowIncomingCallAudio` | `boolean` | `true` | Show the audio answer button on incoming calls. Forced `false` when `answerwithVideoCall=true` |\n\n### Priority Order\n\nProps are used as **initial defaults**. If the user has previously saved a config via the settings panel, `localStorage` takes priority. This allows the user to override props from within the app.\n\n---\n\n## Examples\n\n### Basic — manual config via settings panel\n\n```jsx\n<Softphone />\n```\n\n### Pre-configured — auto-connect on load\n\n```jsx\n<Softphone\n  server=\"192.168.1.100\"\n  wsProtocol=\"ws\"\n  wsPort=\"8088\"\n  extension=\"1001\"\n  password=\"mypassword\"\n  displayName=\"John Doe\"\n/>\n```\n\n### Full configuration\n\n```jsx\n<Softphone\n  // SIP Connection\n  server=\"192.168.1.100\"\n  wsProtocol=\"ws\"\n  wsPort=\"8088\"\n  extension=\"1001\"\n  password=\"mypassword\"\n  displayName=\"John Doe\"\n  // UI\n  enabledBubble={true}\n  showDialer={true}\n  showSetting={true}\n  showOpacity={true}\n  answerwithVideoCall={false}\n  ShowIncomingCallVideoBtn={true}\n  ShowIncomingCallAudio={true}\n/>\n```\n\n### Audio-only mode (no video answer button)\n\n```jsx\n<Softphone\n  ShowIncomingCallVideoBtn={false}\n  ShowIncomingCallAudio={true}\n/>\n```\n\n### Auto-answer with video\n\n```jsx\n<Softphone\n  answerwithVideoCall={true}\n  // ShowIncomingCallAudio is automatically false\n/>\n```\n\n### wss:// for HTTPS pages\n\n```jsx\n<Softphone\n  server=\"pbx.yourdomain.com\"\n  wsProtocol=\"wss\"\n  wsPort=\"8089\"\n  extension=\"1001\"\n  password=\"mypassword\"\n/>\n```\n\n### Settings Configuration\n\n```jsx\n<Softphone\n  settingConfigToggles={{\n    bubble: true,\n    dialer: true,\n    settings: false,  // Hide settings toggle\n    opacity: true,\n    autoAnswerVideo: true,\n    answerButtonVideo: false,  // Hide video answer button toggle\n    answerButtonAudio: true,\n    fullscreen: true,\n    autoRecording: true,\n  }}\n  settingConfigTogglesActiveState={{\n    bubble: true,\n    dialer: true,\n    settings: true,\n    opacity: true,\n    autoAnswerVideo: true,  // Start with auto-answer video ON\n    answerButtonVideo: true,\n    answerButtonAudio: false,\n    fullscreen: false,\n    autoRecording: true,  // Start with recording ON\n  }}\n  settingConfigCodecs={{\n    audio: { visible: true, codecs: [\"PCMU\", \"PCMA\"] },  // Only show these 2\n    video: { visible: false, codecs: [\"VP8\", \"H264\"] },  // Hide video codecs section\n  }}\n/>\n```\n\n### Auto Recording with API Upload\n\n```jsx\n<Softphone\n  autoRecord={true}\n  recordingDir=\"video/recordings/Ksip\"\n  uploadApiUrl=\"https://your-domain.com/api/recordings/upload\"\n/>\n```\n\n### Complete Example with All Features\n\n```jsx\n<Softphone\n  // SIP Connection\n  server=\"192.168.1.100\"\n  wsProtocol=\"ws\"\n  wsPort=\"8088\"\n  extension=\"1001\"\n  password=\"mypassword\"\n  displayName=\"John Doe\"\n  \n  // UI Controls\n  enabledBubble={true}\n  showDialer={true}\n  showSetting={true}\n  showOpacity={true}\n  answerwithVideoCall={false}\n  ShowIncomingCallVideoBtn={true}\n  ShowIncomingCallAudio={true}\n  fullscreen={false}\n  \n  // Recording\n  autoRecord={true}\n  recordingDir=\"video/recordings/Ksip\"\n  uploadApiUrl=\"https://your-domain.com/api/recordings/upload\"\n  \n  // Settings Configuration\n  settingConfigToggles={{\n    bubble: true,\n    dialer: true,\n    settings: true,\n    opacity: true,\n    autoAnswerVideo: true,\n    answerButtonVideo: true,\n    answerButtonAudio: true,\n    fullscreen: true,\n    autoRecording: true,\n  }}\n  settingConfigTogglesActiveState={{\n    bubble: true,\n    dialer: true,\n    settings: true,\n    opacity: true,\n    autoAnswerVideo: false,\n    answerButtonVideo: true,\n    answerButtonAudio: true,\n    fullscreen: false,\n    autoRecording: true,\n  }}\n  settingConfigCodecs={{\n    audio: { visible: true, codecs: [\"PCMU\", \"PCMA\", \"opus\"] },\n    video: { visible: true, codecs: [\"VP8\", \"H264\"] },\n  }}\n/>\n```\n\n---\n\n## ksipcall API\n\nThe `ksipcall` global lets you trigger calls from **anywhere** — plain JavaScript, Vue, Angular, or any other framework.\n\n### Audio Call\n\n```js\nksipcall.audio(\"123\");\n```\n\n### Video Call\n\n```js\nksipcall.video(\"123\");\n```\n\n### Via window (plain HTML / vanilla JS)\n\n```html\n<button onclick=\"ksipcall.audio('123')\">Call Support</button>\n<button onclick=\"ksipcall.video('456')\">Video Call</button>\n```\n\n### Import in React / other frameworks\n\n```js\nimport { ksipcall } from 'juv-ksip-softphone';\n\nksipcall.audio(\"123\");\nksipcall.video(\"123\");\n```\n\n---\n\n\n## Settings Panel\n\nThe in-app settings panel (accessible via the ⚙ button) has a **3-column layout**:\n\n### Column 1 — SIP Configuration\n| Field | Description | Example |\n|---|---|---|\n| FreePBX Server IP | IP or hostname | `192.168.1.100` |\n| Extension | SIP extension number | `1001` |\n| Password | Extension SIP password | `mypassword` |\n| Display Name | Caller ID name (optional) | `John Doe` |\n| Protocol | `ws://` or `wss://` | `ws://` |\n| Port | WebSocket port | `8088` |\n\n### Column 2 — Codecs\n\n**Audio Codecs:** `PCMU`, `PCMA`, `G722`, `G729`, `opus`\n\n**Video Codecs:** `VP8`, `VP9`, `H264`, `H265`, `AV1`\n\n**Opacity:** FAB bubble opacity slider (min 30%)\n\n### Column 3 — UI Preferences\n\nAll UI props are configurable as toggles:\n- Show Bubble\n- Show Dialer Button\n- Show Settings Button\n- Show Opacity Button\n- Answer with Video\n- Show Video Answer Button\n- Show Audio Answer Button\n\n---\n\n## CDN Usage (Browser)\n\nYou can use this package directly in the browser via CDN — no build tools required.\n\n### 1. Include scripts\n\n```html\n<!-- Styles -->\n<link rel=\"stylesheet\" href=\"https://cdn.jsdelivr.net/npm/juv-ksip-softphone@1.0.35/dist/juv-ksip-softphone.css\">\n\n<!-- Softphone — fully self-contained, no other scripts needed -->\n<script src=\"https://cdn.jsdelivr.net/npm/juv-ksip-softphone@1.0.35/dist/juv-ksip-softphone.cdn.js\"></script>\n```\n\n### 2. Mount the softphone\n\nThe CDN build exposes `window.JuvKsipSoftphone` with `Softphone`, `createRoot`, and `ksipcall` all included.\n\n```html\n<div id=\"softphone\"></div>\n\n<script>\n  const { Softphone, createRoot, createElement } = window.JuvKsipSoftphone;\n  createRoot(document.getElementById('softphone')).render(\n    createElement(Softphone, {\n      server: \"192.168.1.100\",\n      wsProtocol: \"ws\",\n      wsPort: \"8088\",\n      extension: \"1001\",\n      password: \"mypassword\",\n      displayName: \"John Doe\"\n    })\n  );\n</script>\n```\n\n### Minimal (zero-config — use the settings panel)\n\n```html\n<div id=\"softphone\"></div>\n\n<script>\n  const { Softphone, createRoot, createElement } = window.JuvKsipSoftphone;\n  createRoot(document.getElementById('softphone')).render(\n    createElement(Softphone)\n  );\n</script>\n```\n\n### 3. Trigger calls via `ksipcall` (CDN)\n\nOnce the softphone is mounted, `window.ksipcall` is automatically available:\n\n```html\n<button onclick=\"window.ksipcall.audio('1002')\">Audio Call</button>\n<button onclick=\"window.ksipcall.video('1002')\">Video Call</button>\n```\n\n> **Notes:**\n> - No React or ReactDOM scripts needed — everything is bundled in `juv-ksip-softphone.cdn.js`.\n> - Use `wss://` when your page is served over HTTPS.\n> - All props from the [Props](#props) section work the same way via `React.createElement`.\n\n---\n\n## WebSocket Protocol\n\n| Protocol | Port | Use Case |\n|---|---|---|\n| `ws://` | `8088` | Local network, HTTP pages |\n| `wss://` | `8089` | Production, HTTPS pages (requires SSL cert on FreePBX) |\n\n> **Note:** Browsers block `ws://` (insecure) when the page is served over `https://`. Use `wss://` with a valid SSL certificate for production.\n\n---\n\n## FreePBX / Asterisk Requirements\n\nFor WebRTC to work, each SIP extension must have these settings:\n\n| Setting | Value |\n|---|---|\n| `webrtc` | `yes` |\n| `use_avpf` | `yes` |\n| `media_encryption` | `dtls` |\n| `ice_support` | `yes` |\n| `bundle` | `yes` |\n| `rtcp_mux` | `yes` |\n| `dtls_setup` | `actpass` |\n\n### Enable via FreePBX Admin UI\n\n1. **Applications → Extensions → [your extension]**\n2. Tab: **Advanced**\n3. Set **WebRTC** → `Yes`\n4. **Submit → Apply Config**\n\n### Enable via CLI (pjsip.endpoint_custom.conf)\n\n```ini\n[1001](+)\nwebrtc=yes\n```\n\n```bash\nasterisk -rx \"pjsip reload\"\n```\n\n---\n\n## Floating Nav Controls\n\n| Button | Action |\n|---|---|\n| 📞 Phone (FAB) | Open / close the nav menu |\n| ⊞ Grid | Toggle dialpad panel |\n| ⚙ Settings | Toggle settings & SIP config panel |\n| ≡ Sliders | Quick opacity adjustment (min 30%) |\n\n### Keyboard Shortcuts\n\n| Shortcut | Action |\n|---|---|\n| `Ctrl + Shift + K` | Toggle settings panel |\n\n---\n\n## Status Toast Notification\n\nA floating status indicator appears at the top-center of the screen showing the current connection state:\n\n| Status | Color | Behavior |\n|---|---|---|\n| **Connected** | 🟢 Green | Auto-hides after 5 seconds |\n| **Reconnecting** | 🟡 Yellow | Stays visible until connected |\n| **Not Connected** | 🔴 Red | Stays visible until connected |\n\n- The status toast includes a settings icon button for quick access to the settings panel\n- Automatically appears when the app loads\n- Shows real-time connection status updates\n- Always visible even when bubble is hidden (can be accessed via `Ctrl + Shift + K`)\n\n### Connection Monitoring\n\nThe softphone includes robust connection monitoring to detect server failures:\n\n- **Automatic Re-registration:** SIP registration refreshes every 10 minutes to maintain active connection\n- **Disconnect Detection:** Immediately detects when WebSocket connection is lost\n- **Auto-reconnect:** Attempts to reconnect every 3 seconds when connection is lost\n- **Unexpected Unregistration:** Automatically attempts to re-register if server rejects registration\n- **Long-running Sessions:** Maintains connection health even after hours of being connected without page reload\n\n**Example Scenario:**\n- App connected at 9:00 AM\n- Server goes down at 9:30 AM (30 minutes later)\n- Status toast immediately shows \"Reconnecting...\" (yellow)\n- Auto-retry every 3 seconds\n- When server comes back online → auto-reconnects → \"Connected\" (green)\n\n---\n\n## localStorage\n\nConfig is automatically saved under the key `sip_softphone_config` and restored on next page load, including:\n- SIP credentials (server, extension, password, display name)\n- WebSocket protocol and port\n- Selected audio/video codecs\n- UI preferences (all toggle states including \"Show Bubble\")\n- Recording settings (autoRecord, recordingDir, uploadApiUrl)\n- Fullscreen preference\n- Directory access permission state\n\n> `showSetting` is **never saved** to localStorage — it is always controlled by props.\n\n### Example localStorage data:\n\n```json\n{\n  \"server\": \"192.168.1.100\",\n  \"extension\": \"1001\",\n  \"password\": \"mypassword\",\n  \"displayName\": \"John Doe\",\n  \"wsProtocol\": \"ws\",\n  \"wsPort\": \"8088\",\n  \"audioCodecs\": [\"PCMU\", \"PCMA\"],\n  \"videoCodecs\": [\"VP8\", \"H264\"],\n  \"enabledBubble\": true,\n  \"showDialer\": true,\n  \"showOpacity\": true,\n  \"answerwithVideoCall\": false,\n  \"ShowIncomingCallVideoBtn\": true,\n  \"ShowIncomingCallAudio\": true,\n  \"fullscreen\": false,\n  \"autoRecord\": true,\n  \"recordingDir\": \"video/recordings/Ksip\",\n  \"uploadApiUrl\": \"https://api.example.com/upload\",\n  \"hasDirectoryAccess\": true\n}\n```\n\n\n---\n\n# Auto Recording Feature\n\n## Overview\nThe softphone now supports automatic recording of audio and video calls with configurable save directory.\n\n## Features\n- ✅ Auto-record toggle in settings panel\n- ✅ Configurable recording directory\n- ✅ Saves as WebM format (video/webm or audio/webm)\n- ✅ Automatic download after call ends\n- ✅ Filename format: `{directory}_{timestamp}.webm`\n- ✅ Props support for default configuration\n- ✅ localStorage persistence\n\n## Props\n\n### `autoRecord` (boolean)\n- **Default:** `false`\n- **Description:** Enable automatic recording of all calls\n- **Example:**\n```jsx\n<Softphone autoRecord={true} />\n```\n\n### `recordingDir` (string)\n- **Default:** `\"video/recordings/Ksip\"`\n- **Description:** Directory path for saved recordings (browser will suggest this path in download dialog)\n- **Example:**\n```jsx\n<Softphone \n  autoRecord={true}\n  recordingDir=\"video/recordings/Ksip\"\n/>\n```\n\n## Settings Panel\n\n### Toggle\n- **Location:** Settings → UI Preferences → \"Auto Record Calls\"\n- **Behavior:** Enable/disable automatic recording\n\n### Directory Input\n- **Location:** Appears below toggle when auto-record is enabled\n- **Placeholder:** \"video/recordings/Ksip\"\n- **Default:** \"video/recordings/Ksip\"\n- **Note:** Browser downloads use this as suggested filename path. Actual folder creation depends on browser settings.\n\n## How It Works\n\n1. **Directory Selection:** When auto-record is enabled for the first time, a modal appears asking to select a directory\n2. **Browser Permission:** User selects a folder using the File System Access API (Chrome/Edge)\n3. **Recording Start:** When a call is established (state = \"active\"), recording starts automatically if enabled\n4. **Recording Stop:** When call ends (state = \"idle\"), recording stops and saves directly to the selected directory\n5. **File Format:** WebM container with VP8/Opus codecs (or fallback to audio-only)\n6. **File Naming:** `{ISO-timestamp}.webm`\n   - Example: `2024-04-21T14-30-45-123Z.webm`\n7. **Persistent Access:** Browser remembers the directory permission for future recordings\n\n## Technical Details\n\n### MediaRecorder API\n- Uses browser's native MediaRecorder API\n- Captures both local and remote audio/video tracks\n- Collects chunks every 1 second\n- Creates downloadable blob on stop\n\n### Codec Support\n- **Preferred:** `video/webm;codecs=vp8,opus`\n- **Fallback 1:** `video/webm`\n- **Fallback 2:** `audio/webm`\n\n### Browser Compatibility\n- Chrome/Edge: Full support\n- Firefox: Full support\n- Safari: Limited (may require polyfill)\n\n## Usage Examples\n\n### Basic Usage\n```jsx\n<Softphone autoRecord={true} />\n```\n\n### Custom Directory\n```jsx\n<Softphone \n  autoRecord={true}\n  recordingDir=\"video/recordings/Ksip\"\n/>\n```\n\n### Full Configuration\n```jsx\n<Softphone\n  server=\"192.168.1.100\"\n  extension=\"1001\"\n  password=\"secret\"\n  autoRecord={true}\n  recordingDir=\"video/recordings/Ksip\"\n/>\n```\n\n### Runtime Toggle\nUsers can enable/disable recording via the settings panel without reloading the page.\n\n## Storage\n\nSettings are saved to `localStorage` under key `sip_softphone_config`:\n```json\n{\n  \"autoRecord\": true,\n  \"recordingDir\": \"video/recordings/Ksip\",\n  ...\n}\n```\n\n## Notes\n\n- **File System Access API** (Chrome/Edge 86+, Opera):\n  - Modal prompts user to select directory on first use\n  - Files save directly to selected folder without download prompts\n  - Browser remembers permission (persists across sessions)\n  - User can revoke permission in browser settings\n\n- **Fallback for unsupported browsers** (Firefox, Safari):\n  - Uses traditional download method\n  - Files download to default Downloads folder\n  - No automatic directory creation\n\n- **Security:**\n  - User must explicitly grant directory access\n  - Permission can be revoked anytime\n  - No access to files outside selected directory\n\n- File size depends on call duration and quality\n- WebM format is widely supported and efficient\n- To convert to MP3, use external tools like FFmpeg:\n  ```bash\n  ffmpeg -i recording.webm -vn -ar 44100 -ac 2 -b:a 192k output.mp3\n  ```\n\n---\n\n## Incoming Call User Lookup\n\nThe softphone automatically fetches user information when an incoming call is received. This allows you to display caller details like name, age, address, and mobile number.\n\n### API Endpoint\n\nThe softphone makes a GET request to:\n```\n/user/extension/{extension}\n```\n\nWhere `{extension}` is the caller's extension number extracted from the incoming SIP session.\n\n### Expected Response Format\n\nThe API should return a JSON object with the following structure:\n\n```json\n{\n  \"name\": \"John Doe\",\n  \"age\": 35,\n  \"address\": \"123 Main St, City, State\",\n  \"mobile_number\": \"09171234567\",\n  \"avatar\": \"https://example.com/avatars/johndoe.jpg\"\n}\n```\n\nAll fields are optional. If a field is not provided, it won't be displayed.\n\n### Laravel Controller Example\n\n```php\nuse Illuminate\\Http\\Request;\nuse App\\Models\\User;\n\nclass UserController extends Controller\n{\n    /**\n     * Get user by extension number\n     * \n     * @param string $ext Extension number (matches user's mobile_number)\n     * @return \\Illuminate\\Http\\JsonResponse\n     */\n    public function getUserByExtension($ext)\n    {\n        // Find user where mobile_number matches the extension\n        $user = User::where('mobile_number', $ext)->first();\n        \n        // Return 404 if user not found\n        if (!$user) {\n            return response()->json([\n                'error' => 'User not found'\n            ], 404);\n        }\n        \n        // Return user data\n        return response()->json([\n            'name' => $user->name,\n            'age' => $user->age,\n            'address' => $user->address,\n            'mobile_number' => $user->mobile_number,\n            'avatar' => $user->avatar\n        ]);\n    }\n}\n```\n\n### Route Definition\n\nAdd this route to your `routes/web.php` or `routes/api.php`:\n\n```php\nRoute::get('/user/extension/{ext}', [UserController::class, 'getUserByExtension']);\n```\n\n### Database Schema Example\n\nYour `users` table should have these columns:\n\n```php\nSchema::create('users', function (Blueprint $table) {\n    $table->id();\n    $table->string('name');\n    $table->integer('age')->nullable();\n    $table->text('address')->nullable();\n    $table->string('mobile_number')->unique();\n    $table->string('avatar')->nullable();\n    $table->timestamps();\n});\n```\n\n### How It Works\n\n1. When an incoming call is received, the softphone extracts the caller's extension number from the SIP session\n2. A GET request is made to `/user/extension/{extension}`\n3. Your Laravel controller queries the database using `User::where('mobile_number', $ext)`\n4. The user data is returned as JSON\n5. The softphone displays the caller's avatar (if provided), name, age, address, and mobile number in the incoming call UI\n6. If no avatar is provided, a default phone icon is shown\n\n### Error Handling\n\nIf the API request fails or returns an error, the softphone will:\n- Log the error to the browser console\n- Display the default caller information (extension number or display name from SIP)\n- Continue to function normally\n\n---\n\n## Author\n\n**Kenneth H. Gimpao**\nPublished: 2026\n\n**GitHub Repository:** [https://github.com/kitenebie/SoftPhone-FreePBX](https://github.com/kitenebie/SoftPhone-FreePBX)\n\n**php-ksip-telnet (laravel):** [https://github.com/kitenebie/FreePBX-Console.git](https://github.com/kitenebie/FreePBX-Console.git)\n---\n\n## License\n\nMIT\n","readmeFilename":"README.md"}