feat: implement frontend for calendar integration (Phase 3 & 4)

## Frontend Implementation ### Meeting Selection & Management - Created MeetingSelection component for choosing between multiple active meetings - Shows both active meetings and upcoming calendar events (30 min ahead) - Displays meeting metadata with privacy controls (owner-only details) - Supports creation of unscheduled meetings alongside calendar meetings ### Waiting Room - Added waiting page for users joining before scheduled start time - Shows countdown timer until meeting begins - Auto-transitions to meeting when calendar event becomes active - Handles early joining with proper routing ### Meeting Info Panel - Created collapsible info panel showing meeting details - Displays calendar metadata (title, description, attendees) - Shows participant count and duration - Privacy-aware: sensitive info only visible to room owners ### ICS Configuration UI - Integrated ICS settings into room configuration dialog - Test connection functionality with immediate feedback - Manual sync trigger with detailed results - Shows last sync time and ETag for monitoring - Configurable sync intervals (1 min to 1 hour) ### Routing & Navigation - New /room/{roomName} route for meeting selection - Waiting room at /room/{roomName}/wait?eventId={id} - Classic room page at /{roomName} with meeting info - Uses sessionStorage to pass selected meeting between pages ### API Integration - Added new endpoints for active/upcoming meetings - Regenerated TypeScript client with latest OpenAPI spec - Proper error handling and loading states - Auto-refresh every 30 seconds for live updates ### UI/UX Improvements - Color-coded badges for meeting status - Attendee status indicators (accepted/declined/tentative) - Responsive design with Chakra UI components - Clear visual hierarchy between active and upcoming meetings - Smart truncation for long attendee lists This completes the frontend implementation for calendar integration, enabling users to seamlessly join scheduled meetings from their calendar applications.
feat: implement Phase 2 - Multiple active meetings per room with grace period
2025-12-20 20:29:06 +00:00 · 2025-08-18 19:29:56 -06:00 · 2025-08-18 19:03:41 -06:00 · 2025-08-18 17:22:41 -06:00 · 2025-08-18 17:03:23 -06:00 · 2025-08-18 16:51:30 -06:00
260 changed files with 15508 additions and 26452 deletions
--- a/.github/workflows/db_migrations.yml
+++ b/.github/workflows/db_migrations.yml
@@ -2,8 +2,6 @@ name: Test Database Migrations

 on:
  push:
-    branches:
-      - main
    paths:
      - "server/migrations/**"
      - "server/reflector/db/**"
@@ -19,9 +17,6 @@ on:
 jobs:
  test-migrations:
    runs-on: ubuntu-latest
-    concurrency:
-      group: db-ubuntu-latest-${{ github.ref }}
-      cancel-in-progress: true
    services:
      postgres:
        image: postgres:17
--- a/.github/workflows/deploy.yml
+++ b/.github/workflows/deploy.yml
@@ -8,30 +8,18 @@ env:
  ECR_REPOSITORY: reflector

 jobs:
-  build:
-    strategy:
-      matrix:
-        include:
-          - platform: linux/amd64
-            runner: linux-amd64
-            arch: amd64
-          - platform: linux/arm64
-            runner: linux-arm64
-            arch: arm64
-
-    runs-on: ${{ matrix.runner }}
+  deploy:
+    runs-on: ubuntu-latest

    permissions:
+      deployments: write
      contents: read

-    outputs:
-      registry: ${{ steps.login-ecr.outputs.registry }}
-
    steps:
-      - uses: actions/checkout@v4
+      - uses: actions/checkout@v3

      - name: Configure AWS credentials
-        uses: aws-actions/configure-aws-credentials@v4
+        uses: aws-actions/configure-aws-credentials@0e613a0980cbf65ed5b322eb7a1e075d28913a83
        with:
          aws-access-key-id: ${{ secrets.AWS_ACCESS_KEY_ID }}
          aws-secret-access-key: ${{ secrets.AWS_SECRET_ACCESS_KEY }}
@@ -39,52 +27,21 @@ jobs:

      - name: Login to Amazon ECR
        id: login-ecr
-        uses: aws-actions/amazon-ecr-login@v2
+        uses: aws-actions/amazon-ecr-login@62f4f872db3836360b72999f4b87f1ff13310f3a
+
+      - name: Set up QEMU
+        uses: docker/setup-qemu-action@v2

      - name: Set up Docker Buildx
-        uses: docker/setup-buildx-action@v3
+        uses: docker/setup-buildx-action@v2

-      - name: Build and push ${{ matrix.arch }}
-        uses: docker/build-push-action@v5
+      - name: Build and push
+        id: docker_build
+        uses: docker/build-push-action@v4
        with:
          context: server
-          platforms: ${{ matrix.platform }}
+          platforms: linux/amd64,linux/arm64
          push: true
-          tags: ${{ steps.login-ecr.outputs.registry }}/${{ env.ECR_REPOSITORY }}:latest-${{ matrix.arch }}
-          cache-from: type=gha,scope=${{ matrix.arch }}
-          cache-to: type=gha,mode=max,scope=${{ matrix.arch }}
-          github-token: ${{ secrets.GHA_CACHE_TOKEN }}
-          provenance: false
-
-  create-manifest:
-    runs-on: ubuntu-latest
-    needs: [build]
-
-    permissions:
-      deployments: write
-      contents: read
-
-    steps:
-      - name: Configure AWS credentials
-        uses: aws-actions/configure-aws-credentials@v4
-        with:
-          aws-access-key-id: ${{ secrets.AWS_ACCESS_KEY_ID }}
-          aws-secret-access-key: ${{ secrets.AWS_SECRET_ACCESS_KEY }}
-          aws-region: ${{ env.AWS_REGION }}
-
-      - name: Login to Amazon ECR
-        uses: aws-actions/amazon-ecr-login@v2
-
-      - name: Create and push multi-arch manifest
-        run: |
-          # Get the registry URL (since we can't easily access job outputs in matrix)
-          ECR_REGISTRY=$(aws ecr describe-registry --query 'registryId' --output text).dkr.ecr.${{ env.AWS_REGION }}.amazonaws.com
-
-          docker manifest create \
-            $ECR_REGISTRY/${{ env.ECR_REPOSITORY }}:latest \
-            $ECR_REGISTRY/${{ env.ECR_REPOSITORY }}:latest-amd64 \
-            $ECR_REGISTRY/${{ env.ECR_REPOSITORY }}:latest-arm64
-
-          docker manifest push $ECR_REGISTRY/${{ env.ECR_REPOSITORY }}:latest
-
-          echo "✅ Multi-arch manifest pushed: $ECR_REGISTRY/${{ env.ECR_REPOSITORY }}:latest"
+          tags: ${{ steps.login-ecr.outputs.registry }}/${{ env.ECR_REPOSITORY }}:latest
+          cache-from: type=gha
+          cache-to: type=gha,mode=max
--- a/.github/workflows/test_next_server.yml
+++ b/.github/workflows/test_next_server.yml
@@ -1,45 +0,0 @@
-name: Test Next Server
-
-on:
-  pull_request:
-    paths:
-      - "www/**"
-  push:
-    branches:
-      - main
-    paths:
-      - "www/**"
-
-jobs:
-  test-next-server:
-    runs-on: ubuntu-latest
-
-    defaults:
-      run:
-        working-directory: ./www
-
-    steps:
-    - uses: actions/checkout@v4
-
-    - name: Setup Node.js
-      uses: actions/setup-node@v4
-      with:
-        node-version: '20'
-
-    - name: Install pnpm
-      uses: pnpm/action-setup@v4
-      with:
-        version: 8
-
-    - name: Setup Node.js cache
-      uses: actions/setup-node@v4
-      with:
-        node-version: '20'
-        cache: 'pnpm'
-        cache-dependency-path: './www/pnpm-lock.yaml'
-
-    - name: Install dependencies
-      run: pnpm install
-
-    - name: Run tests
-      run: pnpm test
--- a/.github/workflows/test_server.yml
+++ b/.github/workflows/test_server.yml
@@ -5,17 +5,12 @@ on:
    paths:
      - "server/**"
  push:
-    branches:
-      - main
    paths:
      - "server/**"

 jobs:
  pytest:
    runs-on: ubuntu-latest
-    concurrency:
-      group: pytest-${{ github.ref }}
-      cancel-in-progress: true
    services:
      redis:
        image: redis:6
@@ -24,47 +19,29 @@ jobs:
    steps:
      - uses: actions/checkout@v4
      - name: Install uv
-        uses: astral-sh/setup-uv@v6
+        uses: astral-sh/setup-uv@v3
        with:
          enable-cache: true
          working-directory: server
+
      - name: Tests
        run: |
          cd server
          uv run -m pytest -v tests

-  docker-amd64:
-    runs-on: linux-amd64
-    concurrency:
-      group: docker-amd64-${{ github.ref }}
-      cancel-in-progress: true
+  docker:
+    runs-on: ubuntu-latest
    steps:
      - uses: actions/checkout@v4
+      - name: Set up QEMU
+        uses: docker/setup-qemu-action@v2
      - name: Set up Docker Buildx
-        uses: docker/setup-buildx-action@v3
-      - name: Build AMD64
-        uses: docker/build-push-action@v6
+        uses: docker/setup-buildx-action@v2
+      - name: Build and push
+        id: docker_build
+        uses: docker/build-push-action@v4
        with:
          context: server
-          platforms: linux/amd64
-          cache-from: type=gha,scope=amd64
-          cache-to: type=gha,mode=max,scope=amd64
-          github-token: ${{ secrets.GHA_CACHE_TOKEN }}
-
-  docker-arm64:
-    runs-on: linux-arm64
-    concurrency:
-      group: docker-arm64-${{ github.ref }}
-      cancel-in-progress: true
-    steps:
-      - uses: actions/checkout@v4
-      - name: Set up Docker Buildx
-        uses: docker/setup-buildx-action@v3
-      - name: Build ARM64
-        uses: docker/build-push-action@v6
-        with:
-          context: server
-          platforms: linux/arm64
-          cache-from: type=gha,scope=arm64
-          cache-to: type=gha,mode=max,scope=arm64
-          github-token: ${{ secrets.GHA_CACHE_TOKEN }}
+          platforms: linux/amd64,linux/arm64
+          cache-from: type=gha
+          cache-to: type=gha,mode=max
--- a/.gitignore
+++ b/.gitignore
@@ -14,7 +14,4 @@ data/
 www/REFACTOR.md
 www/reload-frontend
 server/test.sqlite
-CLAUDE.local.md
-www/.env.development
-www/.env.production
-.playwright-mcp
+CLAUDE.local.md
--- a/.gitleaksignore
+++ b/.gitleaksignore
@@ -1 +0,0 @@
-b9d891d3424f371642cb032ecfd0e2564470a72c:server/tests/test_transcripts_recording_deletion.py:generic-api-key:15
--- a/.pre-commit-config.yaml
+++ b/.pre-commit-config.yaml
@@ -27,8 +27,3 @@ repos:
        files: ^server/
      - id: ruff-format
        files: ^server/
-
-  - repo: https://github.com/gitleaks/gitleaks
-    rev: v8.28.0
-    hooks:
-      - id: gitleaks
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -1,106 +1,5 @@
 # Changelog

-## [0.10.0](https://github.com/Monadical-SAS/reflector/compare/v0.9.0...v0.10.0) (2025-09-11)
-
-
-### Features
-
-* replace nextjs-config with environment variables ([#632](https://github.com/Monadical-SAS/reflector/issues/632)) ([369ecdf](https://github.com/Monadical-SAS/reflector/commit/369ecdff13f3862d926a9c0b87df52c9d94c4dde))
-
-
-### Bug Fixes
-
-* anonymous users transcript permissions ([#621](https://github.com/Monadical-SAS/reflector/issues/621)) ([f81fe99](https://github.com/Monadical-SAS/reflector/commit/f81fe9948a9237b3e0001b2d8ca84f54d76878f9))
-* auth post ([#624](https://github.com/Monadical-SAS/reflector/issues/624)) ([cde99ca](https://github.com/Monadical-SAS/reflector/commit/cde99ca2716f84ba26798f289047732f0448742e))
-* auth post ([#626](https://github.com/Monadical-SAS/reflector/issues/626)) ([3b85ff3](https://github.com/Monadical-SAS/reflector/commit/3b85ff3bdf4fb053b103070646811bc990c0e70a))
-* auth post ([#627](https://github.com/Monadical-SAS/reflector/issues/627)) ([962038e](https://github.com/Monadical-SAS/reflector/commit/962038ee3f2a555dc3c03856be0e4409456e0996))
-* missing follow_redirects=True on modal endpoint ([#630](https://github.com/Monadical-SAS/reflector/issues/630)) ([fc363bd](https://github.com/Monadical-SAS/reflector/commit/fc363bd49b17b075e64f9186e5e0185abc325ea7))
-* sync backend and frontend token refresh logic ([#614](https://github.com/Monadical-SAS/reflector/issues/614)) ([5a5b323](https://github.com/Monadical-SAS/reflector/commit/5a5b3233820df9536da75e87ce6184a983d4713a))
-
-## [0.9.0](https://github.com/Monadical-SAS/reflector/compare/v0.8.2...v0.9.0) (2025-09-06)
-
-
-### Features
-
-* frontend openapi react query ([#606](https://github.com/Monadical-SAS/reflector/issues/606)) ([c4d2825](https://github.com/Monadical-SAS/reflector/commit/c4d2825c81f81ad8835629fbf6ea8c7383f8c31b))
-
-
-### Bug Fixes
-
-* align whisper transcriber api with parakeet ([#602](https://github.com/Monadical-SAS/reflector/issues/602)) ([0663700](https://github.com/Monadical-SAS/reflector/commit/0663700a615a4af69a03c96c410f049e23ec9443))
-* kv use tls explicit ([#610](https://github.com/Monadical-SAS/reflector/issues/610)) ([08d88ec](https://github.com/Monadical-SAS/reflector/commit/08d88ec349f38b0d13e0fa4cb73486c8dfd31836))
-* source kind for file processing ([#601](https://github.com/Monadical-SAS/reflector/issues/601)) ([dc82f8b](https://github.com/Monadical-SAS/reflector/commit/dc82f8bb3bdf3ab3d4088e592a30fd63907319e1))
-* token refresh locking ([#613](https://github.com/Monadical-SAS/reflector/issues/613)) ([7f5a4c9](https://github.com/Monadical-SAS/reflector/commit/7f5a4c9ddc7fd098860c8bdda2ca3b57f63ded2f))
-
-## [0.8.2](https://github.com/Monadical-SAS/reflector/compare/v0.8.1...v0.8.2) (2025-08-29)
-
-
-### Bug Fixes
-
-* search-logspam ([#593](https://github.com/Monadical-SAS/reflector/issues/593)) ([695d1a9](https://github.com/Monadical-SAS/reflector/commit/695d1a957d4cd862753049f9beed88836cabd5ab))
-
-## [0.8.1](https://github.com/Monadical-SAS/reflector/compare/v0.8.0...v0.8.1) (2025-08-29)
-
-
-### Bug Fixes
-
-* make webhook secret/url allowing null ([#590](https://github.com/Monadical-SAS/reflector/issues/590)) ([84a3812](https://github.com/Monadical-SAS/reflector/commit/84a381220bc606231d08d6f71d4babc818fa3c75))
-
-## [0.8.0](https://github.com/Monadical-SAS/reflector/compare/v0.7.3...v0.8.0) (2025-08-29)
-
-
-### Features
-
-* **cleanup:** add automatic data retention for public instances ([#574](https://github.com/Monadical-SAS/reflector/issues/574)) ([6f0c7c1](https://github.com/Monadical-SAS/reflector/commit/6f0c7c1a5e751713366886c8e764c2009e12ba72))
-* **rooms:** add webhook for transcript completion ([#578](https://github.com/Monadical-SAS/reflector/issues/578)) ([88ed7cf](https://github.com/Monadical-SAS/reflector/commit/88ed7cfa7804794b9b54cad4c3facc8a98cf85fd))
-
-
-### Bug Fixes
-
-* file pipeline status reporting and websocket updates ([#589](https://github.com/Monadical-SAS/reflector/issues/589)) ([9dfd769](https://github.com/Monadical-SAS/reflector/commit/9dfd76996f851cc52be54feea078adbc0816dc57))
-* Igor/evaluation ([#575](https://github.com/Monadical-SAS/reflector/issues/575)) ([124ce03](https://github.com/Monadical-SAS/reflector/commit/124ce03bf86044c18313d27228a25da4bc20c9c5))
-* optimize parakeet transcription batching algorithm ([#577](https://github.com/Monadical-SAS/reflector/issues/577)) ([7030e0f](https://github.com/Monadical-SAS/reflector/commit/7030e0f23649a8cf6c1eb6d5889684a41ce849ec))
-
-## [0.7.3](https://github.com/Monadical-SAS/reflector/compare/v0.7.2...v0.7.3) (2025-08-22)
-
-
-### Bug Fixes
-
-* cleaned repo, and get git-leaks clean ([359280d](https://github.com/Monadical-SAS/reflector/commit/359280dd340433ba4402ed69034094884c825e67))
-* restore previous behavior on live pipeline + audio downscaler ([#561](https://github.com/Monadical-SAS/reflector/issues/561)) ([9265d20](https://github.com/Monadical-SAS/reflector/commit/9265d201b590d23c628c5f19251b70f473859043))
-
-## [0.7.2](https://github.com/Monadical-SAS/reflector/compare/v0.7.1...v0.7.2) (2025-08-21)
-
-
-### Bug Fixes
-
-* docker image not loading libgomp.so.1 for torch ([#560](https://github.com/Monadical-SAS/reflector/issues/560)) ([773fccd](https://github.com/Monadical-SAS/reflector/commit/773fccd93e887c3493abc2e4a4864dddce610177))
-* include shared rooms to search ([#558](https://github.com/Monadical-SAS/reflector/issues/558)) ([499eced](https://github.com/Monadical-SAS/reflector/commit/499eced3360b84fb3a90e1c8a3b554290d21adc2))
-
-## [0.7.1](https://github.com/Monadical-SAS/reflector/compare/v0.7.0...v0.7.1) (2025-08-21)
-
-
-### Bug Fixes
-
-* webvtt db null expectation mismatch ([#556](https://github.com/Monadical-SAS/reflector/issues/556)) ([e67ad1a](https://github.com/Monadical-SAS/reflector/commit/e67ad1a4a2054467bfeb1e0258fbac5868aaaf21))
-
-## [0.7.0](https://github.com/Monadical-SAS/reflector/compare/v0.6.1...v0.7.0) (2025-08-21)
-
-
-### Features
-
-* delete recording with transcript ([#547](https://github.com/Monadical-SAS/reflector/issues/547)) ([99cc984](https://github.com/Monadical-SAS/reflector/commit/99cc9840b3f5de01e0adfbfae93234042d706d13))
-* pipeline improvement with file processing, parakeet, silero-vad ([#540](https://github.com/Monadical-SAS/reflector/issues/540)) ([bcc29c9](https://github.com/Monadical-SAS/reflector/commit/bcc29c9e0050ae215f89d460e9d645aaf6a5e486))
-* postgresql migration and removal of sqlite in pytest ([#546](https://github.com/Monadical-SAS/reflector/issues/546)) ([cd1990f](https://github.com/Monadical-SAS/reflector/commit/cd1990f8f0fe1503ef5069512f33777a73a93d7f))
-* search backend ([#537](https://github.com/Monadical-SAS/reflector/issues/537)) ([5f9b892](https://github.com/Monadical-SAS/reflector/commit/5f9b89260c9ef7f3c921319719467df22830453f))
-* search frontend  ([#551](https://github.com/Monadical-SAS/reflector/issues/551)) ([3657242](https://github.com/Monadical-SAS/reflector/commit/365724271ca6e615e3425125a69ae2b46ce39285))
-
-
-### Bug Fixes
-
-* evaluation cli event wrap ([#536](https://github.com/Monadical-SAS/reflector/issues/536)) ([941c3db](https://github.com/Monadical-SAS/reflector/commit/941c3db0bdacc7b61fea412f3746cc5a7cb67836))
-* use structlog not logging ([#550](https://github.com/Monadical-SAS/reflector/issues/550)) ([27e2f81](https://github.com/Monadical-SAS/reflector/commit/27e2f81fda5232e53edc729d3e99c5ef03adbfe9))
-
 ## [0.6.1](https://github.com/Monadical-SAS/reflector/compare/v0.6.0...v0.6.1) (2025-08-06)


--- a/CLAUDE.md
+++ b/CLAUDE.md
@@ -66,6 +66,7 @@ pnpm install

 # Copy configuration templates
 cp .env_template .env
+cp config-template.ts config.ts
 ```

 **Development:**
--- a/ICS_IMPLEMENTATION.md
+++ b/ICS_IMPLEMENTATION.md
@@ -0,0 +1,497 @@
+# ICS Calendar Integration - Implementation Guide
+
+## Overview
+This document provides detailed implementation guidance for integrating ICS calendar feeds with Reflector rooms. Unlike CalDAV which requires complex authentication and protocol handling, ICS integration uses simple HTTP(S) fetching of calendar files.
+
+## Key Differences from CalDAV Approach
+
+| Aspect | CalDAV | ICS |
+|--------|--------|-----|
+| Protocol | WebDAV extension | HTTP/HTTPS GET |
+| Authentication | Username/password, OAuth | Tokens embedded in URL |
+| Data Access | Selective event queries | Full calendar download |
+| Implementation | Complex (caldav library) | Simple (requests + icalendar) |
+| Real-time Updates | Supported | Polling only |
+| Write Access | Yes | No (read-only) |
+
+## Technical Architecture
+
+### 1. ICS Fetching Service
+
+```python
+# reflector/services/ics_sync.py
+
+import requests
+from icalendar import Calendar
+from typing import List, Optional
+from datetime import datetime, timedelta
+
+class ICSFetchService:
+    def __init__(self):
+        self.session = requests.Session()
+        self.session.headers.update({'User-Agent': 'Reflector/1.0'})
+
+    def fetch_ics(self, url: str) -> str:
+        """Fetch ICS file from URL (authentication via URL token if needed)."""
+        response = self.session.get(url, timeout=30)
+        response.raise_for_status()
+        return response.text
+
+    def parse_ics(self, ics_content: str) -> Calendar:
+        """Parse ICS content into calendar object."""
+        return Calendar.from_ical(ics_content)
+
+    def extract_room_events(self, calendar: Calendar, room_url: str) -> List[dict]:
+        """Extract events that match the room URL."""
+        events = []
+
+        for component in calendar.walk():
+            if component.name == "VEVENT":
+                # Check if event matches this room
+                if self._event_matches_room(component, room_url):
+                    events.append(self._parse_event(component))
+
+        return events
+
+    def _event_matches_room(self, event, room_url: str) -> bool:
+        """Check if event location or description contains room URL."""
+        location = str(event.get('LOCATION', ''))
+        description = str(event.get('DESCRIPTION', ''))
+
+        # Support various URL formats
+        patterns = [
+            room_url,
+            room_url.replace('https://', ''),
+            room_url.split('/')[-1],  # Just room name
+        ]
+
+        for pattern in patterns:
+            if pattern in location or pattern in description:
+                return True
+
+        return False
+```
+
+### 2. Database Schema
+
+```sql
+-- Modify room table
+ALTER TABLE room ADD COLUMN ics_url TEXT;  -- encrypted to protect embedded tokens
+ALTER TABLE room ADD COLUMN ics_fetch_interval INTEGER DEFAULT 300;  -- seconds
+ALTER TABLE room ADD COLUMN ics_enabled BOOLEAN DEFAULT FALSE;
+ALTER TABLE room ADD COLUMN ics_last_sync TIMESTAMP;
+ALTER TABLE room ADD COLUMN ics_last_etag TEXT;  -- for caching
+
+-- Calendar events table
+CREATE TABLE calendar_event (
+    id UUID PRIMARY KEY DEFAULT gen_random_uuid(),
+    room_id UUID REFERENCES room(id) ON DELETE CASCADE,
+    external_id TEXT NOT NULL,  -- ICS UID
+    title TEXT,
+    description TEXT,
+    start_time TIMESTAMP NOT NULL,
+    end_time TIMESTAMP NOT NULL,
+    attendees JSONB,
+    location TEXT,
+    ics_raw_data TEXT,  -- Store raw VEVENT for reference
+    last_synced TIMESTAMP DEFAULT NOW(),
+    is_deleted BOOLEAN DEFAULT FALSE,
+    created_at TIMESTAMP DEFAULT NOW(),
+    updated_at TIMESTAMP DEFAULT NOW(),
+    UNIQUE(room_id, external_id)
+);
+
+-- Index for efficient queries
+CREATE INDEX idx_calendar_event_room_start ON calendar_event(room_id, start_time);
+CREATE INDEX idx_calendar_event_deleted ON calendar_event(is_deleted) WHERE NOT is_deleted;
+```
+
+### 3. Background Tasks
+
+```python
+# reflector/worker/tasks/ics_sync.py
+
+from celery import shared_task
+from datetime import datetime, timedelta
+import hashlib
+
+@shared_task
+def sync_ics_calendars():
+    """Sync all enabled ICS calendars based on their fetch intervals."""
+    rooms = Room.query.filter_by(ics_enabled=True).all()
+
+    for room in rooms:
+        # Check if it's time to sync based on fetch interval
+        if should_sync(room):
+            sync_room_calendar.delay(room.id)
+
+@shared_task
+def sync_room_calendar(room_id: str):
+    """Sync calendar for a specific room."""
+    room = Room.query.get(room_id)
+    if not room or not room.ics_enabled:
+        return
+
+    try:
+        # Fetch ICS file (decrypt URL first)
+        service = ICSFetchService()
+        decrypted_url = decrypt_ics_url(room.ics_url)
+        ics_content = service.fetch_ics(decrypted_url)
+
+        # Check if content changed (using ETag or hash)
+        content_hash = hashlib.md5(ics_content.encode()).hexdigest()
+        if room.ics_last_etag == content_hash:
+            logger.info(f"No changes in ICS for room {room_id}")
+            return
+
+        # Parse and extract events
+        calendar = service.parse_ics(ics_content)
+        events = service.extract_room_events(calendar, room.url)
+
+        # Update database
+        sync_events_to_database(room_id, events)
+
+        # Update sync metadata
+        room.ics_last_sync = datetime.utcnow()
+        room.ics_last_etag = content_hash
+        db.session.commit()
+
+    except Exception as e:
+        logger.error(f"Failed to sync ICS for room {room_id}: {e}")
+
+def should_sync(room) -> bool:
+    """Check if room calendar should be synced."""
+    if not room.ics_last_sync:
+        return True
+
+    time_since_sync = datetime.utcnow() - room.ics_last_sync
+    return time_since_sync.total_seconds() >= room.ics_fetch_interval
+```
+
+### 4. Celery Beat Schedule
+
+```python
+# reflector/worker/celeryconfig.py
+
+from celery.schedules import crontab
+
+beat_schedule = {
+    'sync-ics-calendars': {
+        'task': 'reflector.worker.tasks.ics_sync.sync_ics_calendars',
+        'schedule': 60.0,  # Check every minute which calendars need syncing
+    },
+    'pre-create-meetings': {
+        'task': 'reflector.worker.tasks.ics_sync.pre_create_calendar_meetings',
+        'schedule': 60.0,  # Check every minute for upcoming meetings
+    },
+}
+```
+
+## API Endpoints
+
+### Room ICS Configuration
+
+```python
+# PATCH /v1/rooms/{room_id}
+{
+    "ics_url": "https://calendar.google.com/calendar/ical/.../private-token/basic.ics",
+    "ics_fetch_interval": 300,  # seconds
+    "ics_enabled": true
+    # URL will be encrypted in database to protect embedded tokens
+}
+```
+
+### Manual Sync Trigger
+
+```python
+# POST /v1/rooms/{room_name}/ics/sync
+# Response:
+{
+    "status": "syncing",
+    "last_sync": "2024-01-15T10:30:00Z",
+    "events_found": 5
+}
+```
+
+### ICS Status
+
+```python
+# GET /v1/rooms/{room_name}/ics/status
+# Response:
+{
+    "enabled": true,
+    "last_sync": "2024-01-15T10:30:00Z",
+    "next_sync": "2024-01-15T10:35:00Z",
+    "fetch_interval": 300,
+    "events_count": 12,
+    "upcoming_events": 3
+}
+```
+
+## ICS Parsing Details
+
+### Event Field Mapping
+
+| ICS Field | Database Field | Notes |
+|-----------|---------------|-------|
+| UID | external_id | Unique identifier |
+| SUMMARY | title | Event title |
+| DESCRIPTION | description | Full description |
+| DTSTART | start_time | Convert to UTC |
+| DTEND | end_time | Convert to UTC |
+| LOCATION | location | Check for room URL |
+| ATTENDEE | attendees | Parse into JSON |
+| ORGANIZER | attendees | Add as organizer |
+| STATUS | (internal) | Filter cancelled events |
+
+### Handling Recurring Events
+
+```python
+def expand_recurring_events(event, start_date, end_date):
+    """Expand recurring events into individual occurrences."""
+    from dateutil.rrule import rrulestr
+
+    if 'RRULE' not in event:
+        return [event]
+
+    # Parse recurrence rule
+    rrule_str = event['RRULE'].to_ical().decode()
+    dtstart = event['DTSTART'].dt
+
+    # Generate occurrences
+    rrule = rrulestr(rrule_str, dtstart=dtstart)
+    occurrences = []
+
+    for dt in rrule.between(start_date, end_date):
+        # Clone event with new date
+        occurrence = event.copy()
+        occurrence['DTSTART'].dt = dt
+        if 'DTEND' in event:
+            duration = event['DTEND'].dt - event['DTSTART'].dt
+            occurrence['DTEND'].dt = dt + duration
+
+        # Unique ID for each occurrence
+        occurrence['UID'] = f"{event['UID']}_{dt.isoformat()}"
+        occurrences.append(occurrence)
+
+    return occurrences
+```
+
+### Timezone Handling
+
+```python
+def normalize_datetime(dt):
+    """Convert various datetime formats to UTC."""
+    import pytz
+    from datetime import datetime
+
+    if hasattr(dt, 'dt'):  # icalendar property
+        dt = dt.dt
+
+    if isinstance(dt, datetime):
+        if dt.tzinfo is None:
+            # Assume local timezone if naive
+            dt = pytz.timezone('UTC').localize(dt)
+        else:
+            # Convert to UTC
+            dt = dt.astimezone(pytz.UTC)
+
+    return dt
+```
+
+## Security Considerations
+
+### 1. URL Validation
+
+```python
+def validate_ics_url(url: str) -> bool:
+    """Validate ICS URL for security."""
+    from urllib.parse import urlparse
+
+    parsed = urlparse(url)
+
+    # Must be HTTPS in production
+    if not settings.DEBUG and parsed.scheme != 'https':
+        return False
+
+    # Prevent local file access
+    if parsed.scheme in ('file', 'ftp'):
+        return False
+
+    # Prevent internal network access
+    if is_internal_ip(parsed.hostname):
+        return False
+
+    return True
+```
+
+### 2. Rate Limiting
+
+```python
+# Implement per-room rate limiting
+RATE_LIMITS = {
+    'min_fetch_interval': 60,  # Minimum 1 minute between fetches
+    'max_requests_per_hour': 60,  # Max 60 requests per hour per room
+    'max_file_size': 10 * 1024 * 1024,  # Max 10MB ICS file
+}
+```
+
+### 3. ICS URL Encryption
+
+```python
+from cryptography.fernet import Fernet
+
+class URLEncryption:
+    def __init__(self):
+        self.cipher = Fernet(settings.ENCRYPTION_KEY)
+
+    def encrypt_url(self, url: str) -> str:
+        """Encrypt ICS URL to protect embedded tokens."""
+        return self.cipher.encrypt(url.encode()).decode()
+
+    def decrypt_url(self, encrypted: str) -> str:
+        """Decrypt ICS URL for fetching."""
+        return self.cipher.decrypt(encrypted.encode()).decode()
+
+    def mask_url(self, url: str) -> str:
+        """Mask sensitive parts of URL for display."""
+        from urllib.parse import urlparse, urlunparse
+
+        parsed = urlparse(url)
+        # Keep scheme, host, and path structure but mask tokens
+        if '/private-' in parsed.path:
+            # Google Calendar format
+            parts = parsed.path.split('/private-')
+            masked_path = parts[0] + '/private-***' + parts[1].split('/')[-1]
+        elif 'token=' in url:
+            # Query parameter token
+            masked_path = parsed.path
+            parsed = parsed._replace(query='token=***')
+        else:
+            # Generic masking of path segments that look like tokens
+            import re
+            masked_path = re.sub(r'/[a-zA-Z0-9]{20,}/', '/***/', parsed.path)
+
+        return urlunparse(parsed._replace(path=masked_path))
+```
+
+## Testing Strategy
+
+### 1. Unit Tests
+
+```python
+# tests/test_ics_sync.py
+
+def test_ics_parsing():
+    """Test ICS file parsing."""
+    ics_content = """BEGIN:VCALENDAR
+VERSION:2.0
+BEGIN:VEVENT
+UID:test-123
+SUMMARY:Team Meeting
+LOCATION:https://reflector.monadical.com/engineering
+DTSTART:20240115T100000Z
+DTEND:20240115T110000Z
+END:VEVENT
+END:VCALENDAR"""
+
+    service = ICSFetchService()
+    calendar = service.parse_ics(ics_content)
+    events = service.extract_room_events(
+        calendar,
+        "https://reflector.monadical.com/engineering"
+    )
+
+    assert len(events) == 1
+    assert events[0]['title'] == 'Team Meeting'
+```
+
+### 2. Integration Tests
+
+```python
+def test_full_sync_flow():
+    """Test complete sync workflow."""
+    # Create room with ICS URL (encrypt URL to protect tokens)
+    encryption = URLEncryption()
+    room = Room(
+        name="test-room",
+        ics_url=encryption.encrypt_url("https://example.com/calendar.ics?token=secret"),
+        ics_enabled=True
+    )
+
+    # Mock ICS fetch
+    with patch('requests.get') as mock_get:
+        mock_get.return_value.text = sample_ics_content
+
+        # Run sync
+        sync_room_calendar(room.id)
+
+        # Verify events created
+        events = CalendarEvent.query.filter_by(room_id=room.id).all()
+        assert len(events) > 0
+```
+
+## Common ICS Provider Configurations
+
+### Google Calendar
+- URL Format: `https://calendar.google.com/calendar/ical/{calendar_id}/private-{token}/basic.ics`
+- Authentication via token embedded in URL
+- Updates every 3-8 hours by default
+
+### Outlook/Office 365
+- URL Format: `https://outlook.office365.com/owa/calendar/{id}/calendar.ics`
+- May include token in URL path or query parameters
+- Real-time updates
+
+### Apple iCloud
+- URL Format: `webcal://p{XX}-caldav.icloud.com/published/2/{token}`
+- Convert webcal:// to https://
+- Token embedded in URL path
+- Public calendars only
+
+### Nextcloud/ownCloud
+- URL Format: `https://cloud.example.com/remote.php/dav/public-calendars/{token}`
+- Token embedded in URL path
+- Configurable update frequency
+
+## Migration from CalDAV
+
+If migrating from an existing CalDAV implementation:
+
+1. **Database Migration**: Rename fields from `caldav_*` to `ics_*`
+2. **URL Conversion**: Most CalDAV servers provide ICS export endpoints
+3. **Authentication**: Convert from username/password to URL-embedded tokens
+4. **Remove Dependencies**: Uninstall caldav library, add icalendar
+5. **Update Background Tasks**: Replace CalDAV sync with ICS fetch
+
+## Performance Optimizations
+
+1. **Caching**: Use ETag/Last-Modified headers to avoid refetching unchanged calendars
+2. **Incremental Sync**: Store last sync timestamp, only process new/modified events
+3. **Batch Processing**: Process multiple room calendars in parallel
+4. **Connection Pooling**: Reuse HTTP connections for multiple requests
+5. **Compression**: Support gzip encoding for large ICS files
+
+## Monitoring and Debugging
+
+### Metrics to Track
+- Sync success/failure rate per room
+- Average sync duration
+- ICS file sizes
+- Number of events processed
+- Failed event matches
+
+### Debug Logging
+```python
+logger.debug(f"Fetching ICS from {room.ics_url}")
+logger.debug(f"ICS content size: {len(ics_content)} bytes")
+logger.debug(f"Found {len(events)} matching events")
+logger.debug(f"Event UIDs: {[e['external_id'] for e in events]}")
+```
+
+### Common Issues
+1. **SSL Certificate Errors**: Add certificate validation options
+2. **Timeout Issues**: Increase timeout for large calendars
+3. **Encoding Problems**: Handle various character encodings
+4. **Timezone Mismatches**: Always convert to UTC
+5. **Memory Issues**: Stream large ICS files instead of loading entirely
--- a/PLAN.md
+++ b/PLAN.md
@@ -0,0 +1,337 @@
+# ICS Calendar Integration Plan
+
+## Core Concept
+ICS calendar URLs are attached to rooms (not users) to enable automatic meeting tracking and management through periodic fetching of calendar data.
+
+## Database Schema Updates
+
+### 1. Add ICS configuration to rooms
+- Add `ics_url` field to room table (URL to .ics file, may include auth token)
+- Add `ics_fetch_interval` field to room table (default: 5 minutes, configurable)
+- Add `ics_enabled` boolean field to room table
+- Add `ics_last_sync` timestamp field to room table
+
+### 2. Create calendar_events table
+- `id` - UUID primary key
+- `room_id` - Foreign key to room
+- `external_id` - ICS event UID
+- `title` - Event title
+- `description` - Event description
+- `start_time` - Event start timestamp
+- `end_time` - Event end timestamp
+- `attendees` - JSON field with attendee list and status
+- `location` - Meeting location (should contain room name)
+- `last_synced` - Last sync timestamp
+- `is_deleted` - Boolean flag for soft delete (preserve past events)
+- `ics_raw_data` - TEXT field to store raw VEVENT data for reference
+
+### 3. Update meeting table
+- Add `calendar_event_id` - Foreign key to calendar_events
+- Add `calendar_metadata` - JSON field for additional calendar data
+- Remove unique constraint on room_id + active status (allow multiple active meetings per room)
+
+## Backend Implementation
+
+### 1. ICS Sync Service
+- Create background task that runs based on room's `ics_fetch_interval` (default: 5 minutes)
+- For each room with ICS enabled, fetch the .ics file via HTTP/HTTPS
+- Parse ICS file using icalendar library
+- Extract VEVENT components and filter events looking for room URL (e.g., "https://reflector.monadical.com/max")
+- Store matching events in calendar_events table
+- Mark events as "upcoming" if start_time is within next 30 minutes
+- Pre-create Whereby meetings 1 minute before start (ensures no delay when users join)
+- Soft-delete future events that were removed from calendar (set is_deleted=true)
+- Never delete past events (preserve for historical record)
+- Support authenticated ICS feeds via tokens embedded in URL
+
+### 2. Meeting Management Updates
+- Allow multiple active meetings per room
+- Pre-create meeting record 1 minute before calendar event starts (ensures meeting is ready)
+- Link meeting to calendar_event for metadata
+- Keep meeting active for 15 minutes after last participant leaves (grace period)
+- Don't auto-close if new participant joins within grace period
+
+### 3. API Endpoints
+- `GET /v1/rooms/{room_name}/meetings` - List all active and upcoming meetings for a room
+  - Returns filtered data based on user role (owner vs participant)
+- `GET /v1/rooms/{room_name}/meetings/upcoming` - List upcoming meetings (next 30 min)
+  - Returns filtered data based on user role
+- `POST /v1/rooms/{room_name}/meetings/{meeting_id}/join` - Join specific meeting
+- `PATCH /v1/rooms/{room_id}` - Update room settings (including ICS configuration)
+  - ICS fields only visible/editable by room owner
+- `POST /v1/rooms/{room_name}/ics/sync` - Trigger manual ICS sync
+  - Only accessible by room owner
+- `GET /v1/rooms/{room_name}/ics/status` - Get ICS sync status and last fetch time
+  - Only accessible by room owner
+
+## Frontend Implementation
+
+### 1. Room Settings Page
+- Add ICS configuration section
+- Field for ICS URL (e.g., Google Calendar private URL, Outlook ICS export)
+- Field for fetch interval (dropdown: 1 min, 5 min, 10 min, 30 min, 1 hour)
+- Test connection button (validates ICS file can be fetched and parsed)
+- Manual sync button
+- Show last sync time and next scheduled sync
+
+### 2. Meeting Selection Page (New)
+- Show when accessing `/room/{room_name}`
+- **Host view** (room owner):
+  - Full calendar event details
+  - Meeting title and description
+  - Complete attendee list with RSVP status
+  - Number of current participants
+  - Duration (how long it's been running)
+- **Participant view** (non-owners):
+  - Meeting title only
+  - Date and time
+  - Number of current participants
+  - Duration (how long it's been running)
+  - No attendee list or description (privacy)
+- Display upcoming meetings (visible 30min before):
+  - Show countdown to start
+  - Can click to join early → redirected to waiting page
+  - Waiting page shows countdown until meeting starts
+  - Meeting pre-created by background task (ready when users arrive)
+- Option to create unscheduled meeting (uses existing flow)
+
+### 3. Meeting Room Updates
+- Show calendar metadata in meeting info
+- Display invited attendees vs actual participants
+- Show meeting title from calendar event
+
+## Meeting Lifecycle
+
+### 1. Meeting Creation
+- Automatic: Pre-created 1 minute before calendar event starts (ensures Whereby room is ready)
+- Manual: User creates unscheduled meeting (existing `/rooms/{room_name}/meeting` endpoint)
+- Background task handles pre-creation to avoid delays when users join
+
+### 2. Meeting Join Rules
+- Can join active meetings immediately
+- Can see upcoming meetings 30 minutes before start
+- Can click to join upcoming meetings early → sent to waiting page
+- Waiting page automatically transitions to meeting at scheduled time
+- Unscheduled meetings always joinable (current behavior)
+
+### 3. Meeting Closure Rules
+- All meetings: 15-minute grace period after last participant leaves
+- If participant rejoins within grace period, keep meeting active
+- Calendar meetings: Force close 30 minutes after scheduled end time
+- Unscheduled meetings: Keep active for 8 hours (current behavior)
+
+## ICS Parsing Logic
+
+### 1. Event Matching
+- Parse ICS file using Python icalendar library
+- Iterate through VEVENT components
+- Check LOCATION field for full FQDN URL (e.g., "https://reflector.monadical.com/max")
+- Check DESCRIPTION for room URL or mention
+- Support multiple formats:
+  - Full URL: "https://reflector.monadical.com/max"
+  - With /room path: "https://reflector.monadical.com/room/max"
+  - Partial paths: "room/max", "/max room"
+
+### 2. Attendee Extraction
+- Parse ATTENDEE properties from VEVENT
+- Extract email (MAILTO), name (CN parameter), and RSVP status (PARTSTAT)
+- Store as JSON in calendar_events.attendees
+
+### 3. Sync Strategy
+- Fetch complete ICS file (contains all events)
+- Filter events from (now - 1 hour) to (now + 24 hours) for processing
+- Update existing events if LAST-MODIFIED or SEQUENCE changed
+- Delete future events that no longer exist in ICS (start_time > now)
+- Keep past events for historical record (never delete if start_time < now)
+- Handle recurring events (RRULE) - expand to individual instances
+- Track deleted calendar events to clean up future meetings
+- Cache ICS file hash to detect changes and skip unnecessary processing
+
+## Security Considerations
+
+### 1. ICS URL Security
+- ICS URLs may contain authentication tokens (e.g., Google Calendar private URLs)
+- Store full ICS URLs encrypted using Fernet to protect embedded tokens
+- Validate ICS URLs (must be HTTPS for production)
+- Never expose full ICS URLs in API responses (return masked version)
+- Rate limit ICS fetching to prevent abuse
+
+### 2. Room Access
+- Only room owner can configure ICS URL
+- ICS URL shown as masked version to room owner (hides embedded tokens)
+- ICS settings not visible to other users
+- Meeting list visible to all room participants
+- ICS fetch logs only visible to room owner
+
+### 3. Meeting Privacy
+- Full calendar details visible only to room owner
+- Participants see limited info: title, date/time only
+- Attendee list and description hidden from non-owners
+- Meeting titles visible in room listing to all
+
+## Implementation Phases
+
+### Phase 1: Database and ICS Setup (Week 1) ✅ COMPLETED (2025-08-18)
+1. ✅ Created database migrations for ICS fields and calendar_events table
+   - Added ics_url, ics_fetch_interval, ics_enabled, ics_last_sync, ics_last_etag to room table
+   - Created calendar_event table with ics_uid (instead of external_id) and proper typing
+   - Added calendar_event_id and calendar_metadata (JSONB) to meeting table
+   - Removed server_default from datetime fields for consistency
+2. ✅ Installed icalendar Python library for ICS parsing
+   - Added icalendar>=6.0.0 to dependencies
+   - No encryption needed - ICS URLs are read-only
+3. ✅ Built ICS fetch and sync service
+   - Simple HTTP fetching without unnecessary validation
+   - Proper TypedDict typing for event data structures
+   - Supports any standard ICS format
+   - Event matching on full room URL only
+4. ✅ API endpoints for ICS configuration
+   - Room model updated to support ICS fields via existing PATCH endpoint
+   - POST /v1/rooms/{room_name}/ics/sync - Trigger manual sync (owner only)
+   - GET /v1/rooms/{room_name}/ics/status - Get sync status (owner only)
+   - GET /v1/rooms/{room_name}/meetings - List meetings with privacy controls
+   - GET /v1/rooms/{room_name}/meetings/upcoming - List upcoming meetings
+5. ✅ Celery background tasks for periodic sync
+   - sync_room_ics - Sync individual room calendar
+   - sync_all_ics_calendars - Check all rooms and queue sync based on fetch intervals
+   - pre_create_upcoming_meetings - Pre-create Whereby meetings 1 minute before start
+   - Tasks scheduled in beat schedule (every minute for checking, respects individual intervals)
+6. ✅ Tests written and passing
+   - 6 tests for Room ICS fields
+   - 7 tests for CalendarEvent model
+   - 7 tests for ICS sync service
+   - 11 tests for API endpoints
+   - 6 tests for background tasks
+   - All 31 ICS-related tests passing
+
+### Phase 2: Meeting Management (Week 2) ✅ COMPLETED (2025-08-19)
+1. ✅ Updated meeting lifecycle logic with grace period support
+   - 15-minute grace period after last participant leaves
+   - Automatic reactivation when participants rejoin
+   - Force close calendar meetings 30 minutes after scheduled end
+2. ✅ Support multiple active meetings per room
+   - Removed unique constraint on active meetings
+   - Added get_all_active_for_room() method
+   - Added get_active_by_calendar_event() method
+3. ✅ Implemented grace period logic
+   - Added last_participant_left_at and grace_period_minutes fields
+   - Process meetings task handles grace period checking
+   - Whereby webhooks clear grace period on participant join
+4. ✅ Link meetings to calendar events
+   - Pre-created meetings properly linked via calendar_event_id
+   - Calendar metadata stored with meeting
+   - API endpoints for listing and joining specific meetings
+
+### Phase 3: Frontend Meeting Selection (Week 3)
+1. Build meeting selection page
+2. Show active and upcoming meetings
+3. Implement waiting page for early joiners
+4. Add automatic transition from waiting to meeting
+5. Support unscheduled meeting creation
+
+### Phase 4: Calendar Integration UI (Week 4)
+1. Add ICS settings to room configuration
+2. Display calendar metadata in meetings
+3. Show attendee information
+4. Add sync status indicators
+5. Show fetch interval and next sync time
+
+## Success Metrics
+- Zero merged meetings from consecutive calendar events
+- Successful ICS sync from major providers (Google Calendar, Outlook, Apple Calendar, Nextcloud)
+- Meeting join accuracy: correct meeting 100% of the time
+- Grace period prevents 90% of accidental meeting closures
+- Configurable fetch intervals reduce unnecessary API calls
+
+## Design Decisions
+1. **ICS attached to room, not user** - Prevents duplicate meetings from multiple calendars
+2. **Multiple active meetings per room** - Supported with meeting selection page
+3. **Grace period for rejoining** - 15 minutes after last participant leaves
+4. **Upcoming meeting visibility** - Show 30 minutes before, join only on time
+5. **Calendar data storage** - Attached to meeting record for full context
+6. **No "ad-hoc" meetings** - Use existing meeting creation flow (unscheduled meetings)
+7. **ICS configuration via room PATCH** - Reuse existing room configuration endpoint
+8. **Event deletion handling** - Soft-delete future events, preserve past meetings
+9. **Configurable fetch interval** - Balance between freshness and server load
+10. **ICS over CalDAV** - Simpler implementation, wider compatibility, no complex auth
+
+## Phase 2 Implementation Files
+
+### Database Migrations
+- `/server/migrations/versions/6025e9b2bef2_remove_one_active_meeting_per_room_.py` - Remove unique constraint
+- `/server/migrations/versions/d4a1c446458c_add_grace_period_fields_to_meeting.py` - Add grace period fields
+
+### Updated Models
+- `/server/reflector/db/meetings.py` - Added grace period fields and new query methods
+
+### Updated Services
+- `/server/reflector/worker/process.py` - Enhanced with grace period logic and multiple meeting support
+
+### Updated API
+- `/server/reflector/views/rooms.py` - Added endpoints for listing active meetings and joining specific meetings
+- `/server/reflector/views/whereby.py` - Clear grace period on participant join
+
+### Tests
+- `/server/tests/test_multiple_active_meetings.py` - Comprehensive tests for Phase 2 features (5 tests)
+
+## Phase 1 Implementation Files Created
+
+### Database Models
+- `/server/reflector/db/rooms.py` - Updated with ICS fields (url, fetch_interval, enabled, last_sync, etag)
+- `/server/reflector/db/calendar_events.py` - New CalendarEvent model with ics_uid and proper typing
+- `/server/reflector/db/meetings.py` - Updated with calendar_event_id and calendar_metadata (JSONB)
+
+### Services
+- `/server/reflector/services/ics_sync.py` - ICS fetching and parsing with TypedDict for proper typing
+
+### API Endpoints
+- `/server/reflector/views/rooms.py` - Added ICS management endpoints with privacy controls
+
+### Background Tasks
+- `/server/reflector/worker/ics_sync.py` - Celery tasks for automatic periodic sync
+- `/server/reflector/worker/app.py` - Updated beat schedule for ICS tasks
+
+### Tests
+- `/server/tests/test_room_ics.py` - Room model ICS fields tests (6 tests)
+- `/server/tests/test_calendar_event.py` - CalendarEvent model tests (7 tests)
+- `/server/tests/test_ics_sync.py` - ICS sync service tests (7 tests)
+- `/server/tests/test_room_ics_api.py` - API endpoint tests (11 tests)
+- `/server/tests/test_ics_background_tasks.py` - Background task tests (6 tests)
+
+### Key Design Decisions
+- No encryption needed - ICS URLs are read-only access
+- Using ics_uid instead of external_id for clarity
+- Proper TypedDict typing for event data structures
+- Removed unnecessary URL validation and webcal handling
+- calendar_metadata in meetings stores flexible calendar data (organizer, recurrence, etc)
+- Background tasks query all rooms directly to avoid filtering issues
+- Sync intervals respected per-room configuration
+
+## Implementation Approach
+
+### ICS Fetching vs CalDAV
+- **ICS Benefits**:
+  - Simpler implementation (HTTP GET vs CalDAV protocol)
+  - Wider compatibility (all calendar apps can export ICS)
+  - No authentication complexity (simple URL with optional token)
+  - Easier debugging (ICS is plain text)
+  - Lower server requirements (no CalDAV library dependencies)
+
+### Supported Calendar Providers
+1. **Google Calendar**: Private ICS URL from calendar settings
+2. **Outlook/Office 365**: ICS export URL from calendar sharing
+3. **Apple Calendar**: Published calendar ICS URL
+4. **Nextcloud**: Public/private calendar ICS export
+5. **Any CalDAV server**: Via ICS export endpoint
+
+### ICS URL Examples
+- Google: `https://calendar.google.com/calendar/ical/{calendar_id}/private-{token}/basic.ics`
+- Outlook: `https://outlook.live.com/owa/calendar/{id}/calendar.ics`
+- Custom: `https://example.com/calendars/room-schedule.ics`
+
+### Fetch Interval Configuration
+- 1 minute: For critical/high-activity rooms
+- 5 minutes (default): Balance of freshness and efficiency
+- 10 minutes: Standard meeting rooms
+- 30 minutes: Low-activity rooms
+- 1 hour: Rarely-used rooms or stable schedules
--- a/README.md
+++ b/README.md
@@ -1,60 +1,43 @@
 <div align="center">
-<img width="100" alt="image" src="https://github.com/user-attachments/assets/66fb367b-2c89-4516-9912-f47ac59c6a7f"/>

 # Reflector

-Reflector is an AI-powered audio transcription and meeting analysis platform that provides real-time transcription, speaker diarization, translation and summarization for audio content and live meetings. It works 100% with local models (whisper/parakeet, pyannote, seamless-m4t, and your local llm like phi-4).
+Reflector Audio Management and Analysis is a cutting-edge web application under development by Monadical. It utilizes AI to record meetings, providing a permanent record with transcripts, translations, and automated summaries.

-[![Tests](https://github.com/monadical-sas/reflector/actions/workflows/test_server.yml/badge.svg?branch=main&event=push)](https://github.com/monadical-sas/reflector/actions/workflows/test_server.yml)
+[![Tests](https://github.com/monadical-sas/reflector/actions/workflows/pytests.yml/badge.svg?branch=main&event=push)](https://github.com/monadical-sas/reflector/actions/workflows/pytests.yml)
 [![License: MIT](https://img.shields.io/badge/license-MIT-green.svg)](https://opensource.org/licenses/MIT)
 </div>
-</div>
+
+## Screenshots
 <table>
  <tr>
    <td>
-      <a href="https://github.com/user-attachments/assets/21f5597c-2930-4899-a154-f7bd61a59e97">
-        <img width="700" alt="image" src="https://github.com/user-attachments/assets/21f5597c-2930-4899-a154-f7bd61a59e97" />
+      <a href="https://github.com/user-attachments/assets/3a976930-56c1-47ef-8c76-55d3864309e3">
+        <img width="700" alt="image" src="https://github.com/user-attachments/assets/3a976930-56c1-47ef-8c76-55d3864309e3" />
      </a>
    </td>
    <td>
-      <a href="https://github.com/user-attachments/assets/f6b9399a-5e51-4bae-b807-59128d0a940c">
-        <img width="700" alt="image" src="https://github.com/user-attachments/assets/f6b9399a-5e51-4bae-b807-59128d0a940c" />
+      <a href="https://github.com/user-attachments/assets/bfe3bde3-08af-4426-a9a1-11ad5cd63b33">
+        <img width="700" alt="image" src="https://github.com/user-attachments/assets/bfe3bde3-08af-4426-a9a1-11ad5cd63b33" />
      </a>
    </td>
    <td>
-      <a href="https://github.com/user-attachments/assets/a42ce460-c1fd-4489-a995-270516193897">
-        <img width="700" alt="image" src="https://github.com/user-attachments/assets/a42ce460-c1fd-4489-a995-270516193897" />
-      </a>
-    </td>
-    <td>
-      <a href="https://github.com/user-attachments/assets/21929f6d-c309-42fe-9c11-f1299e50fbd4">
-        <img width="700" alt="image" src="https://github.com/user-attachments/assets/21929f6d-c309-42fe-9c11-f1299e50fbd4" />
+      <a href="https://github.com/user-attachments/assets/7b60c9d0-efe4-474f-a27b-ea13bd0fabdc">
+        <img width="700" alt="image" src="https://github.com/user-attachments/assets/7b60c9d0-efe4-474f-a27b-ea13bd0fabdc" />
      </a>
    </td>
  </tr>
 </table>

-## What is Reflector?
-
-Reflector is a web application that utilizes local models to process audio content, providing:
-
- **Real-time Transcription**: Convert speech to text using [Whisper](https://github.com/openai/whisper) (multi-language) or [Parakeet](https://huggingface.co/nvidia/parakeet-tdt-0.6b-v2) (English) models
- **Speaker Diarization**: Identify and label different speakers using [Pyannote](https://github.com/pyannote/pyannote-audio) 3.1
- **Live Translation**: Translate audio content in real-time to many languages with [Facebook Seamless-M4T](https://github.com/facebookresearch/seamless_communication)
- **Topic Detection & Summarization**: Extract key topics and generate concise summaries using LLMs
- **Meeting Recording**: Create permanent records of meetings with searchable transcripts
-
-Currently we provide [modal.com](https://modal.com/) gpu template to deploy.
-
 ## Background

 The project architecture consists of three primary components:

- **Back-End**: Python server that offers an API and data persistence, found in `server/`.
 - **Front-End**: NextJS React project hosted on Vercel, located in `www/`.
- **GPU implementation**: Providing services such as speech-to-text transcription, topic generation, automated summaries, and translations.
+- **Back-End**: Python server that offers an API and data persistence, found in `server/`.
+- **GPU implementation**: Providing services such as speech-to-text transcription, topic generation, automated summaries, and translations. Most reliable option is Modal deployment

-It also uses authentik for authentication if activated.
+It also uses authentik for authentication if activated, and Vercel for deployment and configuration of the front-end.

 ## Contribution Guidelines

@@ -89,8 +72,6 @@ Note: We currently do not have instructions for Windows users.

 ## Installation

-*Note: we're working toward better installation, theses instructions are not accurate for now*
-
 ### Frontend

 Start with `cd www`.
@@ -99,10 +80,11 @@ Start with `cd www`.

 ```bash
 pnpm install
-cp .env.example .env
+cp .env_template .env
+cp config-template.ts config.ts
 ```

-Then, fill in the environment variables in `.env` as needed. If you are unsure on how to proceed, ask in Zulip.
+Then, fill in the environment variables in `.env` and the configuration in `config.ts` as needed. If you are unsure on how to proceed, ask in Zulip.

 **Run in development mode**

@@ -167,34 +149,3 @@ You can manually process an audio file by calling the process tool:
 ```bash
 uv run python -m reflector.tools.process path/to/audio.wav
 ```
-
-
-## Feature Flags
-
-Reflector uses environment variable-based feature flags to control application functionality. These flags allow you to enable or disable features without code changes.
-
-### Available Feature Flags
-
-| Feature Flag | Environment Variable |
-|-------------|---------------------|
-| `requireLogin` | `NEXT_PUBLIC_FEATURE_REQUIRE_LOGIN` |
-| `privacy` | `NEXT_PUBLIC_FEATURE_PRIVACY` |
-| `browse` | `NEXT_PUBLIC_FEATURE_BROWSE` |
-| `sendToZulip` | `NEXT_PUBLIC_FEATURE_SEND_TO_ZULIP` |
-| `rooms` | `NEXT_PUBLIC_FEATURE_ROOMS` |
-
-### Setting Feature Flags
-
-Feature flags are controlled via environment variables using the pattern `NEXT_PUBLIC_FEATURE_{FEATURE_NAME}` where `{FEATURE_NAME}` is the SCREAMING_SNAKE_CASE version of the feature name.
-
-**Examples:**
-```bash
-# Enable user authentication requirement
-NEXT_PUBLIC_FEATURE_REQUIRE_LOGIN=true
-
-# Disable browse functionality
-NEXT_PUBLIC_FEATURE_BROWSE=false
-
-# Enable Zulip integration
-NEXT_PUBLIC_FEATURE_SEND_TO_ZULIP=true
-```
--- a/compose.yml
+++ b/compose.yml
@@ -6,7 +6,6 @@ services:
      - 1250:1250
    volumes:
      - ./server/:/app/
-      - /app/.venv
    env_file:
      - ./server/.env
    environment:
@@ -17,7 +16,6 @@ services:
      context: server
    volumes:
      - ./server/:/app/
-      - /app/.venv
    env_file:
      - ./server/.env
    environment:
@@ -28,7 +26,6 @@ services:
      context: server
    volumes:
      - ./server/:/app/
-      - /app/.venv
    env_file:
      - ./server/.env
    environment:
--- a/docs/jitsi.md
+++ b/docs/jitsi.md
@@ -1,369 +0,0 @@
-# Jitsi Integration for Reflector
-
-This document contains research and planning notes for integrating Jitsi Meet as a replacement for Whereby in Reflector.
-
-## Overview
-
-Jitsi Meet is an open-source video conferencing solution that can replace Whereby in Reflector, providing:
- Cost reduction (no per-minute charges)
- Direct recording access via Jibri
- Real-time event webhooks
- Full customization and control
-
-## Current Whereby Integration Analysis
-
-### Architecture
-1. **Room Creation**: User creates a "room" template in Reflector DB with settings
-2. **Meeting Creation**: `/rooms/{room_name}/meeting` endpoint calls Whereby API to create meeting
-3. **Recording**: Whereby handles recording automatically to S3 bucket
-4. **Webhooks**: Whereby sends events for participant tracking
-
-### Database Structure
-```python
-# Room = Template/Configuration
-class Room:
-    id, name, user_id
-    recording_type, recording_trigger  # cloud, automatic-2nd-participant
-    webhook_url, webhook_secret
-
-# Meeting = Actual Whereby Meeting Instance
-class Meeting:
-    id  # Whereby meetingId
-    room_name  # Generated by Whereby
-    room_url, host_room_url  # Whereby URLs
-    num_clients  # Updated via webhooks
-```
-
-## Jitsi Components
-
-### Core Architecture
- **Jitsi Meet**: Web frontend (Next.js + React)
- **Prosody**: XMPP server for messaging/rooms
- **Jicofo**: Conference focus (orchestration)
- **JVB**: Videobridge (media routing)
- **Jibri**: Recording service
- **Jigasi**: SIP gateway (optional, for phone dial-in)
-
-### Exposure Requirements
- **Web service**: 443/80 (frontend)
- **JVB**: 10000/UDP (media streams) - **MUST EXPOSE**
- **Prosody**: 5280 (BOSH/WebSocket) - can proxy via web
- **Jicofo, Jibri, Jigasi**: Internal only
-
-## Recording with Jibri
-
-### How Jibri Works
- Each Jibri instance handles **one recording at a time**
- Records mixed audio/video to MP4 format
- Uses Chrome headless + ffmpeg for capture
- Supports finalize scripts for post-processing
-
-### Jibri Pool for Scaling
- Multiple Jibri instances join "jibribrewery" MUC
- Jicofo distributes recording requests to available instances
- Automatic load balancing and failover
-
-```yaml
-# Multiple Jibri instances
-jibri1:
-  environment:
-    - JIBRI_INSTANCE_ID=jibri1
-    - JIBRI_BREWERY_MUC=jibribrewery
-
-jibri2:
-  environment:
-    - JIBRI_INSTANCE_ID=jibri2
-    - JIBRI_BREWERY_MUC=jibribrewery
-```
-
-### Recording Automation Options
-1. **Environment Variables**: `ENABLE_RECORDING=1`, `AUTO_RECORDING=1`
-2. **URL Parameters**: `?config.autoRecord=true`
-3. **JWT Token**: Include recording permissions in JWT
-4. **API Control**: `api.executeCommand('startRecording')`
-
-### Post-Processing Integration
-```bash
-#!/bin/bash
-# finalize.sh - runs after recording completion
-RECORDING_FILE=$1
-MEETING_METADATA=$2
-ROOM_NAME=$3
-
-# Copy to Reflector-accessible location
-cp "$RECORDING_FILE" /shared/reflector-uploads/
-
-# Trigger Reflector processing
-curl -X POST "http://reflector-api:8000/v1/transcripts/process" \
-  -H "Content-Type: application/json" \
-  -d "{
-    \"file_path\": \"/shared/reflector-uploads/$(basename $RECORDING_FILE)\",
-    \"room_name\": \"$ROOM_NAME\",
-    \"source\": \"jitsi\"
-  }"
-```
-
-## React Integration
-
-### Official React SDK
-```bash
-npm i @jitsi/react-sdk
-```
-
-```jsx
-import { JitsiMeeting } from '@jitsi/react-sdk'
-
-<JitsiMeeting
-  room="meeting-room"
-  serverURL="https://your-jitsi.domain"
-  jwt="your-jwt-token"
-  config={{
-    startWithAudioMuted: true,
-    fileRecordingsEnabled: true,
-    autoRecord: true
-  }}
-  onParticipantJoined={(participant) => {
-    // Track participant events
-  }}
-  onRecordingStatusChanged={(status) => {
-    // Handle recording events
-  }}
-/>
-```
-
-## Authentication & Room Control
-
-### JWT-Based Access Control
-```python
-def generate_jitsi_jwt(payload):
-    return jwt.encode({
-        "aud": "jitsi",
-        "iss": "reflector",
-        "sub": "reflector-user",
-        "room": payload["room"],
-        "exp": int(payload["exp"].timestamp()),
-        "context": {
-            "user": {
-                "name": payload["user_name"],
-                "moderator": payload.get("moderator", False)
-            },
-            "features": {
-                "recording": payload.get("recording", True)
-            }
-        }
-    }, JITSI_JWT_SECRET)
-```
-
-### Prevent Anonymous Room Creation
-```bash
-# Environment configuration
-ENABLE_AUTH=1
-ENABLE_GUESTS=0
-AUTH_TYPE=jwt
-JWT_APP_ID=reflector
-JWT_APP_SECRET=your-secret-key
-```
-
-## Webhook Integration
-
-### Real-time Events via Prosody
-Custom event-sync module can send webhooks for:
- Participant join/leave
- Recording start/stop
- Room creation/destruction
- Mute/unmute events
-
-```lua
-- mod_event_sync.lua
-module:hook("muc-occupant-joined", function(event)
-    send_event({
-        type = "participant_joined",
-        room = event.room.jid,
-        participant = {
-            nick = event.occupant.nick,
-            jid = event.occupant.jid,
-        },
-        timestamp = os.time(),
-    });
-end);
-```
-
-### Jibri Recording Webhooks
-```bash
-# Environment variable
-JIBRI_WEBHOOK_SUBSCRIBERS=https://your-reflector.com/webhooks/jibri
-```
-
-## Proposed Reflector Integration
-
-### Modified Database Schema
-```python
-class Meeting(BaseModel):
-    id: str  # Our generated meeting ID
-    room_name: str  # Generated: reflector-{room.name}-{timestamp}
-    room_url: str  # https://jitsi.domain/room_name?jwt=token
-    host_room_url: str  # Same but with moderator JWT
-    # Add Jitsi-specific fields
-    jitsi_jwt: str  # JWT token
-    jitsi_room_id: str  # Internal room identifier
-    recording_status: str  # pending, recording, completed
-    recording_file_path: Optional[str]
-```
-
-### API Replacement
-```python
-# Replace whereby.py with jitsi.py
-async def create_meeting(room_name_prefix: str, end_date: datetime, room: Room):
-    # Generate unique room name
-    jitsi_room = f"reflector-{room.name}-{int(time.time())}"
-
-    # Generate JWT tokens
-    user_jwt = generate_jwt(room=jitsi_room, moderator=False, exp=end_date)
-    host_jwt = generate_jwt(room=jitsi_room, moderator=True, exp=end_date)
-
-    return {
-        "meetingId": generate_uuid4(),  # Our ID
-        "roomName": jitsi_room,
-        "roomUrl": f"https://jitsi.domain/{jitsi_room}?jwt={user_jwt}",
-        "hostRoomUrl": f"https://jitsi.domain/{jitsi_room}?jwt={host_jwt}",
-        "startDate": datetime.now().isoformat(),
-        "endDate": end_date.isoformat(),
-    }
-```
-
-### Webhook Endpoints
-```python
-# Replace whereby webhook with jitsi webhooks
-@router.post("/jitsi/events")
-async def jitsi_events_webhook(event_data: dict):
-    event_type = event_data.get("event")
-    room_name = event_data.get("room", "").split("@")[0]
-
-    meeting = await Meeting.get_by_room(room_name)
-
-    if event_type == "muc-occupant-joined":
-        # Update participant count
-        meeting.num_clients += 1
-
-    elif event_type == "jibri-recording-on":
-        meeting.recording_status = "recording"
-
-    elif event_type == "jibri-recording-off":
-        meeting.recording_status = "processing"
-        await process_meeting_recording.delay(meeting.id)
-
-@router.post("/jibri/recording-complete")
-async def recording_complete(data: dict):
-    # Handle finalize script webhook
-    room_name = data.get("room_name")
-    file_path = data.get("file_path")
-
-    meeting = await Meeting.get_by_room(room_name)
-    meeting.recording_file_path = file_path
-    meeting.recording_status = "completed"
-
-    # Start Reflector processing
-    await process_recording_for_transcription(meeting.id, file_path)
-```
-
-## Deployment with Docker
-
-### Official docker-jitsi-meet
-```bash
-# Download official release
-wget $(wget -q -O - https://api.github.com/repos/jitsi/docker-jitsi-meet/releases/latest | grep zip | cut -d\" -f4)
-
-# Setup
-mkdir -p ~/.jitsi-meet-cfg/{web,transcripts,prosody/config,prosody/prosody-plugins-custom,jicofo,jvb,jigasi,jibri}
-./gen-passwords.sh  # Generate secure passwords
-docker compose up -d
-```
-
-### Coolify Integration
-```yaml
-services:
-  web:
-    ports: ["80:80", "443:443"]
-  jvb:
-    ports: ["10000:10000/udp"]  # Must expose for media
-  jibri1:
-    environment:
-      - JIBRI_INSTANCE_ID=jibri1
-      - JIBRI_FINALIZE_RECORDING_SCRIPT_PATH=/config/finalize.sh
-  jibri2:
-    environment:
-      - JIBRI_INSTANCE_ID=jibri2
-```
-
-## Benefits vs Whereby
-
-### Cost & Control
-✅ **No per-minute charges** - significant cost savings
-✅ **Full recording control** - direct file access
-✅ **Custom branding** - complete UI control
-✅ **Self-hosted** - no vendor lock-in
-
-### Technical Advantages
-✅ **Real-time events** - immediate webhook notifications
-✅ **Rich participant metadata** - detailed tracking
-✅ **JWT security** - token-based access with expiration
-✅ **Multiple recording formats** - audio-only options
-✅ **Scalable architecture** - horizontal Jibri scaling
-
-### Integration Benefits
-✅ **Same API surface** - minimal changes to existing code
-✅ **React SDK** - better frontend integration
-✅ **Direct processing** - no S3 download delays
-✅ **Event-driven architecture** - better real-time capabilities
-
-## Implementation Plan
-
-1. **Deploy Jitsi Stack** - Set up docker-jitsi-meet with multiple Jibri instances
-2. **Create jitsi.py** - Replace whereby.py with Jitsi API functions
-3. **Update Database** - Add Jitsi-specific fields to Meeting model
-4. **Webhook Integration** - Replace Whereby webhooks with Jitsi events
-5. **Frontend Updates** - Replace Whereby embed with Jitsi React SDK
-6. **Testing & Migration** - Gradual rollout with fallback to Whereby
-
-## Recording Limitations & Considerations
-
-### Current Limitations
- **Mixed audio only** - Jibri doesn't separate participant tracks natively
- **One recording per Jibri** - requires multiple instances for concurrent recordings
- **Chrome dependency** - Jibri uses headless Chrome for recording
-
-### Metadata Capabilities
-✅ **Participant join/leave timestamps** - via webhooks
-✅ **Speaking time tracking** - via audio level events
-✅ **Meeting duration** - precise timing
-✅ **Room-specific data** - custom metadata in JWT
-
-### Alternative Recording Methods
- **Local recording** - browser-based, per-participant
- **Custom recording** - lib-jitsi-meet for individual streams
- **Third-party solutions** - Recall.ai, Otter.ai integrations
-
-## Security Considerations
-
-### JWT Configuration
- **Room-specific tokens** - limit access to specific rooms
- **Time-based expiration** - automatic cleanup
- **Feature permissions** - control recording, moderation rights
- **User identification** - embed user metadata in tokens
-
-### Access Control
- **No anonymous rooms** - all rooms require valid JWT
- **API-only creation** - prevent direct room access
- **Webhook verification** - HMAC signature validation
-
-## Next Steps
-
-1. **Deploy test Jitsi instance** - validate recording pipeline
-2. **Prototype jitsi.py** - create equivalent API functions
-3. **Test webhook integration** - ensure event delivery works
-4. **Performance testing** - validate multiple concurrent recordings
-5. **Migration strategy** - plan gradual transition from Whereby
-
---
-
-*This document serves as the comprehensive planning and research notes for Jitsi integration in Reflector. It should be updated as implementation progresses and new insights are discovered.*
--- a/docs/video-jitsi.md
+++ b/docs/video-jitsi.md
@@ -1,720 +0,0 @@
-# Jitsi Meet Integration Configuration Guide
-
-This guide explains how to configure Reflector to use your self-hosted Jitsi Meet installation for video meetings, recording, and participant tracking.
-
-## Overview
-
-Jitsi Meet is an open-source video conferencing platform that can be self-hosted. Reflector integrates with Jitsi Meet to:
-
- Create secure meeting rooms with JWT authentication
- Track participant join/leave events via Prosody webhooks
- Record meetings using Jibri recording service
- Process recordings for transcription and analysis
-
-## Requirements
-
-### Self-Hosted Jitsi Meet
-
-You need a complete Jitsi Meet installation including:
-
-1. **Jitsi Meet Web Interface** - The main meeting interface
-2. **Prosody XMPP Server** - Handles room management and authentication
-3. **Jicofo (JItsi COnference FOcus)** - Manages media sessions
-4. **Jitsi Videobridge (JVB)** - Handles WebRTC media routing
-5. **Jibri Recording Service** - Records meetings (optional but recommended)
-
-### System Requirements
-
- **Domain with SSL Certificate** - Required for WebRTC functionality
- **Prosody mod_event_sync** - For webhook event handling
- **JWT Authentication** - For secure room access control
- **Storage Solution** - For recording files (local or cloud)
-
-## Configuration Variables
-
-Add the following environment variables to your Reflector `.env` file:
-
-### Required Variables
-
-```bash
-# Jitsi Meet Domain (without https://)
-JITSI_DOMAIN=meet.example.com
-
-# JWT Secret for room authentication (generate with: openssl rand -hex 32)
-JITSI_JWT_SECRET=your-64-character-hex-secret-here
-
-# Webhook secret for event handling (generate with: openssl rand -hex 16)
-JITSI_WEBHOOK_SECRET=your-32-character-hex-secret-here
-```
-
-### Optional Variables
-
-```bash
-# Application identifier (should match Jitsi configuration)
-JITSI_APP_ID=reflector
-
-# JWT issuer and audience (should match Jitsi configuration)
-JITSI_JWT_ISSUER=reflector
-JITSI_JWT_AUDIENCE=jitsi
-```
-
-## Installation Steps
-
-### 1. Jitsi Meet Server Installation
-
-#### Quick Installation (Ubuntu/Debian)
-
-```bash
-# Add Jitsi repository
-curl -fsSL https://download.jitsi.org/jitsi-key.gpg.key | sudo gpg --dearmor -o /usr/share/keyrings/jitsi-keyring.gpg
-echo "deb [signed-by=/usr/share/keyrings/jitsi-keyring.gpg] https://download.jitsi.org stable/" | sudo tee /etc/apt/sources.list.d/jitsi-stable.list
-
-# Install Jitsi Meet
-sudo apt update
-sudo apt install jitsi-meet
-
-# Configure SSL certificate
-sudo /usr/share/jitsi-meet/scripts/install-letsencrypt-cert.sh
-```
-
-#### Docker Installation
-
-```bash
-# Clone Jitsi Docker repository
-git clone https://github.com/jitsi/docker-jitsi-meet
-cd docker-jitsi-meet
-
-# Copy environment template
-cp env.example .env
-
-# Edit configuration
-nano .env
-
-# Start services
-docker-compose up -d
-```
-
-### 2. JWT Authentication Setup
-
-#### Update Prosody Configuration
-
-Edit `/etc/prosody/conf.d/your-domain.cfg.lua`:
-
-```lua
-VirtualHost "meet.example.com"
-    authentication = "token"
-    app_id = "reflector"
-    app_secret = "your-jwt-secret-here"
-
-    -- Allow anonymous access for guests
-    c2s_require_encryption = false
-    admins = { "focusUser@auth.meet.example.com" }
-
-    modules_enabled = {
-        "bosh";
-        "pubsub";
-        "ping";
-        "roster";
-        "saslauth";
-        "tls";
-        "dialback";
-        "disco";
-        "carbons";
-        "pep";
-        "private";
-        "blocklist";
-        "vcard";
-        "version";
-        "uptime";
-        "time";
-        "ping";
-        "register";
-        "admin_adhoc";
-        "token_verification";
-        "event_sync";  -- Required for webhooks
-    }
-```
-
-#### Configure Jitsi Meet Interface
-
-Edit `/etc/jitsi/meet/your-domain-config.js`:
-
-```javascript
-var config = {
-    hosts: {
-        domain: 'meet.example.com',
-        muc: 'conference.meet.example.com'
-    },
-
-    // Enable JWT authentication
-    enableUserRolesBasedOnToken: true,
-
-    // Recording configuration
-    fileRecordingsEnabled: true,
-    liveStreamingEnabled: false,
-
-    // Reflector integration settings
-    prejoinPageEnabled: true,
-    requireDisplayName: true
-};
-```
-
-### 3. Webhook Event Configuration
-
-#### Install Event Sync Module
-
-```bash
-# Download the module
-cd /usr/share/jitsi-meet/prosody-plugins/
-wget https://raw.githubusercontent.com/jitsi-contrib/prosody-plugins/main/mod_event_sync.lua
-```
-
-#### Configure Event Sync
-
-Add to your Prosody configuration:
-
-```lua
-Component "conference.meet.example.com" "muc"
-    storage = "memory"
-    modules_enabled = {
-        "muc_meeting_id";
-        "muc_domain_mapper";
-        "polls";
-        "event_sync";  -- Enable event sync
-    }
-
-    -- Event sync webhook configuration
-    event_sync_url = "https://your-reflector-domain.com/v1/jitsi/events"
-    event_sync_secret = "your-webhook-secret-here"
-
-    -- Events to track
-    event_sync_events = {
-        "muc-occupant-joined",
-        "muc-occupant-left",
-        "jibri-recording-on",
-        "jibri-recording-off"
-    }
-
-#### Webhook Event Payload Examples
-
-**Participant Joined Event:**
-```json
-{
-  "event": "muc-occupant-joined",
-  "room": "reflector-my-room-uuid123",
-  "timestamp": "2025-01-15T10:30:00.000Z",
-  "data": {
-    "occupant_id": "participant-456",
-    "nick": "John Doe",
-    "role": "participant",
-    "affiliation": "none"
-  }
-}
-```
-
-**Recording Started Event:**
-```json
-{
-  "event": "jibri-recording-on",
-  "room": "reflector-my-room-uuid123",
-  "timestamp": "2025-01-15T10:32:00.000Z",
-  "data": {
-    "recording_id": "rec-789",
-    "initiator": "moderator-123"
-  }
-}
-```
-
-**Recording Completed Event:**
-```json
-{
-  "room_name": "reflector-my-room-uuid123",
-  "recording_file": "/var/recordings/rec-789.mp4",
-  "recording_status": "completed",
-  "timestamp": "2025-01-15T11:15:00.000Z"
-}
-```
-
-### 4. Jibri Recording Setup (Optional)
-
-#### Install Jibri
-
-```bash
-# Install Jibri package
-sudo apt install jibri
-
-# Create recording directory
-sudo mkdir -p /var/recordings
-sudo chown jibri:jibri /var/recordings
-```
-
-#### Configure Jibri
-
-Edit `/etc/jitsi/jibri/jibri.conf`:
-
-```hocon
-jibri {
-    recording {
-        recordings-directory = "/var/recordings"
-        finalize-script = "/opt/jitsi/jibri/finalize.sh"
-    }
-
-    api {
-        xmpp {
-            environments = [{
-                name = "prod environment"
-                xmpp-server-hosts = ["meet.example.com"]
-                xmpp-domain = "meet.example.com"
-
-                control-muc {
-                    domain = "internal.auth.meet.example.com"
-                    room-name = "JibriBrewery"
-                    nickname = "jibri-nickname"
-                }
-
-                control-login {
-                    domain = "auth.meet.example.com"
-                    username = "jibri"
-                    password = "jibri-password"
-                }
-            }]
-        }
-    }
-}
-```
-
-#### Create Finalize Script
-
-Create `/opt/jitsi/jibri/finalize.sh`:
-
-```bash
-#!/bin/bash
-# Jibri finalize script for Reflector integration
-
-RECORDING_FILE="$1"
-ROOM_NAME="$2"
-REFLECTOR_API_URL="${REFLECTOR_API_URL:-http://localhost:1250}"
-
-# Prepare webhook payload
-TIMESTAMP=$(date -u +%Y-%m-%dT%H:%M:%S.%3NZ)
-PAYLOAD=$(cat <<EOF
-{
-    "room_name": "$ROOM_NAME",
-    "recording_file": "$RECORDING_FILE",
-    "recording_status": "completed",
-    "timestamp": "$TIMESTAMP"
-}
-EOF
-)
-
-# Generate signature
-SIGNATURE=$(echo -n "$PAYLOAD" | openssl dgst -sha256 -hmac "$JITSI_WEBHOOK_SECRET" | cut -d' ' -f2)
-
-# Send webhook to Reflector
-curl -X POST "$REFLECTOR_API_URL/v1/jibri/recording-complete" \
-    -H "Content-Type: application/json" \
-    -H "X-Jitsi-Signature: $SIGNATURE" \
-    -d "$PAYLOAD"
-
-echo "Recording finalization webhook sent for room: $ROOM_NAME"
-```
-
-Make executable:
-```bash
-sudo chmod +x /opt/jitsi/jibri/finalize.sh
-```
-
-### 5. Restart Services
-
-After configuration changes:
-
-```bash
-sudo systemctl restart prosody
-sudo systemctl restart jicofo
-sudo systemctl restart jitsi-videobridge2
-sudo systemctl restart jibri
-sudo systemctl restart nginx
-```
-
-## Room Configuration
-
-### Creating Jitsi Rooms
-
-Create rooms with Jitsi platform in Reflector:
-
-```bash
-curl -X POST "https://your-reflector-domain.com/v1/rooms" \
-  -H "Content-Type: application/json" \
-  -H "Authorization: Bearer $AUTH_TOKEN" \
-  -d '{
-    "name": "my-jitsi-room",
-    "platform": "jitsi",
-    "recording_type": "cloud",
-    "recording_trigger": "automatic-2nd-participant",
-    "is_locked": false,
-    "room_mode": "normal"
-  }'
-```
-
-### Meeting Creation
-
-Meetings automatically use JWT authentication:
-
-```bash
-curl -X POST "https://your-reflector-domain.com/v1/rooms/my-jitsi-room/meeting" \
-  -H "Authorization: Bearer $AUTH_TOKEN"
-```
-
-Response includes JWT-authenticated URLs:
-```json
-{
-  "id": "meeting-uuid",
-  "room_name": "reflector-my-jitsi-room-123456",
-  "room_url": "https://meet.example.com/room?jwt=user-token",
-  "host_room_url": "https://meet.example.com/room?jwt=moderator-token"
-}
-```
-
-## Features and Capabilities
-
-### JWT Authentication
-
-Reflector automatically generates JWT tokens with:
- **Room Access Control** - Secure room entry
- **User Roles** - Moderator vs participant permissions
- **Expiration** - Configurable token lifetime (default 8 hours)
- **Custom Claims** - Room-specific metadata
-
-### Recording Options
-
-**Recording Types:**
- `"none"` - No recording
- `"local"` - Local Jibri recording
- `"cloud"` - Cloud recording (requires external storage)
-
-**Recording Triggers:**
- `"none"` - Manual recording only
- `"prompt"` - Prompt users to start
- `"automatic"` - Start immediately
- `"automatic-2nd-participant"` - Start when 2nd person joins
-
-### Event Tracking and Storage
-
-Reflector automatically stores all webhook events in the `meetings` table for comprehensive meeting analytics:
-
-**Supported Event Types:**
- `muc-occupant-joined` - Participant joined the meeting
- `muc-occupant-left` - Participant left the meeting
- `jibri-recording-on` - Recording started
- `jibri-recording-off` - Recording stopped
- `recording_completed` - Recording file ready for processing
-
-**Event Storage Structure:**
-Each webhook event is stored as a JSON object in the `meetings.events` column:
-```json
-{
-  "type": "muc-occupant-joined",
-  "timestamp": "2025-01-15T10:30:00.123456Z",
-  "data": {
-    "timestamp": "2025-01-15T10:30:00Z",
-    "user_id": "participant-123",
-    "display_name": "John Doe"
-  }
-}
-```
-
-**Querying Stored Events:**
-```sql
-- Get all events for a meeting
-SELECT events FROM meeting WHERE id = 'meeting-uuid';
-
-- Count participant joins
-SELECT json_array_length(
-  json_extract(events, '$[*] ? (@.type == "muc-occupant-joined")')
-) as total_joins FROM meeting WHERE id = 'meeting-uuid';
-```
-
-## Testing and Verification
-
-### Health Check
-
-Test Jitsi webhook integration:
-
-```bash
-curl "https://your-reflector-domain.com/v1/jitsi/health"
-```
-
-Expected response:
-```json
-{
-    "status": "ok",
-    "service": "jitsi-webhooks",
-    "timestamp": "2025-01-15T10:30:00.000Z",
-    "webhook_secret_configured": true
-}
-```
-
-### JWT Token Testing
-
-Verify JWT generation works:
-```bash
-# Create a test meeting
-MEETING=$(curl -X POST "https://your-reflector-domain.com/v1/rooms/test-room/meeting" \
-  -H "Authorization: Bearer $AUTH_TOKEN" | jq -r '.room_url')
-
-echo "Test meeting URL: $MEETING"
-```
-
-### Webhook Testing
-
-#### Manual Webhook Event Testing
-
-Test participant join event:
-```bash
-# Generate proper signature
-PAYLOAD='{"event":"muc-occupant-joined","room":"reflector-test-room-uuid","timestamp":"2025-01-15T10:30:00.000Z","data":{"user_id":"test-user","display_name":"Test User"}}'
-SIGNATURE=$(echo -n "$PAYLOAD" | openssl dgst -sha256 -hmac "$JITSI_WEBHOOK_SECRET" | cut -d' ' -f2)
-
-curl -X POST "https://your-reflector-domain.com/v1/jitsi/events" \
-  -H "Content-Type: application/json" \
-  -H "X-Jitsi-Signature: $SIGNATURE" \
-  -d "$PAYLOAD"
-```
-
-Expected response:
-```json
-{
-  "status": "ok",
-  "event": "muc-occupant-joined",
-  "room": "reflector-test-room-uuid"
-}
-```
-
-#### Recording Webhook Testing
-
-Test recording completion event:
-```bash
-PAYLOAD='{"room_name":"reflector-test-room-uuid","recording_file":"/recordings/test.mp4","recording_status":"completed","timestamp":"2025-01-15T10:30:00.000Z"}'
-SIGNATURE=$(echo -n "$PAYLOAD" | openssl dgst -sha256 -hmac "$JITSI_WEBHOOK_SECRET" | cut -d' ' -f2)
-
-curl -X POST "https://your-reflector-domain.com/v1/jibri/recording-complete" \
-  -H "Content-Type: application/json" \
-  -H "X-Jitsi-Signature: $SIGNATURE" \
-  -d "$PAYLOAD"
-```
-
-#### Event Storage Verification
-
-Verify events were stored:
-```bash
-# Check meeting events via API (requires authentication)
-curl -H "Authorization: Bearer $AUTH_TOKEN" \
-  "https://your-reflector-domain.com/v1/meetings/{meeting-id}"
-```
-
-## Troubleshooting
-
-### Common Issues
-
-#### JWT Authentication Failures
-
-**Symptoms**: Users cannot join rooms, "Authentication failed" errors
-
-**Solutions**:
-1. Verify `JITSI_JWT_SECRET` matches Prosody configuration
-2. Check JWT token hasn't expired (default 8 hours)
-3. Ensure system clocks are synchronized between servers
-4. Validate JWT issuer/audience configuration matches
-
-**Debug JWT tokens**:
-```bash
-# Decode JWT payload
-echo "JWT_TOKEN_HERE" | cut -d'.' -f2 | base64 -d | jq
-```
-
-#### Webhook Events Not Received
-
-**Symptoms**: Participant counts not updating, no recording events
-
-**Solutions**:
-1. Verify `mod_event_sync` is loaded in Prosody
-2. Check webhook URL is accessible from Jitsi server
-3. Validate webhook signature generation
-4. Review Prosody and Reflector logs
-
-**Debug webhook connectivity**:
-```bash
-# Test from Jitsi server
-curl -v "https://your-reflector-domain.com/v1/jitsi/health"
-
-# Check Prosody logs
-sudo tail -f /var/log/prosody/prosody.log
-```
-
-#### Webhook Signature Verification Issues
-
-**Symptoms**: HTTP 401 "Invalid webhook signature" errors
-
-**Solutions**:
-1. Verify webhook secret matches between Jitsi and Reflector
-2. Check payload encoding (no extra whitespace)
-3. Ensure proper HMAC-SHA256 signature generation
-
-**Debug signature generation**:
-```bash
-# Test signature manually
-PAYLOAD='{"event":"test","room":"test","timestamp":"2025-01-15T10:30:00.000Z","data":{}}'
-SECRET="your-webhook-secret-here"
-
-# Generate signature (should match X-Jitsi-Signature header)
-echo -n "$PAYLOAD" | openssl dgst -sha256 -hmac "$SECRET" | cut -d' ' -f2
-
-# Test with curl
-curl -X POST "https://your-reflector-domain.com/v1/jitsi/events" \
-  -H "Content-Type: application/json" \
-  -H "X-Jitsi-Signature: $(echo -n "$PAYLOAD" | openssl dgst -sha256 -hmac "$SECRET" | cut -d' ' -f2)" \
-  -d "$PAYLOAD" -v
-```
-
-#### Event Storage Problems
-
-**Symptoms**: Events received but not stored in database
-
-**Solutions**:
-1. Check database connectivity and permissions
-2. Verify meeting exists before event processing
-3. Review Reflector application logs
-4. Ensure JSON column support in database
-
-**Debug event storage**:
-```bash
-# Check meeting exists
-curl -H "Authorization: Bearer $TOKEN" \
-  "https://your-reflector-domain.com/v1/meetings/{meeting-id}"
-
-# Monitor database queries (if using PostgreSQL)
-sudo -u postgres psql -c "SELECT * FROM pg_stat_activity WHERE query LIKE '%meeting%';"
-
-# Check Reflector logs for event processing
-sudo journalctl -u reflector -f | grep -E "(event|webhook|jitsi)"
-```
-
-#### Recording Issues
-
-**Symptoms**: Recordings not starting, finalize script errors
-
-**Solutions**:
-1. Verify Jibri service status: `sudo systemctl status jibri`
-2. Check recording directory permissions: `/var/recordings`
-3. Validate finalize script execution permissions
-4. Monitor Jibri logs: `sudo journalctl -u jibri -f`
-
-**Test finalize script**:
-```bash
-sudo -u jibri /opt/jitsi/jibri/finalize.sh "/test/recording.mp4" "test-room"
-```
-
-#### Meeting Creation Failures
-
-**Symptoms**: HTTP 500 errors when creating meetings
-
-**Solutions**:
-1. Check Reflector logs for JWT generation errors
-2. Verify all required environment variables are set
-3. Ensure Jitsi domain is accessible from Reflector
-4. Test JWT secret configuration
-
-### Debug Commands
-
-```bash
-# Verify Prosody configuration
-sudo prosodyctl check config
-
-# Check Jitsi services status
-sudo systemctl status prosody jicofo jitsi-videobridge2
-
-# Test JWT generation
-curl -X POST "https://your-reflector-domain.com/v1/rooms/test/meeting" \
-  -H "Authorization: Bearer $TOKEN" -v
-
-# Monitor webhook events
-sudo tail -f /var/log/reflector/app.log | grep jitsi
-
-# Check SSL certificates
-sudo certbot certificates
-```
-
-### Performance Optimization
-
-#### Scaling Considerations
-
-**Single Server Limits:**
- ~50 concurrent participants per JVB instance
- ~10 concurrent Jibri recordings
- CPU and bandwidth become bottlenecks
-
-**Multi-Server Setup:**
- Multiple JVB instances for scaling
- Dedicated Jibri recording servers
- Load balancing for high availability
-
-#### Resource Monitoring
-
-```bash
-# Monitor JVB performance
-sudo systemctl status jitsi-videobridge2
-sudo journalctl -u jitsi-videobridge2 -f
-
-# Check Prosody connections
-sudo prosodyctl mod_admin_telnet
-> c2s:show()
-> muc:rooms()
-```
-
-## Security Best Practices
-
-### JWT Security
- Use strong, unique secrets (32+ characters)
- Rotate JWT secrets regularly
- Implement proper token expiration
- Never log or expose JWT tokens
-
-### Network Security
- Use HTTPS/WSS for all communications
- Implement proper firewall rules
- Consider VPN for server-to-server communication
- Monitor for unauthorized access attempts
-
-### Recording Security
- Encrypt recordings at rest
- Implement access controls for recording files
- Regular security audits of file permissions
- Comply with data protection regulations
-
-## Migration from Whereby
-
-If migrating from Whereby to Jitsi:
-
-1. **Parallel Setup** - Configure Jitsi alongside existing Whereby
-2. **Room Migration** - Update room platform field to "jitsi"
-3. **Test Integration** - Verify meeting creation and webhooks
-4. **User Training** - Different UI and feature set
-5. **Monitor Performance** - Watch for issues during transition
-6. **Cleanup** - Remove Whereby configuration when stable
-
-## Support and Resources
-
-### Jitsi Community Resources
- **Documentation**: [jitsi.github.io/handbook](https://jitsi.github.io/handbook/)
- **Community Forum**: [community.jitsi.org](https://community.jitsi.org/)
- **GitHub Issues**: [github.com/jitsi/jitsi-meet](https://github.com/jitsi/jitsi-meet)
-
-### Professional Support
- **8x8 Commercial Support** - Professional Jitsi hosting and support
- **Community Consulting** - Third-party Jitsi implementation services
-
-### Monitoring and Maintenance
- Monitor system resources (CPU, memory, bandwidth)
- Regular security updates for all components
- Backup configuration files and certificates
- Test disaster recovery procedures
--- a/docs/video-whereby.md
+++ b/docs/video-whereby.md
@@ -1,276 +0,0 @@
-# Whereby Integration Configuration Guide
-
-This guide explains how to configure Reflector to use Whereby as your video meeting platform for room creation, recording, and participant tracking.
-
-## Overview
-
-Whereby is a browser-based video meeting platform that provides hosted meeting rooms with recording capabilities. Reflector integrates with Whereby's API to:
-
- Create secure meeting rooms with custom branding
- Handle participant join/leave events via webhooks
- Automatically record meetings to AWS S3 storage
- Track meeting sessions and participant counts
-
-## Requirements
-
-### Whereby Account Setup
-
-1. **Whereby Account**: Sign up for a Whereby business account at [whereby.com](https://whereby.com/business)
-2. **API Access**: Request API access from Whereby support (required for programmatic room creation)
-3. **Webhook Configuration**: Configure webhooks in your Whereby dashboard to point to your Reflector instance
-
-### AWS S3 Storage
-
-Whereby requires AWS S3 for recording storage. You need:
- AWS account with S3 access
- Dedicated S3 bucket for Whereby recordings
- AWS IAM credentials with S3 write permissions
-
-## Configuration Variables
-
-Add the following environment variables to your Reflector `.env` file:
-
-### Required Variables
-
-```bash
-# Whereby API Configuration
-WHEREBY_API_KEY=your-whereby-jwt-api-key
-WHEREBY_WEBHOOK_SECRET=your-webhook-secret-from-whereby
-
-# AWS S3 Storage for Recordings
-AWS_WHEREBY_ACCESS_KEY_ID=your-aws-access-key
-AWS_WHEREBY_ACCESS_KEY_SECRET=your-aws-secret-key
-RECORDING_STORAGE_AWS_BUCKET_NAME=your-s3-bucket-name
-```
-
-### Optional Variables
-
-```bash
-# Whereby API URL (defaults to production)
-WHEREBY_API_URL=https://api.whereby.dev/v1
-
-# SQS Configuration (for recording processing)
-AWS_PROCESS_RECORDING_QUEUE_URL=https://sqs.region.amazonaws.com/account/queue
-SQS_POLLING_TIMEOUT_SECONDS=60
-```
-
-## Configuration Steps
-
-### 1. Whereby API Key Setup
-
-1. **Contact Whereby Support** to request API access for your account
-2. **Generate JWT Token** in your Whereby dashboard under API settings
-3. **Copy the JWT token** and set it as `WHEREBY_API_KEY` in your environment
-
-The API key is a JWT token that looks like:
-```
-eyJ[...truncated JWT token...]
-```
-
-### 2. Webhook Configuration
-
-1. **Access Whereby Dashboard** and navigate to webhook settings
-2. **Set Webhook URL** to your Reflector instance:
-   ```
-   https://your-reflector-domain.com/v1/whereby
-   ```
-3. **Configure Events** to send the following event types:
-   - `room.client.joined` - When participants join
-   - `room.client.left` - When participants leave
-4. **Generate Webhook Secret** and set it as `WHEREBY_WEBHOOK_SECRET`
-5. **Save Configuration** in your Whereby dashboard
-
-### 3. AWS S3 Storage Setup
-
-1. **Create S3 Bucket** dedicated for Whereby recordings
-2. **Create IAM User** with programmatic access
-3. **Attach S3 Policy** with the following permissions:
-   ```json
-   {
-     "Version": "2012-10-17",
-     "Statement": [
-       {
-         "Effect": "Allow",
-         "Action": [
-           "s3:PutObject",
-           "s3:PutObjectAcl",
-           "s3:GetObject"
-         ],
-         "Resource": "arn:aws:s3:::your-bucket-name/*"
-       }
-     ]
-   }
-   ```
-4. **Configure Environment Variables** with the IAM credentials
-
-### 4. Room Configuration
-
-When creating rooms in Reflector, set the platform to use Whereby:
-
-```bash
-curl -X POST "https://your-reflector-domain.com/v1/rooms" \
-  -H "Content-Type: application/json" \
-  -H "Authorization: Bearer $AUTH_TOKEN" \
-  -d '{
-    "name": "my-whereby-room",
-    "platform": "whereby",
-    "recording_type": "cloud",
-    "recording_trigger": "automatic-2nd-participant",
-    "is_locked": false,
-    "room_mode": "normal"
-  }'
-```
-
-## Meeting Features
-
-### Recording Options
-
-Whereby supports three recording types:
- **`none`**: No recording
- **`local`**: Local recording (not recommended for production)
- **`cloud`**: Cloud recording to S3 (recommended)
-
-### Recording Triggers
-
-Control when recordings start:
- **`none`**: No automatic recording
- **`prompt`**: Prompt users to start recording
- **`automatic`**: Start immediately when meeting begins
- **`automatic-2nd-participant`**: Start when second participant joins
-
-### Room Modes
-
- **`normal`**: Standard meeting room
- **`group`**: Group meeting with advanced features
-
-## Webhook Event Handling
-
-Reflector automatically handles these Whereby webhook events:
-
-### Participant Tracking
-```json
-{
-  "type": "room.client.joined",
-  "data": {
-    "meetingId": "room-uuid",
-    "numClients": 2
-  }
-}
-```
-
-### Recording Events
-Whereby sends recording completion events that trigger Reflector's processing pipeline:
- Audio transcription
- Speaker diarization
- Summary generation
-
-## Troubleshooting
-
-### Common Issues
-
-#### API Authentication Errors
-**Symptoms**: 401 Unauthorized errors when creating meetings
-
-**Solutions**:
-1. Verify your `WHEREBY_API_KEY` is correct and not expired
-2. Ensure you have API access enabled on your Whereby account
-3. Contact Whereby support if API access is not available
-
-#### Webhook Signature Validation Failed
-**Symptoms**: Webhook events rejected with 401 errors
-
-**Solutions**:
-1. Verify `WHEREBY_WEBHOOK_SECRET` matches your Whereby dashboard configuration
-2. Check webhook URL is correctly configured in Whereby dashboard
-3. Ensure webhook endpoint is accessible from Whereby servers
-
-#### Recording Upload Failures
-**Symptoms**: Recordings not appearing in S3 bucket
-
-**Solutions**:
-1. Verify AWS credentials have S3 write permissions
-2. Check S3 bucket name is correct and accessible
-3. Ensure AWS region settings match your bucket location
-4. Review AWS CloudTrail logs for permission issues
-
-#### Participant Count Not Updating
-**Symptoms**: Meeting participant counts remain at 0
-
-**Solutions**:
-1. Verify webhook events are being received at `/v1/whereby`
-2. Check webhook signature validation is passing
-3. Ensure meeting IDs match between Whereby and Reflector database
-
-### Debug Commands
-
-```bash
-# Test Whereby API connectivity
-curl -H "Authorization: Bearer $WHEREBY_API_KEY" \
-  https://api.whereby.dev/v1/meetings
-
-# Check webhook endpoint health
-curl https://your-reflector-domain.com/v1/whereby/health
-
-# Verify S3 bucket access
-aws s3 ls s3://your-bucket-name --profile whereby-user
-```
-
-## Security Considerations
-
-### API Key Security
- Store API keys securely using environment variables
- Rotate API keys regularly
- Never commit API keys to version control
- Use separate keys for development and production
-
-### Webhook Security
- Always validate webhook signatures using HMAC-SHA256
- Use HTTPS for all webhook endpoints
- Implement rate limiting on webhook endpoints
- Monitor webhook events for suspicious activity
-
-### Recording Privacy
- Ensure S3 bucket access is restricted to authorized users
- Consider encryption at rest for sensitive recordings
- Implement retention policies for recorded content
- Comply with data protection regulations (GDPR, etc.)
-
-## Performance Optimization
-
-### Meeting Scaling
- Monitor concurrent meeting limits on your Whereby plan
- Implement meeting cleanup for expired sessions
- Use appropriate room modes for different use cases
-
-### Recording Processing
- Configure SQS for asynchronous recording processing
- Monitor S3 storage usage and costs
- Implement automatic cleanup of processed recordings
-
-### Webhook Reliability
- Implement webhook retry mechanisms
- Monitor webhook delivery success rates
- Log webhook events for debugging and auditing
-
-## Migration from Other Platforms
-
-If migrating from another video platform:
-
-1. **Update Room Configuration**: Change existing rooms to use `"platform": "whereby"`
-2. **Configure Webhooks**: Set up Whereby webhook endpoints
-3. **Test Integration**: Verify meeting creation and event handling
-4. **Monitor Performance**: Watch for any issues during transition
-5. **Update Documentation**: Inform users of any workflow changes
-
-## Support
-
-For Whereby-specific issues:
- **Whereby Support**: [whereby.com/support](https://whereby.com/support)
- **API Documentation**: [whereby.dev](https://whereby.dev)
- **Status Page**: [status.whereby.com](https://status.whereby.com)
-
-For Reflector integration issues:
- Check application logs for error details
- Verify environment variable configuration
- Test webhook connectivity and authentication
- Review AWS permissions and S3 access
--- a/docs/video_platforms.md
+++ b/docs/video_platforms.md
@@ -1,474 +0,0 @@
-# Video Platforms Architecture (PR #529 Analysis)
-
-This document analyzes the video platforms refactoring implemented in PR #529 for daily.co integration, providing a blueprint for extending support to Jitsi and other video conferencing platforms.
-
-## Overview
-
-The video platforms refactoring introduces a clean abstraction layer that allows Reflector to support multiple video conferencing providers (Whereby, Daily.co, etc.) without changing core application logic. This architecture enables:
-
- Seamless switching between video platforms
- Platform-specific feature support
- Isolated platform code organization
- Consistent API surface across platforms
- Feature flags for gradual migration
-
-## Architecture Components
-
-### 1. **Directory Structure**
-
-```
-server/reflector/video_platforms/
-├── __init__.py              # Public API exports
-├── base.py                  # Abstract base classes
-├── factory.py               # Platform client factory
-├── registry.py              # Platform registration system
-├── whereby.py               # Whereby implementation
-├── daily.py                 # Daily.co implementation
-└── mock.py                  # Testing implementation
-```
-
-### 2. **Core Abstract Classes**
-
-#### `VideoPlatformClient` (base.py)
-Abstract base class defining the interface all platforms must implement:
-
-```python
-class VideoPlatformClient(ABC):
-    PLATFORM_NAME: str = ""
-
-    @abstractmethod
-    async def create_meeting(self, room_name_prefix: str, end_date: datetime, room: Room) -> MeetingData
-
-    @abstractmethod
-    async def get_room_sessions(self, room_name: str) -> Dict[str, Any]
-
-    @abstractmethod
-    async def delete_room(self, room_name: str) -> bool
-
-    @abstractmethod
-    async def upload_logo(self, room_name: str, logo_path: str) -> bool
-
-    @abstractmethod
-    def verify_webhook_signature(self, body: bytes, signature: str, timestamp: Optional[str] = None) -> bool
-```
-
-#### `MeetingData` (base.py)
-Standardized meeting data structure returned by all platforms:
-
-```python
-class MeetingData(BaseModel):
-    meeting_id: str
-    room_name: str
-    room_url: str
-    host_room_url: str
-    platform: str
-    extra_data: Dict[str, Any] = {}  # Platform-specific data
-```
-
-#### `VideoPlatformConfig` (base.py)
-Unified configuration structure for all platforms:
-
-```python
-class VideoPlatformConfig(BaseModel):
-    api_key: str
-    webhook_secret: str
-    api_url: Optional[str] = None
-    subdomain: Optional[str] = None
-    s3_bucket: Optional[str] = None
-    s3_region: Optional[str] = None
-    aws_role_arn: Optional[str] = None
-    aws_access_key_id: Optional[str] = None
-    aws_access_key_secret: Optional[str] = None
-```
-
-### 3. **Platform Registration System**
-
-#### Registry Pattern (registry.py)
- Automatic registration of built-in platforms
- Runtime platform discovery
- Type-safe client instantiation
-
-```python
-# Auto-registration of platforms
-_PLATFORMS: Dict[str, Type[VideoPlatformClient]] = {}
-
-def register_platform(name: str, client_class: Type[VideoPlatformClient])
-def get_platform_client(platform: str, config: VideoPlatformConfig) -> VideoPlatformClient
-```
-
-#### Factory System (factory.py)
- Configuration management per platform
- Platform selection logic
- Feature flag integration
-
-```python
-def get_platform_for_room(room_id: Optional[str] = None) -> str:
-    """Determine which platform to use based on feature flags."""
-    if not settings.DAILY_MIGRATION_ENABLED:
-        return "whereby"
-
-    if room_id and room_id in settings.DAILY_MIGRATION_ROOM_IDS:
-        return "daily"
-
-    return settings.DEFAULT_VIDEO_PLATFORM
-```
-
-### 4. **Database Schema Changes**
-
-#### Room Model Updates
-Added `platform` field to track which video platform each room uses:
-
-```python
-# Database Schema
-platform_column = sqlalchemy.Column(
-    "platform",
-    sqlalchemy.String,
-    nullable=False,
-    server_default="whereby"
-)
-
-# Pydantic Model
-class Room(BaseModel):
-    platform: Literal["whereby", "daily"] = "whereby"
-```
-
-#### Meeting Model Updates
-Added `platform` field to meetings for tracking and debugging:
-
-```python
-# Database Schema
-platform_column = sqlalchemy.Column(
-    "platform",
-    sqlalchemy.String,
-    nullable=False,
-    server_default="whereby"
-)
-
-# Pydantic Model
-class Meeting(BaseModel):
-    platform: Literal["whereby", "daily"] = "whereby"
-```
-
-**Key Decision**: No platform-specific fields were added to models. Instead, the `extra_data` field in `MeetingData` handles platform-specific information, following the user's rule of using generic `provider_data` as JSON if needed.
-
-### 5. **Settings Configuration**
-
-#### Feature Flags
-```python
-# Migration control
-DAILY_MIGRATION_ENABLED: bool = True
-DAILY_MIGRATION_ROOM_IDS: list[str] = []
-DEFAULT_VIDEO_PLATFORM: str = "daily"
-
-# Daily.co specific settings
-DAILY_API_KEY: str | None = None
-DAILY_WEBHOOK_SECRET: str | None = None
-DAILY_SUBDOMAIN: str | None = None
-AWS_DAILY_S3_BUCKET: str | None = None
-AWS_DAILY_S3_REGION: str = "us-west-2"
-AWS_DAILY_ROLE_ARN: str | None = None
-```
-
-#### Configuration Pattern
-Each platform gets its own configuration namespace while sharing common patterns:
-
-```python
-def get_platform_config(platform: str) -> VideoPlatformConfig:
-    if platform == "whereby":
-        return VideoPlatformConfig(
-            api_key=settings.WHEREBY_API_KEY or "",
-            webhook_secret=settings.WHEREBY_WEBHOOK_SECRET or "",
-            # ... whereby-specific config
-        )
-    elif platform == "daily":
-        return VideoPlatformConfig(
-            api_key=settings.DAILY_API_KEY or "",
-            webhook_secret=settings.DAILY_WEBHOOK_SECRET or "",
-            # ... daily-specific config
-        )
-```
-
-### 6. **API Integration Updates**
-
-#### Room Creation (views/rooms.py)
-Updated to use platform factory instead of direct Whereby calls:
-
-```python
-@router.post("/rooms/{room_name}/meeting")
-async def rooms_create_meeting(room_name: str, user: UserInfo):
-    # OLD: Direct Whereby integration
-    # whereby_meeting = await create_meeting("", end_date=end_date, room=room)
-
-    # NEW: Platform abstraction
-    platform = get_platform_for_room(room.id)
-    client = create_platform_client(platform)
-
-    meeting_data = await client.create_meeting(
-        room_name_prefix=room.name, end_date=end_date, room=room
-    )
-
-    await client.upload_logo(meeting_data.room_name, "./images/logo.png")
-```
-
-### 7. **Webhook Handling**
-
-#### Separate Webhook Endpoints
-Each platform gets its own webhook endpoint with platform-specific signature verification:
-
-```python
-# views/daily.py
-@router.post("/daily_webhook")
-async def daily_webhook(event: DailyWebhookEvent, request: Request):
-    # Verify Daily.co signature
-    body = await request.body()
-    signature = request.headers.get("X-Daily-Signature", "")
-
-    if not verify_daily_webhook_signature(body, signature):
-        raise HTTPException(status_code=401)
-
-    # Handle platform-specific events
-    if event.type == "participant.joined":
-        await _handle_participant_joined(event)
-```
-
-#### Consistent Event Handling
-Despite different event formats, the core business logic remains the same:
-
-```python
-async def _handle_participant_joined(event):
-    room_name = event.data.get("room", {}).get("name")  # Daily.co format
-    meeting = await meetings_controller.get_by_room_name(room_name)
-    if meeting:
-        current_count = getattr(meeting, "num_clients", 0)
-        await meetings_controller.update_meeting(
-            meeting.id, num_clients=current_count + 1
-        )
-```
-
-### 8. **Worker Task Integration**
-
-#### New Task for Daily.co Recording Processing
-Added platform-specific recording processing while maintaining the same pipeline:
-
-```python
-@shared_task
-@asynctask
-async def process_recording_from_url(recording_url: str, meeting_id: str, recording_id: str):
-    """Process recording from Direct URL (Daily.co webhook)."""
-    logger.info("Processing recording from URL for meeting: %s", meeting_id)
-    # Uses same processing pipeline as Whereby S3 recordings
-```
-
-**Key Decision**: Worker tasks remain in main worker module but could be moved to platform-specific folders as suggested by the user.
-
-### 9. **Testing Infrastructure**
-
-#### Comprehensive Test Suite
- Unit tests for each platform client
- Integration tests for platform switching
- Mock platform for testing without external dependencies
- Webhook signature verification tests
-
-```python
-class TestPlatformIntegration:
-    """Integration tests for platform switching."""
-
-    async def test_platform_switching_preserves_interface(self):
-        """Test that different platforms provide consistent interface."""
-        # Test both Mock and Daily platforms return MeetingData objects
-        # with consistent fields
-```
-
-## Implementation Patterns for Jitsi Integration
-
-Based on the daily.co implementation, here's how Jitsi should be integrated:
-
-### 1. **Jitsi Client Implementation**
-
-```python
-# video_platforms/jitsi.py
-class JitsiClient(VideoPlatformClient):
-    PLATFORM_NAME = "jitsi"
-
-    async def create_meeting(self, room_name_prefix: str, end_date: datetime, room: Room) -> MeetingData:
-        # Generate unique room name
-        jitsi_room = f"reflector-{room.name}-{int(time.time())}"
-
-        # Generate JWT tokens
-        user_jwt = self._generate_jwt(room=jitsi_room, moderator=False, exp=end_date)
-        host_jwt = self._generate_jwt(room=jitsi_room, moderator=True, exp=end_date)
-
-        return MeetingData(
-            meeting_id=generate_uuid4(),
-            room_name=jitsi_room,
-            room_url=f"https://jitsi.domain/{jitsi_room}?jwt={user_jwt}",
-            host_room_url=f"https://jitsi.domain/{jitsi_room}?jwt={host_jwt}",
-            platform=self.PLATFORM_NAME,
-            extra_data={"user_jwt": user_jwt, "host_jwt": host_jwt}
-        )
-```
-
-### 2. **Settings Integration**
-
-```python
-# settings.py
-JITSI_DOMAIN: str = "meet.jit.si"
-JITSI_JWT_SECRET: str | None = None
-JITSI_WEBHOOK_SECRET: str | None = None
-JITSI_API_URL: str | None = None  # If using Jitsi API
-```
-
-### 3. **Factory Registration**
-
-```python
-# registry.py
-def _register_builtin_platforms():
-    from .jitsi import JitsiClient
-    register_platform("jitsi", JitsiClient)
-
-# factory.py
-def get_platform_config(platform: str) -> VideoPlatformConfig:
-    elif platform == "jitsi":
-        return VideoPlatformConfig(
-            api_key="",  # Jitsi may not need API key
-            webhook_secret=settings.JITSI_WEBHOOK_SECRET or "",
-            api_url=settings.JITSI_API_URL,
-        )
-```
-
-### 4. **Webhook Integration**
-
-```python
-# views/jitsi.py
-@router.post("/jitsi/events")
-async def jitsi_events_webhook(event_data: dict):
-    # Handle Prosody event-sync webhook format
-    event_type = event_data.get("event")
-    room_name = event_data.get("room", "").split("@")[0]
-
-    if event_type == "muc-occupant-joined":
-        # Same participant handling logic as other platforms
-```
-
-## Key Benefits of This Architecture
-
-### 1. **Isolation and Organization**
- Platform-specific code contained in separate modules
- No platform logic leaking into core application
- Easy to add/remove platforms without affecting others
-
-### 2. **Consistent Interface**
- All platforms implement the same abstract methods
- Standardized `MeetingData` structure
- Uniform error handling and logging
-
-### 3. **Gradual Migration Support**
- Feature flags for controlled rollouts
- Room-specific platform selection
- Fallback mechanisms for platform failures
-
-### 4. **Configuration Management**
- Centralized settings per platform
- Consistent naming patterns
- Environment-based configuration
-
-### 5. **Testing and Quality**
- Mock platform for testing
- Comprehensive test coverage
- Platform-specific test utilities
-
-## Migration Strategy Applied
-
-The daily.co implementation demonstrates a careful migration approach:
-
-### 1. **Backward Compatibility**
- Default platform remains "whereby"
- Existing rooms continue using Whereby unless explicitly migrated
- Same API endpoints and response formats
-
-### 2. **Feature Flag Control**
-```python
-# Gradual rollout control
-DAILY_MIGRATION_ENABLED: bool = True
-DAILY_MIGRATION_ROOM_IDS: list[str] = []  # Specific rooms to migrate
-DEFAULT_VIDEO_PLATFORM: str = "daily"     # New rooms default
-```
-
-### 3. **Data Integrity**
- Platform field tracks which service each room/meeting uses
- No data loss during migration
- Platform-specific data preserved in `extra_data`
-
-### 4. **Monitoring and Rollback**
- Comprehensive logging of platform selection
- Easy rollback by changing feature flags
- Platform-specific error tracking
-
-## Recommendations for Jitsi Integration
-
-Based on this analysis and the user's requirements:
-
-### 1. **Follow the Pattern**
- Create `video_platforms/jitsi/` directory with:
-  - `client.py` - Main JitsiClient implementation
-  - `tasks.py` - Jitsi-specific worker tasks
-  - `__init__.py` - Module exports
-
-### 2. **Settings Organization**
- Use `JITSI_*` prefix for all Jitsi settings
- Follow the same configuration pattern as Daily.co
- Support both environment variables and config files
-
-### 3. **Generic Database Fields**
- Avoid platform-specific columns in database
- Use `provider_data` JSON field if platform-specific data needed
- Keep `platform` field as simple string identifier
-
-### 4. **Worker Task Migration**
-According to user requirements, migrate platform-specific tasks:
-```
-video_platforms/
-├── whereby/
-│   ├── client.py  (moved from whereby.py)
-│   └── tasks.py   (moved from worker/whereby_tasks.py)
-├── daily/
-│   ├── client.py  (moved from daily.py)
-│   └── tasks.py   (moved from worker/daily_tasks.py)
-└── jitsi/
-    ├── client.py  (new JitsiClient)
-    └── tasks.py   (new Jitsi recording tasks)
-```
-
-### 5. **Webhook Architecture**
- Create `views/jitsi.py` for Jitsi-specific webhooks
- Follow the same signature verification pattern
- Reuse existing participant tracking logic
-
-## Implementation Checklist for Jitsi
-
- [ ] Create `video_platforms/jitsi/` directory structure
- [ ] Implement `JitsiClient` following the abstract interface
- [ ] Add Jitsi settings to configuration
- [ ] Register Jitsi platform in factory/registry
- [ ] Create Jitsi webhook endpoint
- [ ] Implement JWT token generation for room access
- [ ] Add Jitsi recording processing tasks
- [ ] Create comprehensive test suite
- [ ] Update database migrations for platform field
- [ ] Document Jitsi-specific configuration
-
-## Conclusion
-
-The video platforms refactoring in PR #529 provides an excellent foundation for adding Jitsi support. The architecture is well-designed with clear separation of concerns, consistent interfaces, and excellent extensibility. The daily.co implementation demonstrates how to add a new platform while maintaining backward compatibility and providing gradual migration capabilities.
-
-The pattern should be directly applicable to Jitsi integration, with the main differences being:
- JWT-based authentication instead of API keys
- Different webhook event formats
- Jibri recording pipeline integration
- Self-hosted deployment considerations
-
-This architecture successfully achieves the user's goals of:
-1. Settings-based configuration
-2. Generic database fields (no provider-specific columns)
-3. Platform isolation in separate directories
-4. Worker task organization within platform folders
--- a/server/.gitignore
+++ b/server/.gitignore
@@ -176,8 +176,7 @@ artefacts/
 audio_*.wav

 # ignore local database
-*.sqlite3
-*.db
+reflector.sqlite3
 data/

 dump.rdb
--- a/server/Dockerfile
+++ b/server/Dockerfile
@@ -1,8 +1,7 @@
 FROM python:3.12-slim

 ENV PYTHONUNBUFFERED=1 \
-    UV_LINK_MODE=copy \
-    UV_NO_CACHE=1
+    UV_LINK_MODE=copy

 # builder install base dependencies
 WORKDIR /tmp
@@ -14,8 +13,8 @@ ENV PATH="/root/.local/bin/:$PATH"
 # install application dependencies
 RUN mkdir -p /app
 WORKDIR /app
-COPY pyproject.toml uv.lock README.md /app/
-RUN uv sync --compile-bytecode --locked
+COPY pyproject.toml uv.lock /app/
+RUN touch README.md && env uv sync --compile-bytecode --locked

 # pre-download nltk packages
 RUN uv run python -c "import nltk; nltk.download('punkt_tab'); nltk.download('averaged_perceptron_tagger_eng')"
@@ -27,15 +26,4 @@ COPY migrations /app/migrations
 COPY reflector /app/reflector
 WORKDIR /app

-# Create symlink for libgomp if it doesn't exist (for ARM64 compatibility)
-RUN if [ "$(uname -m)" = "aarch64" ] && [ ! -f /usr/lib/libgomp.so.1 ]; then \
-    LIBGOMP_PATH=$(find /app/.venv/lib -path "*/torch.libs/libgomp*.so.*" 2>/dev/null | head -n1); \
-    if [ -n "$LIBGOMP_PATH" ]; then \
-    ln -sf "$LIBGOMP_PATH" /usr/lib/libgomp.so.1; \
-    fi \
-    fi
-
-# Pre-check just to make sure the image will not fail
-RUN uv run python -c "import silero_vad.model"
-
 CMD ["./runserver.sh"]
--- a/server/README.md
+++ b/server/README.md
@@ -40,5 +40,3 @@ uv run python -c "from reflector.pipelines.main_live_pipeline import task_pipeli
 ```bash
 uv run python -c "from reflector.pipelines.main_live_pipeline import pipeline_post; pipeline_post(transcript_id='TRANSCRIPT_ID')"
 ```
-
-.
--- a/server/contrib/jitsi/README.md
+++ b/server/contrib/jitsi/README.md
@@ -1,212 +0,0 @@
-# Event Logger for Docker-Jitsi-Meet
-
-A Prosody module that logs Jitsi meeting events to JSONL files alongside recordings, enabling complete participant tracking and speaker statistics.
-
-## Prerequisites
-
- Running docker-jitsi-meet installation
- Jibri configured for recording
-
-## Installation
-
-### Step 1: Copy the Module
-
-Copy the Prosody module to your custom plugins directory:
-
-```bash
-# Create the directory if it doesn't exist
-mkdir -p ~/.jitsi-meet-cfg/prosody/prosody-plugins-custom
-
-# Copy the module
-cp mod_event_logger.lua ~/.jitsi-meet-cfg/prosody/prosody-plugins-custom/
-```
-
-### Step 2: Update Your .env File
-
-Add or modify these variables in your `.env` file:
-
-```bash
-# If XMPP_MUC_MODULES already exists, append event_logger
-# Example: XMPP_MUC_MODULES=existing_module,event_logger
-XMPP_MUC_MODULES=event_logger
-
-# Optional: Configure the module (these are defaults)
-JIBRI_RECORDINGS_PATH=/config/recordings
-JIBRI_LOG_SPEAKER_STATS=true
-JIBRI_SPEAKER_STATS_INTERVAL=10
-```
-
-**Important**: If you already have `XMPP_MUC_MODULES` defined, add `event_logger` to the comma-separated list:
-```bash
-# Existing modules + our module
-XMPP_MUC_MODULES=mod_info,mod_alert,event_logger
-```
-
-### Step 3: Modify docker-compose.yml
-
-Add a shared recordings volume so Prosody can write events alongside Jibri recordings:
-
-```yaml
-services:
-  prosody:
-    # ... existing configuration ...
-    volumes:
-      - ${CONFIG}/prosody/config:/config:Z
-      - ${CONFIG}/prosody/prosody-plugins-custom:/prosody-plugins-custom:Z
-      - ${CONFIG}/recordings:/config/recordings:Z  # Add this line
-    environment:
-      # Add if not using .env file
-      - XMPP_MUC_MODULES=${XMPP_MUC_MODULES:-event_logger}
-      - JIBRI_RECORDINGS_PATH=/config/recordings
-
-  jibri:
-    # ... existing configuration ...
-    volumes:
-      - ${CONFIG}/jibri:/config:Z
-      - ${CONFIG}/recordings:/config/recordings:Z  # Add this line
-    environment:
-      # For Reflector webhook integration (optional)
-      - REFLECTOR_WEBHOOK_URL=${REFLECTOR_WEBHOOK_URL:-}
-      - JIBRI_FINALIZE_RECORDING_SCRIPT_PATH=/config/finalize.sh
-```
-
-### Step 4: Add Finalize Script (Optional - For Reflector Integration)
-
-If you want to notify Reflector when recordings complete:
-
-```bash
-# Copy the finalize script
-cp finalize.sh ~/.jitsi-meet-cfg/jibri/finalize.sh
-chmod +x ~/.jitsi-meet-cfg/jibri/finalize.sh
-
-# Add to .env
-REFLECTOR_WEBHOOK_URL=http://your-reflector-api:8000
-```
-
-### Step 5: Restart Services
-
-```bash
-docker-compose down
-docker-compose up -d
-```
-
-## What Gets Created
-
-After a recording, you'll find in `~/.jitsi-meet-cfg/recordings/{session-id}/`:
- `recording.mp4` - The video recording (created by Jibri)
- `metadata.json` - Basic metadata (created by Jibri)
- `events.jsonl` - Complete participant timeline (created by this module)
-
-## Event Format
-
-Each line in `events.jsonl` is a JSON object:
-
-```json
-{"type":"room_created","timestamp":1234567890,"room_name":"TestRoom","room_jid":"testroom@conference.meet.jitsi","meeting_url":"https://meet.jitsi/TestRoom"}
-{"type":"recording_started","timestamp":1234567891,"room_name":"TestRoom","session_id":"20240115120000_TestRoom","jibri_jid":"jibri@recorder.meet.jitsi"}
-{"type":"participant_joined","timestamp":1234567892,"room_name":"TestRoom","participant":{"jid":"user1@meet.jitsi/web","nick":"John Doe","id":"user1@meet.jitsi","is_moderator":false}}
-{"type":"speaker_active","timestamp":1234567895,"room_name":"TestRoom","speaker_jid":"user1@meet.jitsi","speaker_nick":"John Doe","duration":10}
-{"type":"participant_left","timestamp":1234567920,"room_name":"TestRoom","participant":{"jid":"user1@meet.jitsi/web","nick":"John Doe","duration_seconds":28}}
-{"type":"recording_stopped","timestamp":1234567950,"room_name":"TestRoom","session_id":"20240115120000_TestRoom","meeting_url":"https://meet.jitsi/TestRoom"}
-```
-
-## Configuration Options
-
-All configuration can be done via environment variables:
-
-| Environment Variable | Default | Description |
-|---------------------|---------|-------------|
-| `JIBRI_RECORDINGS_PATH` | `/config/recordings` | Path where recordings are stored |
-| `JIBRI_LOG_SPEAKER_STATS` | `true` | Enable speaker statistics logging |
-| `JIBRI_SPEAKER_STATS_INTERVAL` | `10` | Seconds between speaker stats updates |
-
-## Verifying Installation
-
-Check that the module is loaded:
-```bash
-docker-compose logs prosody | grep "Event Logger"
-# Should see: "Event Logger loaded - writing to /config/recordings"
-```
-
-Check for events after a recording:
-```bash
-ls -la ~/.jitsi-meet-cfg/recordings/*/events.jsonl
-cat ~/.jitsi-meet-cfg/recordings/*/events.jsonl | jq .
-```
-
-## Troubleshooting
-
-### No events.jsonl file created
-
-1. **Check module is enabled**:
-   ```bash
-   docker-compose exec prosody grep -r "event_logger" /config
-   ```
-
-2. **Verify volume permissions**:
-   ```bash
-   docker-compose exec prosody ls -la /config/recordings
-   ```
-
-3. **Check Prosody logs for errors**:
-   ```bash
-   docker-compose logs prosody | grep -i error
-   ```
-
-### Module not loading
-
-1. **Verify file exists in container**:
-   ```bash
-   docker-compose exec prosody ls -la /prosody-plugins-custom/
-   ```
-
-2. **Check XMPP_MUC_MODULES format** (must be comma-separated, no spaces):
-   - ✅ Correct: `XMPP_MUC_MODULES=mod1,mod2,event_logger`
-   - ❌ Wrong: `XMPP_MUC_MODULES=mod1, mod2, event_logger`
-
-## Common docker-compose.yml Patterns
-
-### Minimal Addition (if you trust defaults)
-```yaml
-services:
-  prosody:
-    volumes:
-      - ${CONFIG}/recordings:/config/recordings:Z  # Just add this
-```
-
-### Full Configuration
-```yaml
-services:
-  prosody:
-    volumes:
-      - ${CONFIG}/prosody/config:/config:Z
-      - ${CONFIG}/prosody/prosody-plugins-custom:/prosody-plugins-custom:Z
-      - ${CONFIG}/recordings:/config/recordings:Z
-    environment:
-      - XMPP_MUC_MODULES=event_logger
-      - JIBRI_RECORDINGS_PATH=/config/recordings
-      - JIBRI_LOG_SPEAKER_STATS=true
-      - JIBRI_SPEAKER_STATS_INTERVAL=10
-
-  jibri:
-    volumes:
-      - ${CONFIG}/jibri:/config:Z
-      - ${CONFIG}/recordings:/config/recordings:Z
-    environment:
-      - JIBRI_RECORDING_DIR=/config/recordings
-      - JIBRI_FINALIZE_RECORDING_SCRIPT_PATH=/config/finalize.sh
-```
-
-## Integration with Reflector
-
-The finalize.sh script will automatically notify Reflector when a recording completes if `REFLECTOR_WEBHOOK_URL` is set. Reflector will receive:
-
-```json
-{
-  "session_id": "20240115120000_TestRoom",
-  "path": "20240115120000_TestRoom",
-  "meeting_url": "https://meet.jitsi/TestRoom"
-}
-```
-
-Reflector then processes the recording along with the complete participant timeline from `events.jsonl`.
--- a/server/contrib/jitsi/finalize.sh
+++ b/server/contrib/jitsi/finalize.sh
@@ -1,49 +0,0 @@
-#!/bin/bash
-# Jibri finalize script to notify Reflector when recording is complete
-# This script is called by Jibri with the recording directory as argument
-
-RECORDING_PATH="$1"
-SESSION_ID=$(basename "$RECORDING_PATH")
-METADATA_FILE="$RECORDING_PATH/metadata.json"
-
-# Extract meeting URL from Jibri's metadata
-MEETING_URL=""
-if [ -f "$METADATA_FILE" ]; then
-    MEETING_URL=$(jq -r '.meeting_url' "$METADATA_FILE" 2>/dev/null || echo "")
-fi
-
-echo "[$(date)] Recording finalized: $RECORDING_PATH"
-echo "[$(date)] Session ID: $SESSION_ID"
-echo "[$(date)] Meeting URL: $MEETING_URL"
-
-# Check if events.jsonl was created by our Prosody module
-if [ -f "$RECORDING_PATH/events.jsonl" ]; then
-    EVENT_COUNT=$(wc -l < "$RECORDING_PATH/events.jsonl")
-    echo "[$(date)] Found events.jsonl with $EVENT_COUNT events"
-else
-    echo "[$(date)] Warning: No events.jsonl found"
-fi
-
-# Notify Reflector if webhook URL is configured
-if [ -n "$REFLECTOR_WEBHOOK_URL" ]; then
-    echo "[$(date)] Notifying Reflector at: $REFLECTOR_WEBHOOK_URL"
-
-    RESPONSE=$(curl -s -w "\n%{http_code}" -X POST "$REFLECTOR_WEBHOOK_URL/api/v1/jibri/recording-ready" \
-        -H "Content-Type: application/json" \
-        -d "{\"session_id\":\"$SESSION_ID\",\"path\":\"$SESSION_ID\",\"meeting_url\":\"$MEETING_URL\"}")
-
-    HTTP_CODE=$(echo "$RESPONSE" | tail -n1)
-    BODY=$(echo "$RESPONSE" | sed '$d')
-
-    if [ "$HTTP_CODE" = "200" ]; then
-        echo "[$(date)] Reflector notified successfully"
-        echo "[$(date)] Response: $BODY"
-    else
-        echo "[$(date)] Failed to notify Reflector. HTTP code: $HTTP_CODE"
-        echo "[$(date)] Response: $BODY"
-    fi
-else
-    echo "[$(date)] No REFLECTOR_WEBHOOK_URL configured, skipping notification"
-fi
-
-echo "[$(date)] Finalize script completed"
--- a/server/contrib/jitsi/mod_event_logger.lua
+++ b/server/contrib/jitsi/mod_event_logger.lua
@@ -1,372 +0,0 @@
-local json = require "util.json"
-local st = require "util.stanza"
-local jid_bare = require "util.jid".bare
-
-local recordings_path = os.getenv("JIBRI_RECORDINGS_PATH") or
-                       module:get_option_string("jibri_recordings_path", "/recordings")
-
-- room_jid -> { session_id, participants = {jid -> info} }
-local active_recordings = {}
-- room_jid -> { participants = {jid -> info}, created_at }
-local room_states = {}
-
-local function get_timestamp()
-    return os.time()
-end
-
-local function write_event(session_id, event)
-    if not session_id then
-        module:log("warn", "No session_id for event: %s", event.type)
-        return
-    end
-
-    local session_dir = string.format("%s/%s", recordings_path, session_id)
-    local event_file = string.format("%s/events.jsonl", session_dir)
-
-    module:log("info", "Writing event %s to %s", event.type, event_file)
-
-    -- Create directory
-    local mkdir_cmd = string.format("mkdir -p '%s' 2>&1", session_dir)
-    local mkdir_result = os.execute(mkdir_cmd)
-    module:log("debug", "mkdir result: %s", tostring(mkdir_result))
-
-    local file, err = io.open(event_file, "a")
-    if file then
-        local json_str = json.encode(event)
-        file:write(json_str .. "\n")
-        file:close()
-        module:log("info", "Successfully wrote event %s", event.type)
-    else
-        module:log("error", "Failed to write event to %s: %s", event_file, err)
-    end
-end
-
-local function extract_participant_info(occupant)
-    local info = {
-        jid = occupant.jid,
-        bare_jid = occupant.bare_jid,
-        nick = occupant.nick,
-        display_name = nil,
-        role = occupant.role
-    }
-
-    local presence = occupant:get_presence()
-    if presence then
-        local nick_element = presence:get_child("nick", "http://jabber.org/protocol/nick")
-        if nick_element then
-            info.display_name = nick_element:get_text()
-        end
-
-        local identity = presence:get_child("identity")
-        if identity then
-            local user = identity:get_child("user")
-            if user then
-                local name = user:get_child("name")
-                if name then
-                    info.display_name = name:get_text()
-                end
-
-                local id_element = user:get_child("id")
-                if id_element then
-                    info.id = id_element:get_text()
-                end
-            end
-        end
-
-        if not info.display_name and occupant.nick then
-            local _, _, resource = occupant.nick:match("([^@]+)@([^/]+)/(.+)")
-            if resource then
-                info.display_name = resource
-            end
-        end
-    end
-
-    return info
-end
-
-local function get_room_participant_count(room)
-    local count = 0
-    for _ in room:each_occupant() do
-        count = count + 1
-    end
-    return count
-end
-
-local function snapshot_room_participants(room)
-    local participants = {}
-    local total = 0
-    local skipped = 0
-
-    module:log("info", "Snapshotting room participants")
-
-    for _, occupant in room:each_occupant() do
-        total = total + 1
-        -- Skip recorders (Jibri)
-        if occupant.bare_jid and (occupant.bare_jid:match("^recorder@") or
-                                  occupant.bare_jid:match("^jibri@")) then
-            skipped = skipped + 1
-        else
-            local info = extract_participant_info(occupant)
-            participants[occupant.jid] = info
-            module:log("debug", "Added participant: %s", info.display_name or info.bare_jid)
-        end
-    end
-
-    module:log("info", "Snapshot: %d total, %d participants", total, total - skipped)
-    return participants
-end
-
-- Import utility functions if available
-local util = module:require "util";
-local get_room_from_jid = util.get_room_from_jid;
-local room_jid_match_rewrite = util.room_jid_match_rewrite;
-
-- Main IQ handler for Jibri stanzas
-module:hook("pre-iq/full", function(event)
-    local stanza = event.stanza
-    if stanza.name ~= "iq" then
-        return
-    end
-
-    local jibri = stanza:get_child('jibri', 'http://jitsi.org/protocol/jibri')
-    if not jibri then
-        return
-    end
-
-    module:log("info", "=== Jibri IQ intercepted ===")
-
-    local action = jibri.attr.action
-    local session_id = jibri.attr.session_id
-    local room_jid = jibri.attr.room
-    local recording_mode = jibri.attr.recording_mode
-    local app_data = jibri.attr.app_data
-
-    module:log("info", "Jibri %s - session: %s, room: %s, mode: %s",
-               action or "?", session_id or "?", room_jid or "?", recording_mode or "?")
-
-    if not room_jid or not session_id then
-        module:log("warn", "Missing room_jid or session_id")
-        return
-    end
-
-    -- Get the room using util function
-    local room = get_room_from_jid(room_jid_match_rewrite(jid_bare(stanza.attr.to)))
-    if not room then
-        -- Try with the room_jid directly
-        room = get_room_from_jid(room_jid)
-    end
-
-    if not room then
-        module:log("error", "Room not found for jid: %s", room_jid)
-        return
-    end
-
-    module:log("info", "Room found: %s", room:get_name() or room_jid)
-
-    if action == "start" then
-        module:log("info", "Recording START for session %s", session_id)
-
-        -- Count and snapshot participants
-        local participant_count = 0
-        for _ in room:each_occupant() do
-            participant_count = participant_count + 1
-        end
-
-        local participants = snapshot_room_participants(room)
-        local participant_list = {}
-        for jid, info in pairs(participants) do
-            table.insert(participant_list, info)
-        end
-
-        active_recordings[room_jid] = {
-            session_id = session_id,
-            participants = participants,
-            started_at = get_timestamp()
-        }
-
-        write_event(session_id, {
-            type = "recording_started",
-            timestamp = get_timestamp(),
-            room_jid = room_jid,
-            room_name = room:get_name(),
-            session_id = session_id,
-            recording_mode = recording_mode,
-            app_data = app_data,
-            participant_count = participant_count,
-            participants_at_start = participant_list
-        })
-
-    elseif action == "stop" then
-        module:log("info", "Recording STOP for session %s", session_id)
-
-        local recording = active_recordings[room_jid]
-        if recording and recording.session_id == session_id then
-            write_event(session_id, {
-                type = "recording_stopped",
-                timestamp = get_timestamp(),
-                room_jid = room_jid,
-                room_name = room:get_name(),
-                session_id = session_id,
-                duration = get_timestamp() - recording.started_at,
-                participant_count = get_room_participant_count(room)
-            })
-
-            active_recordings[room_jid] = nil
-        else
-            module:log("warn", "No active recording found for room %s", room_jid)
-        end
-    end
-end);
-
-- Room and participant event hooks
-local function setup_room_hooks(host_module)
-    module:log("info", "Setting up room hooks on %s", host_module.host or "unknown")
-
-    -- Room created
-    host_module:hook("muc-room-created", function(event)
-        local room = event.room
-        local room_jid = room.jid
-
-        room_states[room_jid] = {
-            participants = {},
-            created_at = get_timestamp()
-        }
-
-        module:log("info", "Room created: %s", room_jid)
-    end)
-
-    -- Room destroyed
-    host_module:hook("muc-room-destroyed", function(event)
-        local room = event.room
-        local room_jid = room.jid
-
-        room_states[room_jid] = nil
-        active_recordings[room_jid] = nil
-
-        module:log("info", "Room destroyed: %s", room_jid)
-    end)
-
-    -- Occupant joined
-    host_module:hook("muc-occupant-joined", function(event)
-        local room = event.room
-        local occupant = event.occupant
-        local room_jid = room.jid
-
-        -- Skip recorders
-        if occupant.bare_jid and (occupant.bare_jid:match("^recorder@") or
-                                   occupant.bare_jid:match("^jibri@")) then
-            return
-        end
-
-        local participant_info = extract_participant_info(occupant)
-
-        -- Update room state
-        if room_states[room_jid] then
-            room_states[room_jid].participants[occupant.jid] = participant_info
-        end
-
-        -- Log to active recording if exists
-        local recording = active_recordings[room_jid]
-        if recording then
-            recording.participants[occupant.jid] = participant_info
-
-            write_event(recording.session_id, {
-                type = "participant_joined",
-                timestamp = get_timestamp(),
-                room_jid = room_jid,
-                room_name = room:get_name(),
-                participant = participant_info,
-                participant_count = get_room_participant_count(room)
-            })
-        end
-
-        module:log("info", "Participant joined %s: %s (%d total)",
-                   room:get_name() or room_jid,
-                   participant_info.display_name or participant_info.bare_jid,
-                   get_room_participant_count(room))
-    end)
-
-    -- Occupant left
-    host_module:hook("muc-occupant-left", function(event)
-        local room = event.room
-        local occupant = event.occupant
-        local room_jid = room.jid
-
-        -- Skip recorders
-        if occupant.bare_jid and (occupant.bare_jid:match("^recorder@") or
-                                   occupant.bare_jid:match("^jibri@")) then
-            return
-        end
-
-        local participant_info = extract_participant_info(occupant)
-
-        -- Update room state
-        if room_states[room_jid] then
-            room_states[room_jid].participants[occupant.jid] = nil
-        end
-
-        -- Log to active recording if exists
-        local recording = active_recordings[room_jid]
-        if recording then
-            if recording.participants[occupant.jid] then
-                recording.participants[occupant.jid] = nil
-            end
-
-            write_event(recording.session_id, {
-                type = "participant_left",
-                timestamp = get_timestamp(),
-                room_jid = room_jid,
-                room_name = room:get_name(),
-                participant = participant_info,
-                participant_count = get_room_participant_count(room)
-            })
-        end
-
-        module:log("info", "Participant left %s: %s (%d remaining)",
-                   room:get_name() or room_jid,
-                   participant_info.display_name or participant_info.bare_jid,
-                   get_room_participant_count(room))
-    end)
-end
-
-- Module initialization
-local current_host = module:get_host()
-local host_type = module:get_host_type()
-
-module:log("info", "Event Logger loading on %s (type: %s)", current_host, host_type or "unknown")
-module:log("info", "Recording path: %s", recordings_path)
-
-- Setup room hooks based on host type
-if host_type == "component" and current_host:match("^[^.]+%.") then
-    setup_room_hooks(module)
-else
-    -- Try to find and hook to MUC component
-    local process_host_module = util.process_host_module
-    local muc_component_host = module:get_option_string("muc_component") or
-                              module:get_option_string("main_muc")
-
-    if not muc_component_host then
-        local possible_hosts = {
-            "muc." .. current_host,
-            "conference." .. current_host,
-            "rooms." .. current_host
-        }
-
-        for _, host in ipairs(possible_hosts) do
-            if prosody.hosts[host] then
-                muc_component_host = host
-                module:log("info", "Auto-detected MUC component: %s", muc_component_host)
-                break
-            end
-        end
-    end
-
-    if muc_component_host then
-        process_host_module(muc_component_host, function(host_module, host)
-            module:log("info", "Hooking to MUC events on %s", host)
-            setup_room_hooks(host_module)
-        end)
-    else
-        module:log("error", "Could not find MUC component")
-    end
-end
--- a/server/docs/data_retention.md
+++ b/server/docs/data_retention.md
@@ -1,95 +0,0 @@
-# Data Retention and Cleanup
-
-## Overview
-
-For public instances of Reflector, a data retention policy is automatically enforced to delete anonymous user data after a configurable period (default: 7 days). This ensures compliance with privacy expectations and prevents unbounded storage growth.
-
-## Configuration
-
-### Environment Variables
-
- `PUBLIC_MODE` (bool): Must be set to `true` to enable automatic cleanup
- `PUBLIC_DATA_RETENTION_DAYS` (int): Number of days to retain anonymous data (default: 7)
-
-### What Gets Deleted
-
-When data reaches the retention period, the following items are automatically removed:
-
-1. **Transcripts** from anonymous users (where `user_id` is NULL):
-   - Database records
-   - Local files (audio.wav, audio.mp3, audio.json waveform)
-   - Storage files (cloud storage if configured)
-
-## Automatic Cleanup
-
-### Celery Beat Schedule
-
-When `PUBLIC_MODE=true`, a Celery beat task runs daily at 3 AM to clean up old data:
-
-```python
-# Automatically scheduled when PUBLIC_MODE=true
-"cleanup_old_public_data": {
-    "task": "reflector.worker.cleanup.cleanup_old_public_data",
-    "schedule": crontab(hour=3, minute=0),  # Daily at 3 AM
-}
-```
-
-### Running the Worker
-
-Ensure both Celery worker and beat scheduler are running:
-
-```bash
-# Start Celery worker
-uv run celery -A reflector.worker.app worker --loglevel=info
-
-# Start Celery beat scheduler (in another terminal)
-uv run celery -A reflector.worker.app beat
-```
-
-## Manual Cleanup
-
-For testing or manual intervention, use the cleanup tool:
-
-```bash
-# Delete data older than 7 days (default)
-uv run python -m reflector.tools.cleanup_old_data
-
-# Delete data older than 30 days
-uv run python -m reflector.tools.cleanup_old_data --days 30
-```
-
-Note: The manual tool uses the same implementation as the Celery worker task to ensure consistency.
-
-## Important Notes
-
-1. **User Data Deletion**: Only anonymous data (where `user_id` is NULL) is deleted. Authenticated user data is preserved.
-
-2. **Storage Cleanup**: The system properly cleans up both local files and cloud storage when configured.
-
-3. **Error Handling**: If individual deletions fail, the cleanup continues and logs errors. Failed deletions are reported in the task output.
-
-4. **Public Instance Only**: The automatic cleanup task only runs when `PUBLIC_MODE=true` to prevent accidental data loss in private deployments.
-
-## Testing
-
-Run the cleanup tests:
-
-```bash
-uv run pytest tests/test_cleanup.py -v
-```
-
-## Monitoring
-
-Check Celery logs for cleanup task execution:
-
-```bash
-# Look for cleanup task logs
-grep "cleanup_old_public_data" celery.log
-grep "Starting cleanup of old public data" celery.log
-```
-
-Task statistics are logged after each run:
- Number of transcripts deleted
- Number of meetings deleted
- Number of orphaned recordings deleted
- Any errors encountered
--- a/server/docs/gpu/api-transcription.md
+++ b/server/docs/gpu/api-transcription.md
@@ -1,194 +0,0 @@
-## Reflector GPU Transcription API (Specification)
-
-This document defines the Reflector GPU transcription API that all implementations must adhere to. Current implementations include NVIDIA Parakeet (NeMo) and Whisper (faster-whisper), both deployed on Modal.com. The API surface and response shapes are OpenAI/Whisper-compatible, so clients can switch implementations by changing only the base URL.
-
-### Base URL and Authentication
-
- Example base URLs (Modal web endpoints):
-
-  - Parakeet: `https://<account>--reflector-transcriber-parakeet-web.modal.run`
-  - Whisper: `https://<account>--reflector-transcriber-web.modal.run`
-
- All endpoints are served under `/v1` and require a Bearer token:
-
-```
-Authorization: Bearer <REFLECTOR_GPU_APIKEY>
-```
-
-Note: To switch implementations, deploy the desired variant and point `TRANSCRIPT_URL` to its base URL. The API is identical.
-
-### Supported file types
-
-`mp3, mp4, mpeg, mpga, m4a, wav, webm`
-
-### Models and languages
-
- Parakeet (NVIDIA NeMo): default `nvidia/parakeet-tdt-0.6b-v2`
-  - Language support: only `en`. Other languages return HTTP 400.
- Whisper (faster-whisper): default `large-v2` (or deployment-specific)
-  - Language support: multilingual (per Whisper model capabilities).
-
-Note: The `model` parameter is accepted by all implementations for interface parity. Some backends may treat it as informational.
-
-### Endpoints
-
-#### POST /v1/audio/transcriptions
-
-Transcribe one or more uploaded audio files.
-
-Request: multipart/form-data
-
- `file` (File) — optional. Single file to transcribe.
- `files` (File[]) — optional. One or more files to transcribe.
- `model` (string) — optional. Defaults to the implementation-specific model (see above).
- `language` (string) — optional, defaults to `en`.
-  - Parakeet: only `en` is accepted; other values return HTTP 400
-  - Whisper: model-dependent; typically multilingual
- `batch` (boolean) — optional, defaults to `false`.
-
-Notes:
-
- Provide either `file` or `files`, not both. If neither is provided, HTTP 400.
- `batch` requires `files`; using `batch=true` without `files` returns HTTP 400.
- Response shape for multiple files is the same regardless of `batch`.
- Files sent to this endpoint are processed in a single pass (no VAD/chunking). This is intended for short clips (roughly ≤ 30s; depends on GPU memory/model). For longer audio, prefer `/v1/audio/transcriptions-from-url` which supports VAD-based chunking.
-
-Responses
-
-Single file response:
-
-```json
-{
-  "text": "transcribed text",
-  "words": [
-    { "word": "hello", "start": 0.0, "end": 0.5 },
-    { "word": "world", "start": 0.5, "end": 1.0 }
-  ],
-  "filename": "audio.mp3"
-}
-```
-
-Multiple files response:
-
-```json
-{
-  "results": [
-    {"filename": "a1.mp3", "text": "...", "words": [...]},
-    {"filename": "a2.mp3", "text": "...", "words": [...]}]
-}
-```
-
-Notes:
-
- Word objects always include keys: `word`, `start`, `end`.
- Some implementations may include a trailing space in `word` to match Whisper tokenization behavior; clients should trim if needed.
-
-Example curl (single file):
-
-```bash
-curl -X POST \
-  -H "Authorization: Bearer $REFLECTOR_GPU_APIKEY" \
-  -F "file=@/path/to/audio.mp3" \
-  -F "language=en" \
-  "$BASE_URL/v1/audio/transcriptions"
-```
-
-Example curl (multiple files, batch):
-
-```bash
-curl -X POST \
-  -H "Authorization: Bearer $REFLECTOR_GPU_APIKEY" \
-  -F "files=@/path/a1.mp3" -F "files=@/path/a2.mp3" \
-  -F "batch=true" -F "language=en" \
-  "$BASE_URL/v1/audio/transcriptions"
-```
-
-#### POST /v1/audio/transcriptions-from-url
-
-Transcribe a single remote audio file by URL.
-
-Request: application/json
-
-Body parameters:
-
- `audio_file_url` (string) — required. URL of the audio file to transcribe.
- `model` (string) — optional. Defaults to the implementation-specific model (see above).
- `language` (string) — optional, defaults to `en`. Parakeet only accepts `en`.
- `timestamp_offset` (number) — optional, defaults to `0.0`. Added to each word's `start`/`end` in the response.
-
-```json
-{
-  "audio_file_url": "https://example.com/audio.mp3",
-  "model": "nvidia/parakeet-tdt-0.6b-v2",
-  "language": "en",
-  "timestamp_offset": 0.0
-}
-```
-
-Response:
-
-```json
-{
-  "text": "transcribed text",
-  "words": [
-    { "word": "hello", "start": 10.0, "end": 10.5 },
-    { "word": "world", "start": 10.5, "end": 11.0 }
-  ]
-}
-```
-
-Notes:
-
- `timestamp_offset` is added to each word’s `start`/`end` in the response.
- Implementations may perform VAD-based chunking and batching for long-form audio; word timings are adjusted accordingly.
-
-Example curl:
-
-```bash
-curl -X POST \
-  -H "Authorization: Bearer $REFLECTOR_GPU_APIKEY" \
-  -H "Content-Type: application/json" \
-  -d '{
-        "audio_file_url": "https://example.com/audio.mp3",
-        "language": "en",
-        "timestamp_offset": 0
-      }' \
-  "$BASE_URL/v1/audio/transcriptions-from-url"
-```
-
-### Error handling
-
- 400 Bad Request
-  - Parakeet: `language` other than `en`
-  - Missing required parameters (`file`/`files` for upload; `audio_file_url` for URL endpoint)
-  - Unsupported file extension
- 401 Unauthorized
-  - Missing or invalid Bearer token
- 404 Not Found
-  - `audio_file_url` does not exist
-
-### Implementation details
-
- GPUs: A10G for small-file/live, L40S for large-file URL transcription (subject to deployment)
- VAD chunking and segment batching; word timings adjusted and overlapping ends constrained
- Pads very short segments (< 0.5s) to avoid model crashes on some backends
-
-### Server configuration (Reflector API)
-
-Set the Reflector server to use the Modal backend and point `TRANSCRIPT_URL` to your chosen deployment:
-
-```
-TRANSCRIPT_BACKEND=modal
-TRANSCRIPT_URL=https://<account>--reflector-transcriber-parakeet-web.modal.run
-TRANSCRIPT_MODAL_API_KEY=<REFLECTOR_GPU_APIKEY>
-```
-
-### Conformance tests
-
-Use the pytest-based conformance tests to validate any new implementation (including self-hosted) against this spec:
-
-```
-TRANSCRIPT_URL=https://<your-deployment-base> \
-TRANSCRIPT_MODAL_API_KEY=your-api-key \
-uv run -m pytest -m gpu_modal --no-cov server/tests/test_gpu_modal_transcript.py
-```
--- a/server/docs/platform-jitsi.md
+++ b/server/docs/platform-jitsi.md
@@ -1,493 +0,0 @@
-# Jitsi Integration Configuration Guide
-
-This guide provides step-by-step instructions for configuring Reflector to work with a self-hosted Jitsi Meet installation for video meetings and recording.
-
-## Prerequisites
-
-Before configuring Jitsi integration, ensure you have:
-
- **Self-hosted Jitsi Meet installation** (version 2.0.8922 or later recommended)
- **Jibri recording service** configured and running
- **Prosody XMPP server** with mod_event_sync module installed
- **Docker or system deployment** of Reflector with access to environment variables
- **SSL certificates** for secure communication between services
-
-## Environment Configuration
-
-Add the following environment variables to your Reflector deployment:
-
-### Required Settings
-
-```bash
-# Jitsi Meet domain (without https://)
-JITSI_DOMAIN=meet.example.com
-
-# JWT secret for room authentication (generate with: openssl rand -hex 32)
-JITSI_JWT_SECRET=your-64-character-hex-secret-here
-
-# Webhook secret for secure event handling (generate with: openssl rand -hex 16)
-JITSI_WEBHOOK_SECRET=your-32-character-hex-secret-here
-
-# Application identifier (should match Jitsi configuration)
-JITSI_APP_ID=reflector
-
-# JWT issuer and audience (should match Jitsi configuration)
-JITSI_JWT_ISSUER=reflector
-JITSI_JWT_AUDIENCE=jitsi
-```
-
-### Example .env Configuration
-
-```bash
-# Add to your server/.env file
-JITSI_DOMAIN=meet.mycompany.com
-JITSI_JWT_SECRET=$(openssl rand -hex 32)
-JITSI_WEBHOOK_SECRET=$(openssl rand -hex 16)
-JITSI_APP_ID=reflector
-JITSI_JWT_ISSUER=reflector
-JITSI_JWT_AUDIENCE=jitsi
-```
-
-## Jitsi Meet Server Configuration
-
-### 1. JWT Authentication Setup
-
-Edit `/etc/prosody/conf.d/[YOUR_DOMAIN].cfg.lua`:
-
-```lua
-VirtualHost "meet.example.com"
-    authentication = "token"
-    app_id = "reflector"
-    app_secret = "your-jwt-secret-here"
-
-    -- Allow anonymous access for non-authenticated users
-    c2s_require_encryption = false
-    admins = { "focusUser@auth.meet.example.com" }
-
-    modules_enabled = {
-        "bosh";
-        "pubsub";
-        "ping";
-        "roster";
-        "saslauth";
-        "tls";
-        "dialback";
-        "disco";
-        "carbons";
-        "pep";
-        "private";
-        "blocklist";
-        "vcard";
-        "version";
-        "uptime";
-        "time";
-        "ping";
-        "register";
-        "admin_adhoc";
-        "token_verification";
-        "event_sync";  -- Required for webhook events
-    }
-```
-
-### 2. Room Access Control
-
-Edit `/etc/jitsi/meet/meet.example.com-config.js`:
-
-```javascript
-var config = {
-    hosts: {
-        domain: 'meet.example.com',
-        muc: 'conference.meet.example.com'
-    },
-
-    // Enable JWT authentication
-    enableUserRolesBasedOnToken: true,
-
-    // Recording configuration
-    fileRecordingsEnabled: true,
-    liveStreamingEnabled: false,
-
-    // Reflector-specific settings
-    prejoinPageEnabled: true,
-    requireDisplayName: true,
-};
-```
-
-### 3. Interface Configuration
-
-Edit `/usr/share/jitsi-meet/interface_config.js`:
-
-```javascript
-var interfaceConfig = {
-    // Customize for Reflector branding
-    APP_NAME: 'Reflector Meeting',
-    DEFAULT_WELCOME_PAGE_LOGO_URL: 'https://your-domain.com/logo.png',
-
-    // Hide unnecessary buttons
-    TOOLBAR_BUTTONS: [
-        'microphone', 'camera', 'closedcaptions', 'desktop',
-        'fullscreen', 'fodeviceselection', 'hangup',
-        'chat', 'recording', 'livestreaming', 'etherpad',
-        'sharedvideo', 'settings', 'raisehand', 'videoquality',
-        'filmstrip', 'invite', 'feedback', 'stats', 'shortcuts',
-        'tileview', 'videobackgroundblur', 'download', 'help',
-        'mute-everyone'
-    ]
-};
-```
-
-## Jibri Configuration
-
-### 1. Recording Service Setup
-
-Edit `/etc/jitsi/jibri/jibri.conf`:
-
-```hocon
-jibri {
-    recording {
-        recordings-directory = "/var/recordings"
-        finalize-script = "/opt/jitsi/jibri/finalize.sh"
-    }
-
-    api {
-        xmpp {
-            environments = [{
-                name = "prod environment"
-                xmpp-server-hosts = ["meet.example.com"]
-                xmpp-domain = "meet.example.com"
-
-                control-muc {
-                    domain = "internal.auth.meet.example.com"
-                    room-name = "JibriBrewery"
-                    nickname = "jibri-nickname"
-                }
-
-                control-login {
-                    domain = "auth.meet.example.com"
-                    username = "jibri"
-                    password = "jibri-password"
-                }
-            }]
-        }
-    }
-}
-```
-
-### 2. Finalize Script Setup
-
-Create `/opt/jitsi/jibri/finalize.sh`:
-
-```bash
-#!/bin/bash
-# Jibri finalize script for Reflector integration
-
-RECORDING_FILE="$1"
-ROOM_NAME="$2"
-REFLECTOR_API_URL="${REFLECTOR_API_URL:-http://localhost:1250}"
-WEBHOOK_SECRET="${JITSI_WEBHOOK_SECRET}"
-
-# Generate webhook signature
-generate_signature() {
-    local payload="$1"
-    echo -n "$payload" | openssl dgst -sha256 -hmac "$WEBHOOK_SECRET" | cut -d' ' -f2
-}
-
-# Prepare webhook payload
-TIMESTAMP=$(date -u +%Y-%m-%dT%H:%M:%S.%3NZ)
-PAYLOAD=$(cat <<EOF
-{
-    "room_name": "$ROOM_NAME",
-    "recording_file": "$RECORDING_FILE",
-    "recording_status": "completed",
-    "timestamp": "$TIMESTAMP"
-}
-EOF
-)
-
-# Generate signature
-SIGNATURE=$(generate_signature "$PAYLOAD")
-
-# Send webhook to Reflector
-curl -X POST "$REFLECTOR_API_URL/v1/jibri/recording-complete" \
-    -H "Content-Type: application/json" \
-    -H "X-Jitsi-Signature: $SIGNATURE" \
-    -d "$PAYLOAD" \
-    --max-time 30
-
-echo "Recording finalization webhook sent for room: $ROOM_NAME"
-```
-
-Make the script executable:
-
-```bash
-chmod +x /opt/jitsi/jibri/finalize.sh
-```
-
-## Prosody Event Configuration
-
-### 1. Event-Sync Module Installation
-
-Install the mod_event_sync module:
-
-```bash
-# Download the module
-cd /usr/share/jitsi-meet/prosody-plugins/
-wget https://raw.githubusercontent.com/jitsi-contrib/prosody-plugins/main/mod_event_sync.lua
-
-# Or if using git
-git clone https://github.com/jitsi-contrib/prosody-plugins.git
-cp prosody-plugins/mod_event_sync.lua /usr/share/jitsi-meet/prosody-plugins/
-```
-
-### 2. Webhook Configuration
-
-Add to `/etc/prosody/conf.d/[YOUR_DOMAIN].cfg.lua`:
-
-```lua
-Component "conference.meet.example.com" "muc"
-    storage = "memory"
-    modules_enabled = {
-        "muc_meeting_id";
-        "muc_domain_mapper";
-        "polls";
-        "event_sync";  -- Enable event sync
-    }
-
-    -- Event sync webhook configuration
-    event_sync_url = "https://your-reflector-domain.com/v1/jitsi/events"
-    event_sync_secret = "your-webhook-secret-here"
-
-    -- Events to track
-    event_sync_events = {
-        "muc-occupant-joined",
-        "muc-occupant-left",
-        "jibri-recording-on",
-        "jibri-recording-off"
-    }
-```
-
-### 3. Restart Services
-
-After configuration changes, restart all services:
-
-```bash
-systemctl restart prosody
-systemctl restart jicofo
-systemctl restart jitsi-videobridge2
-systemctl restart jibri
-systemctl restart nginx
-```
-
-## Reflector Room Configuration
-
-### 1. Create Jitsi Room
-
-When creating rooms in Reflector, set the platform field:
-
-```bash
-curl -X POST "https://your-reflector-domain.com/v1/rooms" \
-    -H "Authorization: Bearer $AUTH_TOKEN" \
-    -H "Content-Type: application/json" \
-    -d '{
-        "name": "my-jitsi-room",
-        "platform": "jitsi",
-        "recording_type": "cloud",
-        "recording_trigger": "automatic-2nd-participant",
-        "is_locked": false,
-        "room_mode": "normal"
-    }'
-```
-
-### 2. Meeting Creation
-
-Meetings will automatically use Jitsi when the room platform is set to "jitsi":
-
-```bash
-curl -X POST "https://your-reflector-domain.com/v1/rooms/my-jitsi-room/meeting" \
-    -H "Authorization: Bearer $AUTH_TOKEN"
-```
-
-## Testing the Integration
-
-### 1. Health Check
-
-Verify Jitsi webhook configuration:
-
-```bash
-curl "https://your-reflector-domain.com/v1/jitsi/health"
-```
-
-Expected response:
-```json
-{
-    "status": "ok",
-    "service": "jitsi-webhooks",
-    "timestamp": "2025-01-15T10:30:00.000Z",
-    "webhook_secret_configured": true
-}
-```
-
-### 2. Room Creation Test
-
-1. Create a Jitsi room via Reflector API
-2. Start a meeting - should generate Jitsi Meet URL with JWT token
-3. Join with multiple participants - should trigger participant events
-4. Start recording - should trigger Jibri recording workflow
-
-### 3. Webhook Event Test
-
-Monitor Reflector logs for incoming webhook events:
-
-```bash
-# Check for participant events
-curl -X POST "https://your-reflector-domain.com/v1/jitsi/events" \
-    -H "Content-Type: application/json" \
-    -H "X-Jitsi-Signature: test-signature" \
-    -d '{
-        "event": "muc-occupant-joined",
-        "room": "test-room-name",
-        "timestamp": "2025-01-15T10:30:00.000Z",
-        "data": {}
-    }'
-```
-
-## Troubleshooting
-
-### Common Issues
-
-#### JWT Authentication Failures
-
-**Symptoms:** Users can't join rooms, "Authentication failed" errors
-
-**Solutions:**
-1. Verify JWT secret matches between Jitsi and Reflector
-2. Check JWT token expiration (default 8 hours)
-3. Ensure system clocks are synchronized
-4. Validate JWT issuer/audience configuration
-
-```bash
-# Debug JWT tokens
-echo "JWT_TOKEN_HERE" | cut -d'.' -f2 | base64 -d | jq
-```
-
-#### Webhook Events Not Received
-
-**Symptoms:** Participant counts not updating, recording events missing
-
-**Solutions:**
-1. Verify event_sync module is loaded in Prosody
-2. Check webhook URL accessibility from Jitsi server
-3. Validate webhook signature generation
-4. Review Prosody and Reflector logs
-
-```bash
-# Test webhook connectivity
-curl -v "https://your-reflector-domain.com/v1/jitsi/health"
-
-# Check Prosody logs
-tail -f /var/log/prosody/prosody.log
-
-# Check Reflector logs
-docker logs your-reflector-container
-```
-
-#### Recording Issues
-
-**Symptoms:** Recordings not starting, finalize script errors
-
-**Solutions:**
-1. Verify Jibri service status and configuration
-2. Check recording directory permissions
-3. Validate finalize script execution permissions
-4. Monitor Jibri logs for errors
-
-```bash
-# Check Jibri status
-systemctl status jibri
-
-# Test finalize script
-sudo -u jibri /opt/jitsi/jibri/finalize.sh "/test/recording.mp4" "test-room"
-
-# Check Jibri logs
-journalctl -u jibri -f
-```
-
-### Debug Commands
-
-```bash
-# Verify Jitsi configuration
-prosodyctl check config
-
-# Test JWT generation
-curl -X POST "https://your-reflector-domain.com/v1/rooms/test/meeting" \
-    -H "Authorization: Bearer $TOKEN" -v
-
-# Monitor webhook events
-tail -f /var/log/reflector/app.log | grep jitsi
-
-# Check room participant counts
-curl "https://your-reflector-domain.com/v1/rooms" \
-    -H "Authorization: Bearer $TOKEN" | jq '.data[].num_clients'
-```
-
-### Performance Optimization
-
-#### For High-Concurrent Usage
-
-1. **Jitsi Videobridge Tuning:**
-```bash
-# /etc/jitsi/videobridge/sip-communicator.properties
-org.jitsi.videobridge.STATISTICS_INTERVAL=5000
-org.jitsi.videobridge.load.INITIAL_STREAM_LIMIT=50
-```
-
-2. **Database Connection Pooling:**
-```python
-# In your Reflector settings
-DATABASE_POOL_SIZE=20
-DATABASE_MAX_OVERFLOW=30
-```
-
-3. **Redis Configuration:**
-```bash
-# For webhook event caching
-REDIS_URL=redis://localhost:6379/1
-WEBHOOK_EVENT_TTL=3600
-```
-
-## Security Considerations
-
-### Network Security
- Use HTTPS/WSS for all communications
- Implement proper firewall rules
- Consider VPN for server-to-server communication
-
-### Authentication Security
- Rotate JWT secrets regularly
- Use strong webhook secrets (32+ characters)
- Implement rate limiting on webhook endpoints
-
-### Recording Security
- Encrypt recordings at rest
- Implement access controls for recording files
- Regular security audits of file permissions
-
-## Support
-
-For additional support:
-
-1. **Reflector Issues:** Check GitHub issues or create new ones
-2. **Jitsi Community:** [Community Forum](https://community.jitsi.org/)
-3. **Documentation:** [Jitsi Developer Guide](https://jitsi.github.io/handbook/)
-
-## Migration from Whereby
-
-If migrating from Whereby integration:
-
-1. Update existing rooms to use "jitsi" platform
-2. Verify webhook configurations are updated
-3. Test recording workflows thoroughly
-4. Monitor participant event accuracy
-5. Update any custom integrations using meeting APIs
-
-The platform abstraction layer ensures smooth migration with minimal API changes.
--- a/server/docs/webhook.md
+++ b/server/docs/webhook.md
@@ -1,212 +0,0 @@
-# Reflector Webhook Documentation
-
-## Overview
-
-Reflector supports webhook notifications to notify external systems when transcript processing is completed. Webhooks can be configured per room and are triggered automatically after a transcript is successfully processed.
-
-## Configuration
-
-Webhooks are configured at the room level with two fields:
- `webhook_url`: The HTTPS endpoint to receive webhook notifications
- `webhook_secret`: Optional secret key for HMAC signature verification (auto-generated if not provided)
-
-## Events
-
-### `transcript.completed`
-
-Triggered when a transcript has been fully processed, including transcription, diarization, summarization, and topic detection.
-
-### `test`
-
-A test event that can be triggered manually to verify webhook configuration.
-
-## Webhook Request Format
-
-### Headers
-
-All webhook requests include the following headers:
-
-| Header | Description | Example |
-|--------|-------------|---------|
-| `Content-Type` | Always `application/json` | `application/json` |
-| `User-Agent` | Identifies Reflector as the source | `Reflector-Webhook/1.0` |
-| `X-Webhook-Event` | The event type | `transcript.completed` or `test` |
-| `X-Webhook-Retry` | Current retry attempt number | `0`, `1`, `2`... |
-| `X-Webhook-Signature` | HMAC signature (if secret configured) | `t=1735306800,v1=abc123...` |
-
-### Signature Verification
-
-If a webhook secret is configured, Reflector includes an HMAC-SHA256 signature in the `X-Webhook-Signature` header to verify the webhook authenticity.
-
-The signature format is: `t={timestamp},v1={signature}`
-
-To verify the signature:
-1. Extract the timestamp and signature from the header
-2. Create the signed payload: `{timestamp}.{request_body}`
-3. Compute HMAC-SHA256 of the signed payload using your webhook secret
-4. Compare the computed signature with the received signature
-
-Example verification (Python):
-```python
-import hmac
-import hashlib
-
-def verify_webhook_signature(payload: bytes, signature_header: str, secret: str) -> bool:
-    # Parse header: "t=1735306800,v1=abc123..."
-    parts = dict(part.split("=") for part in signature_header.split(","))
-    timestamp = parts["t"]
-    received_signature = parts["v1"]
-
-    # Create signed payload
-    signed_payload = f"{timestamp}.{payload.decode('utf-8')}"
-
-    # Compute expected signature
-    expected_signature = hmac.new(
-        secret.encode("utf-8"),
-        signed_payload.encode("utf-8"),
-        hashlib.sha256
-    ).hexdigest()
-
-    # Compare signatures
-    return hmac.compare_digest(expected_signature, received_signature)
-```
-
-## Event Payloads
-
-### `transcript.completed` Event
-
-This event includes a convenient URL for accessing the transcript:
- `frontend_url`: Direct link to view the transcript in the web interface
-
-```json
-{
-  "event": "transcript.completed",
-  "event_id": "transcript.completed-abc-123-def-456",
-  "timestamp": "2025-08-27T12:34:56.789012Z",
-  "transcript": {
-    "id": "abc-123-def-456",
-    "room_id": "room-789",
-    "created_at": "2025-08-27T12:00:00Z",
-    "duration": 1800.5,
-    "title": "Q3 Product Planning Meeting",
-    "short_summary": "Team discussed Q3 product roadmap, prioritizing mobile app features and API improvements.",
-    "long_summary": "The product team met to finalize the Q3 roadmap. Key decisions included...",
-    "webvtt": "WEBVTT\n\n00:00:00.000 --> 00:00:05.000\n<v Speaker 1>Welcome everyone to today's meeting...",
-    "topics": [
-      {
-        "title": "Introduction and Agenda",
-        "summary": "Meeting kickoff with agenda review",
-        "timestamp": 0.0,
-        "duration": 120.0,
-        "webvtt": "WEBVTT\n\n00:00:00.000 --> 00:00:05.000\n<v Speaker 1>Welcome everyone..."
-      },
-      {
-        "title": "Mobile App Features Discussion",
-        "summary": "Team reviewed proposed mobile app features for Q3",
-        "timestamp": 120.0,
-        "duration": 600.0,
-        "webvtt": "WEBVTT\n\n00:02:00.000 --> 00:02:10.000\n<v Speaker 2>Let's talk about the mobile app..."
-      }
-    ],
-    "participants": [
-      {
-        "id": "participant-1",
-        "name": "John Doe",
-        "speaker": "Speaker 1"
-      },
-      {
-        "id": "participant-2",
-        "name": "Jane Smith",
-        "speaker": "Speaker 2"
-      }
-    ],
-    "source_language": "en",
-    "target_language": "en",
-    "status": "completed",
-    "frontend_url": "https://app.reflector.com/transcripts/abc-123-def-456"
-  },
-  "room": {
-    "id": "room-789",
-    "name": "Product Team Room"
-  }
-}
-```
-
-### `test` Event
-
-```json
-{
-  "event": "test",
-  "event_id": "test.2025-08-27T12:34:56.789012Z",
-  "timestamp": "2025-08-27T12:34:56.789012Z",
-  "message": "This is a test webhook from Reflector",
-  "room": {
-    "id": "room-789",
-    "name": "Product Team Room"
-  }
-}
-```
-
-## Retry Policy
-
-Webhooks are delivered with automatic retry logic to handle transient failures. When a webhook delivery fails due to server errors or network issues, Reflector will automatically retry the delivery multiple times over an extended period.
-
-### Retry Mechanism
-
-Reflector implements an exponential backoff strategy for webhook retries:
-
- **Initial retry delay**: 60 seconds after the first failure
- **Exponential backoff**: Each subsequent retry waits approximately twice as long as the previous one
- **Maximum retry interval**: 1 hour (backoff is capped at this duration)
- **Maximum retry attempts**: 30 attempts total
- **Total retry duration**: Retries continue for approximately 24 hours
-
-### How Retries Work
-
-When a webhook fails, Reflector will:
-1. Wait 60 seconds, then retry (attempt #1)
-2. If it fails again, wait ~2 minutes, then retry (attempt #2)
-3. Continue doubling the wait time up to a maximum of 1 hour between attempts
-4. Keep retrying at 1-hour intervals until successful or 30 attempts are exhausted
-
-The `X-Webhook-Retry` header indicates the current retry attempt number (0 for the initial attempt, 1 for first retry, etc.), allowing your endpoint to track retry attempts.
-
-### Retry Behavior by HTTP Status Code
-
-| Status Code | Behavior |
-|-------------|----------|
-| 2xx (Success) | No retry, webhook marked as delivered |
-| 4xx (Client Error) | No retry, request is considered permanently failed |
-| 5xx (Server Error) | Automatic retry with exponential backoff |
-| Network/Timeout Error | Automatic retry with exponential backoff |
-
-**Important Notes:**
- Webhooks timeout after 30 seconds. If your endpoint takes longer to respond, it will be considered a timeout error and retried.
- During the retry period (~24 hours), you may receive the same webhook multiple times if your endpoint experiences intermittent failures.
- There is no mechanism to manually retry failed webhooks after the retry period expires.
-
-## Testing Webhooks
-
-You can test your webhook configuration before processing transcripts:
-
-```http
-POST /v1/rooms/{room_id}/webhook/test
-```
-
-Response:
-```json
-{
-  "success": true,
-  "status_code": 200,
-  "message": "Webhook test successful",
-  "response_preview": "OK"
-}
-```
-
-Or in case of failure:
-```json
-{
-  "success": false,
-  "error": "Webhook request timed out (10 seconds)"
-}
-```
--- a/server/gpu/modal_deployments/README.md
+++ b/server/gpu/modal_deployments/README.md
@@ -4,8 +4,7 @@ This repository hold an API for the GPU implementation of the Reflector API serv
 and use [Modal.com](https://modal.com)

 - `reflector_diarizer.py` - Diarization API
- `reflector_transcriber.py` - Transcription API (Whisper)
- `reflector_transcriber_parakeet.py` - Transcription API (NVIDIA Parakeet)
+- `reflector_transcriber.py` - Transcription API
 - `reflector_translator.py` - Translation API

 ## Modal.com deployment
@@ -20,10 +19,6 @@ $ modal deploy reflector_transcriber.py
 ...
 └── 🔨 Created web => https://xxxx--reflector-transcriber-web.modal.run

-$ modal deploy reflector_transcriber_parakeet.py
-...
-└── 🔨 Created web => https://xxxx--reflector-transcriber-parakeet-web.modal.run
-
 $ modal deploy reflector_llm.py
 ...
 └── 🔨 Created web => https://xxxx--reflector-llm-web.modal.run
@@ -73,86 +68,6 @@ Authorization: bearer <REFLECTOR_APIKEY>

 ### Transcription

-#### Parakeet Transcriber (`reflector_transcriber_parakeet.py`)
-
-NVIDIA Parakeet is a state-of-the-art ASR model optimized for real-time transcription with superior word-level timestamps.
-
-**GPU Configuration:**
- **A10G GPU** - Used for `/v1/audio/transcriptions` endpoint (small files, live transcription)
-  - Higher concurrency (max_inputs=10)
-  - Optimized for multiple small audio files
-  - Supports batch processing for efficiency
-
- **L40S GPU** - Used for `/v1/audio/transcriptions-from-url` endpoint (large files)
-  - Lower concurrency but more powerful processing
-  - Optimized for single large audio files
-  - VAD-based chunking for long-form audio
-
-##### `/v1/audio/transcriptions` - Small file transcription
-
-**request** (multipart/form-data)
- `file` or `files[]` - audio file(s) to transcribe
- `model` - model name (default: `nvidia/parakeet-tdt-0.6b-v2`)
- `language` - language code (default: `en`)
- `batch` - whether to use batch processing for multiple files (default: `true`)
-
-**response**
-```json
-{
-    "text": "transcribed text",
-    "words": [
-        {"word": "hello", "start": 0.0, "end": 0.5},
-        {"word": "world", "start": 0.5, "end": 1.0}
-    ],
-    "filename": "audio.mp3"
-}
-```
-
-For multiple files with batch=true:
-```json
-{
-    "results": [
-        {
-            "filename": "audio1.mp3",
-            "text": "transcribed text",
-            "words": [...]
-        },
-        {
-            "filename": "audio2.mp3",
-            "text": "transcribed text",
-            "words": [...]
-        }
-    ]
-}
-```
-
-##### `/v1/audio/transcriptions-from-url` - Large file transcription
-
-**request** (application/json)
-```json
-{
-    "audio_file_url": "https://example.com/audio.mp3",
-    "model": "nvidia/parakeet-tdt-0.6b-v2",
-    "language": "en",
-    "timestamp_offset": 0.0
-}
-```
-
-**response**
-```json
-{
-    "text": "transcribed text from large file",
-    "words": [
-        {"word": "hello", "start": 0.0, "end": 0.5},
-        {"word": "world", "start": 0.5, "end": 1.0}
-    ]
-}
-```
-
-**Supported file types:** mp3, mp4, mpeg, mpga, m4a, wav, webm
-
-#### Whisper Transcriber (`reflector_transcriber.py`)
-
 `POST /transcribe`

 **request** (multipart/form-data)
--- a/server/gpu/modal_deployments/reflector_diarizer.py
+++ b/server/gpu/modal_deployments/reflector_diarizer.py
@@ -4,80 +4,14 @@ Reflector GPU backend - diarizer
 """

 import os
-import uuid
-from typing import Mapping, NewType
-from urllib.parse import urlparse

-import modal
+import modal.gpu
+from modal import App, Image, Secret, asgi_app, enter, method
+from pydantic import BaseModel

 PYANNOTE_MODEL_NAME: str = "pyannote/speaker-diarization-3.1"
 MODEL_DIR = "/root/diarization_models"
-UPLOADS_PATH = "/uploads"
-SUPPORTED_FILE_EXTENSIONS = ["mp3", "mp4", "mpeg", "mpga", "m4a", "wav", "webm"]
-
-DiarizerUniqFilename = NewType("DiarizerUniqFilename", str)
-AudioFileExtension = NewType("AudioFileExtension", str)
-
-app = modal.App(name="reflector-diarizer")
-
-# Volume for temporary file uploads
-upload_volume = modal.Volume.from_name("diarizer-uploads", create_if_missing=True)
-
-
-def detect_audio_format(url: str, headers: Mapping[str, str]) -> AudioFileExtension:
-    parsed_url = urlparse(url)
-    url_path = parsed_url.path
-
-    for ext in SUPPORTED_FILE_EXTENSIONS:
-        if url_path.lower().endswith(f".{ext}"):
-            return AudioFileExtension(ext)
-
-    content_type = headers.get("content-type", "").lower()
-    if "audio/mpeg" in content_type or "audio/mp3" in content_type:
-        return AudioFileExtension("mp3")
-    if "audio/wav" in content_type:
-        return AudioFileExtension("wav")
-    if "audio/mp4" in content_type:
-        return AudioFileExtension("mp4")
-
-    raise ValueError(
-        f"Unsupported audio format for URL: {url}. "
-        f"Supported extensions: {', '.join(SUPPORTED_FILE_EXTENSIONS)}"
-    )
-
-
-def download_audio_to_volume(
-    audio_file_url: str,
-) -> tuple[DiarizerUniqFilename, AudioFileExtension]:
-    import requests
-    from fastapi import HTTPException
-
-    print(f"Checking audio file at: {audio_file_url}")
-    response = requests.head(audio_file_url, allow_redirects=True)
-    if response.status_code == 404:
-        raise HTTPException(status_code=404, detail="Audio file not found")
-
-    print(f"Downloading audio file from: {audio_file_url}")
-    response = requests.get(audio_file_url, allow_redirects=True)
-
-    if response.status_code != 200:
-        print(f"Download failed with status {response.status_code}: {response.text}")
-        raise HTTPException(
-            status_code=response.status_code,
-            detail=f"Failed to download audio file: {response.status_code}",
-        )
-
-    audio_suffix = detect_audio_format(audio_file_url, response.headers)
-    unique_filename = DiarizerUniqFilename(f"{uuid.uuid4()}.{audio_suffix}")
-    file_path = f"{UPLOADS_PATH}/{unique_filename}"
-
-    print(f"Writing file to: {file_path} (size: {len(response.content)} bytes)")
-    with open(file_path, "wb") as f:
-        f.write(response.content)
-
-    upload_volume.commit()
-    print(f"File saved as: {unique_filename}")
-    return unique_filename, audio_suffix
+app = App(name="reflector-diarizer")


 def migrate_cache_llm():
@@ -105,7 +39,7 @@ def download_pyannote_audio():


 diarizer_image = (
-    modal.Image.debian_slim(python_version="3.10.8")
+    Image.debian_slim(python_version="3.10.8")
    .pip_install(
        "pyannote.audio==3.1.0",
        "requests",
@@ -121,8 +55,7 @@ diarizer_image = (
        "hf-transfer",
    )
    .run_function(
-        download_pyannote_audio,
-        secrets=[modal.Secret.from_name("hf_token")],
+        download_pyannote_audio, secrets=[Secret.from_name("my-huggingface-secret")]
    )
    .run_function(migrate_cache_llm)
    .env(
@@ -137,60 +70,53 @@ diarizer_image = (


@app.cls(
-    gpu="A100",
+    gpu=modal.gpu.A100(size="40GB"),
    timeout=60 * 30,
+    scaledown_window=60,
+    allow_concurrent_inputs=1,
    image=diarizer_image,
-    volumes={UPLOADS_PATH: upload_volume},
-    enable_memory_snapshot=True,
-    experimental_options={"enable_gpu_snapshot": True},
-    secrets=[
-        modal.Secret.from_name("hf_token"),
-    ],
 )
-@modal.concurrent(max_inputs=1)
 class Diarizer:
-    @modal.enter(snap=True)
+    @enter()
    def enter(self):
        import torch
        from pyannote.audio import Pipeline

        self.use_gpu = torch.cuda.is_available()
        self.device = "cuda" if self.use_gpu else "cpu"
-        print(f"Using device: {self.device}")
        self.diarization_pipeline = Pipeline.from_pretrained(
-            PYANNOTE_MODEL_NAME,
-            cache_dir=MODEL_DIR,
-            use_auth_token=os.environ["HF_TOKEN"],
+            PYANNOTE_MODEL_NAME, cache_dir=MODEL_DIR
        )
        self.diarization_pipeline.to(torch.device(self.device))

-    @modal.method()
-    def diarize(self, filename: str, timestamp: float = 0.0):
+    @method()
+    def diarize(self, audio_data: str, audio_suffix: str, timestamp: float):
+        import tempfile
+
        import torchaudio

-        upload_volume.reload()
+        with tempfile.NamedTemporaryFile("wb+", suffix=f".{audio_suffix}") as fp:
+            fp.write(audio_data)

-        file_path = f"{UPLOADS_PATH}/{filename}"
-        if not os.path.exists(file_path):
-            raise FileNotFoundError(f"File not found: {file_path}")
-
-        print(f"Diarizing audio from: {file_path}")
-        waveform, sample_rate = torchaudio.load(file_path)
-        diarization = self.diarization_pipeline(
-            {"waveform": waveform, "sample_rate": sample_rate}
-        )
-
-        words = []
-        for diarization_segment, _, speaker in diarization.itertracks(yield_label=True):
-            words.append(
-                {
-                    "start": round(timestamp + diarization_segment.start, 3),
-                    "end": round(timestamp + diarization_segment.end, 3),
-                    "speaker": int(speaker[-2:]),
-                }
+            print("Diarizing audio")
+            waveform, sample_rate = torchaudio.load(fp.name)
+            diarization = self.diarization_pipeline(
+                {"waveform": waveform, "sample_rate": sample_rate}
            )
-        print("Diarization complete")
-        return {"diarization": words}
+
+            words = []
+            for diarization_segment, _, speaker in diarization.itertracks(
+                yield_label=True
+            ):
+                words.append(
+                    {
+                        "start": round(timestamp + diarization_segment.start, 3),
+                        "end": round(timestamp + diarization_segment.end, 3),
+                        "speaker": int(speaker[-2:]),
+                    }
+                )
+            print("Diarization complete")
+            return {"diarization": words}


 # -------------------------------------------------------------------
@@ -201,18 +127,17 @@ class Diarizer:
@app.function(
    timeout=60 * 10,
    scaledown_window=60 * 3,
+    allow_concurrent_inputs=40,
    secrets=[
-        modal.Secret.from_name("reflector-gpu"),
+        Secret.from_name("reflector-gpu"),
    ],
-    volumes={UPLOADS_PATH: upload_volume},
    image=diarizer_image,
 )
-@modal.concurrent(max_inputs=40)
-@modal.asgi_app()
+@asgi_app()
 def web():
+    import requests
    from fastapi import Depends, FastAPI, HTTPException, status
    from fastapi.security import OAuth2PasswordBearer
-    from pydantic import BaseModel

    diarizerstub = Diarizer()

@@ -228,26 +153,35 @@ def web():
                headers={"WWW-Authenticate": "Bearer"},
            )

+    def validate_audio_file(audio_file_url: str):
+        # Check if the audio file exists
+        response = requests.head(audio_file_url, allow_redirects=True)
+        if response.status_code == 404:
+            raise HTTPException(
+                status_code=response.status_code,
+                detail="The audio file does not exist.",
+            )
+
    class DiarizationResponse(BaseModel):
        result: dict

-    @app.post("/diarize", dependencies=[Depends(apikey_auth)])
-    def diarize(audio_file_url: str, timestamp: float = 0.0) -> DiarizationResponse:
-        unique_filename, audio_suffix = download_audio_to_volume(audio_file_url)
+    @app.post(
+        "/diarize", dependencies=[Depends(apikey_auth), Depends(validate_audio_file)]
+    )
+    def diarize(
+        audio_file_url: str, timestamp: float = 0.0
+    ) -> HTTPException | DiarizationResponse:
+        # Currently the uploaded files are in mp3 format
+        audio_suffix = "mp3"

-        try:
-            func = diarizerstub.diarize.spawn(
-                filename=unique_filename, timestamp=timestamp
-            )
-            result = func.get()
-            return result
-        finally:
-            try:
-                file_path = f"{UPLOADS_PATH}/{unique_filename}"
-                print(f"Deleting file: {file_path}")
-                os.remove(file_path)
-                upload_volume.commit()
-            except Exception as e:
-                print(f"Error cleaning up {unique_filename}: {e}")
+        print("Downloading audio file")
+        response = requests.get(audio_file_url, allow_redirects=True)
+        print("Audio file downloaded successfully")
+
+        func = diarizerstub.diarize.spawn(
+            audio_data=response.content, audio_suffix=audio_suffix, timestamp=timestamp
+        )
+        result = func.get()
+        return result

    return app
--- a/server/gpu/modal_deployments/reflector_transcriber.py
+++ b/server/gpu/modal_deployments/reflector_transcriber.py
@@ -1,78 +1,41 @@
 import os
-import sys
+import tempfile
 import threading
-import uuid
-from typing import Generator, Mapping, NamedTuple, NewType, TypedDict
-from urllib.parse import urlparse

 import modal
+from pydantic import BaseModel
+
+MODELS_DIR = "/models"

 MODEL_NAME = "large-v2"
 MODEL_COMPUTE_TYPE: str = "float16"
 MODEL_NUM_WORKERS: int = 1
+
 MINUTES = 60  # seconds
-SAMPLERATE = 16000
-UPLOADS_PATH = "/uploads"
-CACHE_PATH = "/models"
-SUPPORTED_FILE_EXTENSIONS = ["mp3", "mp4", "mpeg", "mpga", "m4a", "wav", "webm"]
-VAD_CONFIG = {
-    "batch_max_duration": 30.0,
-    "silence_padding": 0.5,
-    "window_size": 512,
-}

-
-WhisperUniqFilename = NewType("WhisperUniqFilename", str)
-AudioFileExtension = NewType("AudioFileExtension", str)
+volume = modal.Volume.from_name("models", create_if_missing=True)

 app = modal.App("reflector-transcriber")

-model_cache = modal.Volume.from_name("models", create_if_missing=True)
-upload_volume = modal.Volume.from_name("whisper-uploads", create_if_missing=True)
-
-
-class TimeSegment(NamedTuple):
-    """Represents a time segment with start and end times."""
-
-    start: float
-    end: float
-
-
-class AudioSegment(NamedTuple):
-    """Represents an audio segment with timing and audio data."""
-
-    start: float
-    end: float
-    audio: any
-
-
-class TranscriptResult(NamedTuple):
-    """Represents a transcription result with text and word timings."""
-
-    text: str
-    words: list["WordTiming"]
-
-
-class WordTiming(TypedDict):
-    """Represents a word with its timing information."""
-
-    word: str
-    start: float
-    end: float
-

 def download_model():
    from faster_whisper import download_model

-    model_cache.reload()
+    volume.reload()

-    download_model(MODEL_NAME, cache_dir=CACHE_PATH)
+    download_model(MODEL_NAME, cache_dir=MODELS_DIR)

-    model_cache.commit()
+    volume.commit()


 image = (
    modal.Image.debian_slim(python_version="3.12")
+    .pip_install(
+        "huggingface_hub==0.27.1",
+        "hf-transfer==0.1.9",
+        "torch==2.5.1",
+        "faster-whisper==1.1.1",
+    )
    .env(
        {
            "HF_HUB_ENABLE_HF_TRANSFER": "1",
@@ -82,98 +45,19 @@ image = (
            ),
        }
    )
-    .apt_install("ffmpeg")
-    .pip_install(
-        "huggingface_hub==0.27.1",
-        "hf-transfer==0.1.9",
-        "torch==2.5.1",
-        "faster-whisper==1.1.1",
-        "fastapi==0.115.12",
-        "requests",
-        "librosa==0.10.1",
-        "numpy<2",
-        "silero-vad==5.1.0",
-    )
-    .run_function(download_model, volumes={CACHE_PATH: model_cache})
+    .run_function(download_model, volumes={MODELS_DIR: volume})
 )


-def detect_audio_format(url: str, headers: Mapping[str, str]) -> AudioFileExtension:
-    parsed_url = urlparse(url)
-    url_path = parsed_url.path
-
-    for ext in SUPPORTED_FILE_EXTENSIONS:
-        if url_path.lower().endswith(f".{ext}"):
-            return AudioFileExtension(ext)
-
-    content_type = headers.get("content-type", "").lower()
-    if "audio/mpeg" in content_type or "audio/mp3" in content_type:
-        return AudioFileExtension("mp3")
-    if "audio/wav" in content_type:
-        return AudioFileExtension("wav")
-    if "audio/mp4" in content_type:
-        return AudioFileExtension("mp4")
-
-    raise ValueError(
-        f"Unsupported audio format for URL: {url}. "
-        f"Supported extensions: {', '.join(SUPPORTED_FILE_EXTENSIONS)}"
-    )
-
-
-def download_audio_to_volume(
-    audio_file_url: str,
-) -> tuple[WhisperUniqFilename, AudioFileExtension]:
-    import requests
-    from fastapi import HTTPException
-
-    response = requests.head(audio_file_url, allow_redirects=True)
-    if response.status_code == 404:
-        raise HTTPException(status_code=404, detail="Audio file not found")
-
-    response = requests.get(audio_file_url, allow_redirects=True)
-    response.raise_for_status()
-
-    audio_suffix = detect_audio_format(audio_file_url, response.headers)
-    unique_filename = WhisperUniqFilename(f"{uuid.uuid4()}.{audio_suffix}")
-    file_path = f"{UPLOADS_PATH}/{unique_filename}"
-
-    with open(file_path, "wb") as f:
-        f.write(response.content)
-
-    upload_volume.commit()
-    return unique_filename, audio_suffix
-
-
-def pad_audio(audio_array, sample_rate: int = SAMPLERATE):
-    """Add 0.5s of silence if audio is shorter than the silence_padding window.
-
-    Whisper does not require this strictly, but aligning behavior with Parakeet
-    avoids edge-case crashes on extremely short inputs and makes comparisons easier.
-    """
-    import numpy as np
-
-    audio_duration = len(audio_array) / sample_rate
-    if audio_duration < VAD_CONFIG["silence_padding"]:
-        silence_samples = int(sample_rate * VAD_CONFIG["silence_padding"])
-        silence = np.zeros(silence_samples, dtype=np.float32)
-        return np.concatenate([audio_array, silence])
-    return audio_array
-
-
@app.cls(
    gpu="A10G",
    timeout=5 * MINUTES,
    scaledown_window=5 * MINUTES,
+    allow_concurrent_inputs=6,
    image=image,
-    volumes={CACHE_PATH: model_cache, UPLOADS_PATH: upload_volume},
+    volumes={MODELS_DIR: volume},
 )
-@modal.concurrent(max_inputs=10)
-class TranscriberWhisperLive:
-    """Live transcriber class for small audio segments (A10G).
-
-    Mirrors the Parakeet live class API but uses Faster-Whisper under the hood.
-    """
-
+class Transcriber:
    @modal.enter()
    def enter(self):
        import faster_whisper
@@ -187,200 +71,23 @@ class TranscriberWhisperLive:
            device=self.device,
            compute_type=MODEL_COMPUTE_TYPE,
            num_workers=MODEL_NUM_WORKERS,
-            download_root=CACHE_PATH,
+            download_root=MODELS_DIR,
            local_files_only=True,
        )
-        print(f"Model is on device: {self.device}")

    @modal.method()
    def transcribe_segment(
        self,
-        filename: str,
-        language: str = "en",
+        audio_data: str,
+        audio_suffix: str,
+        language: str,
    ):
-        """Transcribe a single uploaded audio file by filename."""
-        upload_volume.reload()
-
-        file_path = f"{UPLOADS_PATH}/{filename}"
-        if not os.path.exists(file_path):
-            raise FileNotFoundError(f"File not found: {file_path}")
-
-        with self.lock:
-            with NoStdStreams():
-                segments, _ = self.model.transcribe(
-                    file_path,
-                    language=language,
-                    beam_size=5,
-                    word_timestamps=True,
-                    vad_filter=True,
-                    vad_parameters={"min_silence_duration_ms": 500},
-                )
-
-        segments = list(segments)
-        text = "".join(segment.text for segment in segments).strip()
-        words = [
-            {
-                "word": word.word,
-                "start": round(float(word.start), 2),
-                "end": round(float(word.end), 2),
-            }
-            for segment in segments
-            for word in segment.words
-        ]
-
-        return {"text": text, "words": words}
-
-    @modal.method()
-    def transcribe_batch(
-        self,
-        filenames: list[str],
-        language: str = "en",
-    ):
-        """Transcribe multiple uploaded audio files and return per-file results."""
-        upload_volume.reload()
-
-        results = []
-        for filename in filenames:
-            file_path = f"{UPLOADS_PATH}/{filename}"
-            if not os.path.exists(file_path):
-                raise FileNotFoundError(f"Batch file not found: {file_path}")
-
-            with self.lock:
-                with NoStdStreams():
-                    segments, _ = self.model.transcribe(
-                        file_path,
-                        language=language,
-                        beam_size=5,
-                        word_timestamps=True,
-                        vad_filter=True,
-                        vad_parameters={"min_silence_duration_ms": 500},
-                    )
-
-            segments = list(segments)
-            text = "".join(seg.text for seg in segments).strip()
-            words = [
-                {
-                    "word": w.word,
-                    "start": round(float(w.start), 2),
-                    "end": round(float(w.end), 2),
-                }
-                for seg in segments
-                for w in seg.words
-            ]
-
-            results.append(
-                {
-                    "filename": filename,
-                    "text": text,
-                    "words": words,
-                }
-            )
-
-        return results
-
-
-@app.cls(
-    gpu="L40S",
-    timeout=15 * MINUTES,
-    image=image,
-    volumes={CACHE_PATH: model_cache, UPLOADS_PATH: upload_volume},
-)
-class TranscriberWhisperFile:
-    """File transcriber for larger/longer audio, using VAD-driven batching (L40S)."""
-
-    @modal.enter()
-    def enter(self):
-        import faster_whisper
-        import torch
-        from silero_vad import load_silero_vad
-
-        self.lock = threading.Lock()
-        self.use_gpu = torch.cuda.is_available()
-        self.device = "cuda" if self.use_gpu else "cpu"
-        self.model = faster_whisper.WhisperModel(
-            MODEL_NAME,
-            device=self.device,
-            compute_type=MODEL_COMPUTE_TYPE,
-            num_workers=MODEL_NUM_WORKERS,
-            download_root=CACHE_PATH,
-            local_files_only=True,
-        )
-        self.vad_model = load_silero_vad(onnx=False)
-
-    @modal.method()
-    def transcribe_segment(
-        self, filename: str, timestamp_offset: float = 0.0, language: str = "en"
-    ):
-        import librosa
-        import numpy as np
-        from silero_vad import VADIterator
-
-        def vad_segments(
-            audio_array,
-            sample_rate: int = SAMPLERATE,
-            window_size: int = VAD_CONFIG["window_size"],
-        ) -> Generator[TimeSegment, None, None]:
-            """Generate speech segments as TimeSegment using Silero VAD."""
-            iterator = VADIterator(self.vad_model, sampling_rate=sample_rate)
-            start = None
-            for i in range(0, len(audio_array), window_size):
-                chunk = audio_array[i : i + window_size]
-                if len(chunk) < window_size:
-                    chunk = np.pad(
-                        chunk, (0, window_size - len(chunk)), mode="constant"
-                    )
-                speech = iterator(chunk)
-                if not speech:
-                    continue
-                if "start" in speech:
-                    start = speech["start"]
-                    continue
-                if "end" in speech and start is not None:
-                    end = speech["end"]
-                    yield TimeSegment(
-                        start / float(SAMPLERATE), end / float(SAMPLERATE)
-                    )
-                    start = None
-            iterator.reset_states()
-
-        upload_volume.reload()
-        file_path = f"{UPLOADS_PATH}/{filename}"
-        if not os.path.exists(file_path):
-            raise FileNotFoundError(f"File not found: {file_path}")
-
-        audio_array, _sr = librosa.load(file_path, sr=SAMPLERATE, mono=True)
-
-        # Batch segments up to ~30s windows by merging contiguous VAD segments
-        merged_batches: list[TimeSegment] = []
-        batch_start = None
-        batch_end = None
-        max_duration = VAD_CONFIG["batch_max_duration"]
-        for segment in vad_segments(audio_array):
-            seg_start, seg_end = segment.start, segment.end
-            if batch_start is None:
-                batch_start, batch_end = seg_start, seg_end
-                continue
-            if seg_end - batch_start <= max_duration:
-                batch_end = seg_end
-            else:
-                merged_batches.append(TimeSegment(batch_start, batch_end))
-                batch_start, batch_end = seg_start, seg_end
-        if batch_start is not None and batch_end is not None:
-            merged_batches.append(TimeSegment(batch_start, batch_end))
-
-        all_text = []
-        all_words = []
-
-        for segment in merged_batches:
-            start_time, end_time = segment.start, segment.end
-            s_idx = int(start_time * SAMPLERATE)
-            e_idx = int(end_time * SAMPLERATE)
-            segment = audio_array[s_idx:e_idx]
-            segment = pad_audio(segment, SAMPLERATE)
+        with tempfile.NamedTemporaryFile("wb+", suffix=f".{audio_suffix}") as fp:
+            fp.write(audio_data)

            with self.lock:
                segments, _ = self.model.transcribe(
-                    segment,
+                    fp.name,
                    language=language,
                    beam_size=5,
                    word_timestamps=True,
@@ -389,220 +96,66 @@ class TranscriberWhisperFile:
                )

            segments = list(segments)
-            text = "".join(seg.text for seg in segments).strip()
+            text = "".join(segment.text for segment in segments)
            words = [
-                {
-                    "word": w.word,
-                    "start": round(float(w.start) + start_time + timestamp_offset, 2),
-                    "end": round(float(w.end) + start_time + timestamp_offset, 2),
-                }
-                for seg in segments
-                for w in seg.words
+                {"word": word.word, "start": word.start, "end": word.end}
+                for segment in segments
+                for word in segment.words
            ]
-            if text:
-                all_text.append(text)
-            all_words.extend(words)

-        return {"text": " ".join(all_text), "words": all_words}
-
-
-def detect_audio_format(url: str, headers: dict) -> str:
-    from urllib.parse import urlparse
-
-    from fastapi import HTTPException
-
-    url_path = urlparse(url).path
-    for ext in SUPPORTED_FILE_EXTENSIONS:
-        if url_path.lower().endswith(f".{ext}"):
-            return ext
-
-    content_type = headers.get("content-type", "").lower()
-    if "audio/mpeg" in content_type or "audio/mp3" in content_type:
-        return "mp3"
-    if "audio/wav" in content_type:
-        return "wav"
-    if "audio/mp4" in content_type:
-        return "mp4"
-
-    raise HTTPException(
-        status_code=400,
-        detail=(
-            f"Unsupported audio format for URL. Supported extensions: {', '.join(SUPPORTED_FILE_EXTENSIONS)}"
-        ),
-    )
-
-
-def download_audio_to_volume(audio_file_url: str) -> tuple[str, str]:
-    import requests
-    from fastapi import HTTPException
-
-    response = requests.head(audio_file_url, allow_redirects=True)
-    if response.status_code == 404:
-        raise HTTPException(status_code=404, detail="Audio file not found")
-
-    response = requests.get(audio_file_url, allow_redirects=True)
-    response.raise_for_status()
-
-    audio_suffix = detect_audio_format(audio_file_url, response.headers)
-    unique_filename = f"{uuid.uuid4()}.{audio_suffix}"
-    file_path = f"{UPLOADS_PATH}/{unique_filename}"
-
-    with open(file_path, "wb") as f:
-        f.write(response.content)
-
-    upload_volume.commit()
-    return unique_filename, audio_suffix
+            return {"text": text, "words": words}


@app.function(
    scaledown_window=60,
-    timeout=600,
+    timeout=60,
+    allow_concurrent_inputs=40,
    secrets=[
        modal.Secret.from_name("reflector-gpu"),
    ],
-    volumes={CACHE_PATH: model_cache, UPLOADS_PATH: upload_volume},
-    image=image,
+    volumes={MODELS_DIR: volume},
 )
-@modal.concurrent(max_inputs=40)
@modal.asgi_app()
 def web():
-    from fastapi import (
-        Body,
-        Depends,
-        FastAPI,
-        Form,
-        HTTPException,
-        UploadFile,
-        status,
-    )
+    from fastapi import Body, Depends, FastAPI, HTTPException, UploadFile, status
    from fastapi.security import OAuth2PasswordBearer
+    from typing_extensions import Annotated

-    transcriber_live = TranscriberWhisperLive()
-    transcriber_file = TranscriberWhisperFile()
+    transcriber = Transcriber()

    app = FastAPI()

    oauth2_scheme = OAuth2PasswordBearer(tokenUrl="token")

-    def apikey_auth(apikey: str = Depends(oauth2_scheme)):
-        if apikey == os.environ["REFLECTOR_GPU_APIKEY"]:
-            return
-        raise HTTPException(
-            status_code=status.HTTP_401_UNAUTHORIZED,
-            detail="Invalid API key",
-            headers={"WWW-Authenticate": "Bearer"},
-        )
+    supported_file_types = ["mp3", "mp4", "mpeg", "mpga", "m4a", "wav", "webm"]

-    class TranscriptResponse(dict):
-        pass
+    def apikey_auth(apikey: str = Depends(oauth2_scheme)):
+        if apikey != os.environ["REFLECTOR_GPU_APIKEY"]:
+            raise HTTPException(
+                status_code=status.HTTP_401_UNAUTHORIZED,
+                detail="Invalid API key",
+                headers={"WWW-Authenticate": "Bearer"},
+            )
+
+    class TranscriptResponse(BaseModel):
+        result: dict

    @app.post("/v1/audio/transcriptions", dependencies=[Depends(apikey_auth)])
    def transcribe(
-        file: UploadFile = None,
-        files: list[UploadFile] | None = None,
-        model: str = Form(MODEL_NAME),
-        language: str = Form("en"),
-        batch: bool = Form(False),
-    ):
-        if not file and not files:
-            raise HTTPException(
-                status_code=400, detail="Either 'file' or 'files' parameter is required"
-            )
-        if batch and not files:
-            raise HTTPException(
-                status_code=400, detail="Batch transcription requires 'files'"
-            )
+        file: UploadFile,
+        model: str = "whisper-1",
+        language: Annotated[str, Body(...)] = "en",
+    ) -> TranscriptResponse:
+        audio_data = file.file.read()
+        audio_suffix = file.filename.split(".")[-1]
+        assert audio_suffix in supported_file_types

-        upload_files = [file] if file else files
-
-        uploaded_filenames: list[str] = []
-        for upload_file in upload_files:
-            audio_suffix = upload_file.filename.split(".")[-1]
-            if audio_suffix not in SUPPORTED_FILE_EXTENSIONS:
-                raise HTTPException(
-                    status_code=400,
-                    detail=(
-                        f"Unsupported audio format. Supported extensions: {', '.join(SUPPORTED_FILE_EXTENSIONS)}"
-                    ),
-                )
-
-            unique_filename = f"{uuid.uuid4()}.{audio_suffix}"
-            file_path = f"{UPLOADS_PATH}/{unique_filename}"
-            with open(file_path, "wb") as f:
-                content = upload_file.file.read()
-                f.write(content)
-            uploaded_filenames.append(unique_filename)
-
-        upload_volume.commit()
-
-        try:
-            if batch and len(upload_files) > 1:
-                func = transcriber_live.transcribe_batch.spawn(
-                    filenames=uploaded_filenames,
-                    language=language,
-                )
-                results = func.get()
-                return {"results": results}
-
-            results = []
-            for filename in uploaded_filenames:
-                func = transcriber_live.transcribe_segment.spawn(
-                    filename=filename,
-                    language=language,
-                )
-                result = func.get()
-                result["filename"] = filename
-                results.append(result)
-
-            return {"results": results} if len(results) > 1 else results[0]
-        finally:
-            for filename in uploaded_filenames:
-                try:
-                    file_path = f"{UPLOADS_PATH}/{filename}"
-                    os.remove(file_path)
-                except Exception:
-                    pass
-            upload_volume.commit()
-
-    @app.post("/v1/audio/transcriptions-from-url", dependencies=[Depends(apikey_auth)])
-    def transcribe_from_url(
-        audio_file_url: str = Body(
-            ..., description="URL of the audio file to transcribe"
-        ),
-        model: str = Body(MODEL_NAME),
-        language: str = Body("en"),
-        timestamp_offset: float = Body(0.0),
-    ):
-        unique_filename, _audio_suffix = download_audio_to_volume(audio_file_url)
-        try:
-            func = transcriber_file.transcribe_segment.spawn(
-                filename=unique_filename,
-                timestamp_offset=timestamp_offset,
-                language=language,
-            )
-            result = func.get()
-            return result
-        finally:
-            try:
-                file_path = f"{UPLOADS_PATH}/{unique_filename}"
-                os.remove(file_path)
-                upload_volume.commit()
-            except Exception:
-                pass
+        func = transcriber.transcribe_segment.spawn(
+            audio_data=audio_data,
+            audio_suffix=audio_suffix,
+            language=language,
+        )
+        result = func.get()
+        return result

    return app
-
-
-class NoStdStreams:
-    def __init__(self):
-        self.devnull = open(os.devnull, "w")
-
-    def __enter__(self):
-        self._stdout, self._stderr = sys.stdout, sys.stderr
-        self._stdout.flush()
-        self._stderr.flush()
-        sys.stdout, sys.stderr = self.devnull, self.devnull
-
-    def __exit__(self, exc_type, exc_value, traceback):
-        sys.stdout, sys.stderr = self._stdout, self._stderr
-        self.devnull.close()
--- a/server/gpu/modal_deployments/reflector_transcriber_parakeet.py
+++ b/server/gpu/modal_deployments/reflector_transcriber_parakeet.py
@@ -1,658 +0,0 @@
-import logging
-import os
-import sys
-import threading
-import uuid
-from typing import Generator, Mapping, NamedTuple, NewType, TypedDict
-from urllib.parse import urlparse
-
-import modal
-
-MODEL_NAME = "nvidia/parakeet-tdt-0.6b-v2"
-SUPPORTED_FILE_EXTENSIONS = ["mp3", "mp4", "mpeg", "mpga", "m4a", "wav", "webm"]
-SAMPLERATE = 16000
-UPLOADS_PATH = "/uploads"
-CACHE_PATH = "/cache"
-VAD_CONFIG = {
-    "batch_max_duration": 30.0,
-    "silence_padding": 0.5,
-    "window_size": 512,
-}
-
-ParakeetUniqFilename = NewType("ParakeetUniqFilename", str)
-AudioFileExtension = NewType("AudioFileExtension", str)
-
-
-class TimeSegment(NamedTuple):
-    """Represents a time segment with start and end times."""
-
-    start: float
-    end: float
-
-
-class AudioSegment(NamedTuple):
-    """Represents an audio segment with timing and audio data."""
-
-    start: float
-    end: float
-    audio: any
-
-
-class TranscriptResult(NamedTuple):
-    """Represents a transcription result with text and word timings."""
-
-    text: str
-    words: list["WordTiming"]
-
-
-class WordTiming(TypedDict):
-    """Represents a word with its timing information."""
-
-    word: str
-    start: float
-    end: float
-
-
-app = modal.App("reflector-transcriber-parakeet")
-
-# Volume for caching model weights
-model_cache = modal.Volume.from_name("parakeet-model-cache", create_if_missing=True)
-# Volume for temporary file uploads
-upload_volume = modal.Volume.from_name("parakeet-uploads", create_if_missing=True)
-
-image = (
-    modal.Image.from_registry(
-        "nvidia/cuda:12.8.0-cudnn-devel-ubuntu22.04", add_python="3.12"
-    )
-    .env(
-        {
-            "HF_HUB_ENABLE_HF_TRANSFER": "1",
-            "HF_HOME": "/cache",
-            "DEBIAN_FRONTEND": "noninteractive",
-            "CXX": "g++",
-            "CC": "g++",
-        }
-    )
-    .apt_install("ffmpeg")
-    .pip_install(
-        "hf_transfer==0.1.9",
-        "huggingface_hub[hf-xet]==0.31.2",
-        "nemo_toolkit[asr]==2.3.0",
-        "cuda-python==12.8.0",
-        "fastapi==0.115.12",
-        "numpy<2",
-        "librosa==0.10.1",
-        "requests",
-        "silero-vad==5.1.0",
-        "torch",
-    )
-    .entrypoint([])  # silence chatty logs by container on start
-)
-
-
-def detect_audio_format(url: str, headers: Mapping[str, str]) -> AudioFileExtension:
-    parsed_url = urlparse(url)
-    url_path = parsed_url.path
-
-    for ext in SUPPORTED_FILE_EXTENSIONS:
-        if url_path.lower().endswith(f".{ext}"):
-            return AudioFileExtension(ext)
-
-    content_type = headers.get("content-type", "").lower()
-    if "audio/mpeg" in content_type or "audio/mp3" in content_type:
-        return AudioFileExtension("mp3")
-    if "audio/wav" in content_type:
-        return AudioFileExtension("wav")
-    if "audio/mp4" in content_type:
-        return AudioFileExtension("mp4")
-
-    raise ValueError(
-        f"Unsupported audio format for URL: {url}. "
-        f"Supported extensions: {', '.join(SUPPORTED_FILE_EXTENSIONS)}"
-    )
-
-
-def download_audio_to_volume(
-    audio_file_url: str,
-) -> tuple[ParakeetUniqFilename, AudioFileExtension]:
-    import requests
-    from fastapi import HTTPException
-
-    response = requests.head(audio_file_url, allow_redirects=True)
-    if response.status_code == 404:
-        raise HTTPException(status_code=404, detail="Audio file not found")
-
-    response = requests.get(audio_file_url, allow_redirects=True)
-    response.raise_for_status()
-
-    audio_suffix = detect_audio_format(audio_file_url, response.headers)
-    unique_filename = ParakeetUniqFilename(f"{uuid.uuid4()}.{audio_suffix}")
-    file_path = f"{UPLOADS_PATH}/{unique_filename}"
-
-    with open(file_path, "wb") as f:
-        f.write(response.content)
-
-    upload_volume.commit()
-    return unique_filename, audio_suffix
-
-
-def pad_audio(audio_array, sample_rate: int = SAMPLERATE):
-    """Add 0.5 seconds of silence if audio is less than 500ms.
-
-    This is a workaround for a Parakeet bug where very short audio (<500ms) causes:
-    ValueError: `char_offsets`: [] and `processed_tokens`: [157, 834, 834, 841]
-    have to be of the same length
-
-    See: https://github.com/NVIDIA/NeMo/issues/8451
-    """
-    import numpy as np
-
-    audio_duration = len(audio_array) / sample_rate
-    if audio_duration < 0.5:
-        silence_samples = int(sample_rate * 0.5)
-        silence = np.zeros(silence_samples, dtype=np.float32)
-        return np.concatenate([audio_array, silence])
-    return audio_array
-
-
-@app.cls(
-    gpu="A10G",
-    timeout=600,
-    scaledown_window=300,
-    image=image,
-    volumes={CACHE_PATH: model_cache, UPLOADS_PATH: upload_volume},
-    enable_memory_snapshot=True,
-    experimental_options={"enable_gpu_snapshot": True},
-)
-@modal.concurrent(max_inputs=10)
-class TranscriberParakeetLive:
-    @modal.enter(snap=True)
-    def enter(self):
-        import nemo.collections.asr as nemo_asr
-
-        logging.getLogger("nemo_logger").setLevel(logging.CRITICAL)
-
-        self.lock = threading.Lock()
-        self.model = nemo_asr.models.ASRModel.from_pretrained(model_name=MODEL_NAME)
-        device = next(self.model.parameters()).device
-        print(f"Model is on device: {device}")
-
-    @modal.method()
-    def transcribe_segment(
-        self,
-        filename: str,
-    ):
-        import librosa
-
-        upload_volume.reload()
-
-        file_path = f"{UPLOADS_PATH}/{filename}"
-        if not os.path.exists(file_path):
-            raise FileNotFoundError(f"File not found: {file_path}")
-
-        audio_array, sample_rate = librosa.load(file_path, sr=SAMPLERATE, mono=True)
-        padded_audio = pad_audio(audio_array, sample_rate)
-
-        with self.lock:
-            with NoStdStreams():
-                (output,) = self.model.transcribe([padded_audio], timestamps=True)
-
-        text = output.text.strip()
-        words: list[WordTiming] = [
-            WordTiming(
-                # XXX the space added here is to match the output of whisper
-                # whisper add space to each words, while parakeet don't
-                word=word_info["word"] + " ",
-                start=round(word_info["start"], 2),
-                end=round(word_info["end"], 2),
-            )
-            for word_info in output.timestamp["word"]
-        ]
-
-        return {"text": text, "words": words}
-
-    @modal.method()
-    def transcribe_batch(
-        self,
-        filenames: list[str],
-    ):
-        import librosa
-
-        upload_volume.reload()
-
-        results = []
-        audio_arrays = []
-
-        # Load all audio files with padding
-        for filename in filenames:
-            file_path = f"{UPLOADS_PATH}/{filename}"
-            if not os.path.exists(file_path):
-                raise FileNotFoundError(f"Batch file not found: {file_path}")
-
-            audio_array, sample_rate = librosa.load(file_path, sr=SAMPLERATE, mono=True)
-            padded_audio = pad_audio(audio_array, sample_rate)
-            audio_arrays.append(padded_audio)
-
-        with self.lock:
-            with NoStdStreams():
-                outputs = self.model.transcribe(audio_arrays, timestamps=True)
-
-        # Process results for each file
-        for i, (filename, output) in enumerate(zip(filenames, outputs)):
-            text = output.text.strip()
-
-            words: list[WordTiming] = [
-                WordTiming(
-                    word=word_info["word"] + " ",
-                    start=round(word_info["start"], 2),
-                    end=round(word_info["end"], 2),
-                )
-                for word_info in output.timestamp["word"]
-            ]
-
-            results.append(
-                {
-                    "filename": filename,
-                    "text": text,
-                    "words": words,
-                }
-            )
-
-        return results
-
-
-# L40S class for file transcription (bigger files)
-@app.cls(
-    gpu="L40S",
-    timeout=900,
-    image=image,
-    volumes={CACHE_PATH: model_cache, UPLOADS_PATH: upload_volume},
-    enable_memory_snapshot=True,
-    experimental_options={"enable_gpu_snapshot": True},
-)
-class TranscriberParakeetFile:
-    @modal.enter(snap=True)
-    def enter(self):
-        import nemo.collections.asr as nemo_asr
-        import torch
-        from silero_vad import load_silero_vad
-
-        logging.getLogger("nemo_logger").setLevel(logging.CRITICAL)
-
-        self.model = nemo_asr.models.ASRModel.from_pretrained(model_name=MODEL_NAME)
-        device = next(self.model.parameters()).device
-        print(f"Model is on device: {device}")
-
-        torch.set_num_threads(1)
-        self.vad_model = load_silero_vad(onnx=False)
-        print("Silero VAD initialized")
-
-    @modal.method()
-    def transcribe_segment(
-        self,
-        filename: str,
-        timestamp_offset: float = 0.0,
-    ):
-        import librosa
-        import numpy as np
-        from silero_vad import VADIterator
-
-        def load_and_convert_audio(file_path):
-            audio_array, sample_rate = librosa.load(file_path, sr=SAMPLERATE, mono=True)
-            return audio_array
-
-        def vad_segment_generator(
-            audio_array,
-        ) -> Generator[TimeSegment, None, None]:
-            """Generate speech segments using VAD with start/end sample indices"""
-            vad_iterator = VADIterator(self.vad_model, sampling_rate=SAMPLERATE)
-            window_size = VAD_CONFIG["window_size"]
-            start = None
-
-            for i in range(0, len(audio_array), window_size):
-                chunk = audio_array[i : i + window_size]
-                if len(chunk) < window_size:
-                    chunk = np.pad(
-                        chunk, (0, window_size - len(chunk)), mode="constant"
-                    )
-
-                speech_dict = vad_iterator(chunk)
-                if not speech_dict:
-                    continue
-
-                if "start" in speech_dict:
-                    start = speech_dict["start"]
-                    continue
-
-                if "end" in speech_dict and start is not None:
-                    end = speech_dict["end"]
-                    start_time = start / float(SAMPLERATE)
-                    end_time = end / float(SAMPLERATE)
-
-                    yield TimeSegment(start_time, end_time)
-                    start = None
-
-            vad_iterator.reset_states()
-
-        def batch_speech_segments(
-            segments: Generator[TimeSegment, None, None], max_duration: int
-        ) -> Generator[TimeSegment, None, None]:
-            """
-            Input segments:
-              [0-2] [3-5] [6-8] [10-11] [12-15] [17-19] [20-22]
-
-                                  ↓ (max_duration=10)
-
-              Output batches:
-              [0-8]           [10-19]          [20-22]
-
-            Note: silences are kept for better transcription, previous implementation was
-            passing segments separatly, but the output was less accurate.
-            """
-            batch_start_time = None
-            batch_end_time = None
-
-            for segment in segments:
-                start_time, end_time = segment.start, segment.end
-                if batch_start_time is None or batch_end_time is None:
-                    batch_start_time = start_time
-                    batch_end_time = end_time
-                    continue
-
-                total_duration = end_time - batch_start_time
-
-                if total_duration <= max_duration:
-                    batch_end_time = end_time
-                    continue
-
-                yield TimeSegment(batch_start_time, batch_end_time)
-                batch_start_time = start_time
-                batch_end_time = end_time
-
-            if batch_start_time is None or batch_end_time is None:
-                return
-
-            yield TimeSegment(batch_start_time, batch_end_time)
-
-        def batch_segment_to_audio_segment(
-            segments: Generator[TimeSegment, None, None],
-            audio_array,
-        ) -> Generator[AudioSegment, None, None]:
-            """Extract audio segments and apply padding for Parakeet compatibility.
-
-            Uses pad_audio to ensure segments are at least 0.5s long, preventing
-            Parakeet crashes. This padding may cause slight timing overlaps between
-            segments, which are corrected by enforce_word_timing_constraints.
-            """
-            for segment in segments:
-                start_time, end_time = segment.start, segment.end
-                start_sample = int(start_time * SAMPLERATE)
-                end_sample = int(end_time * SAMPLERATE)
-                audio_segment = audio_array[start_sample:end_sample]
-
-                padded_segment = pad_audio(audio_segment, SAMPLERATE)
-
-                yield AudioSegment(start_time, end_time, padded_segment)
-
-        def transcribe_batch(model, audio_segments: list) -> list:
-            with NoStdStreams():
-                outputs = model.transcribe(audio_segments, timestamps=True)
-            return outputs
-
-        def enforce_word_timing_constraints(
-            words: list[WordTiming],
-        ) -> list[WordTiming]:
-            """Enforce that word end times don't exceed the start time of the next word.
-
-            Due to silence padding added in batch_segment_to_audio_segment for better
-            transcription accuracy, word timings from different segments may overlap.
-            This function ensures there are no overlaps by adjusting end times.
-            """
-            if len(words) <= 1:
-                return words
-
-            enforced_words = []
-            for i, word in enumerate(words):
-                enforced_word = word.copy()
-
-                if i < len(words) - 1:
-                    next_start = words[i + 1]["start"]
-                    if enforced_word["end"] > next_start:
-                        enforced_word["end"] = next_start
-
-                enforced_words.append(enforced_word)
-
-            return enforced_words
-
-        def emit_results(
-            results: list,
-            segments_info: list[AudioSegment],
-        ) -> Generator[TranscriptResult, None, None]:
-            """Yield transcribed text and word timings from model output, adjusting timestamps to absolute positions."""
-            for i, (output, segment) in enumerate(zip(results, segments_info)):
-                start_time, end_time = segment.start, segment.end
-                text = output.text.strip()
-                words: list[WordTiming] = [
-                    WordTiming(
-                        word=word_info["word"] + " ",
-                        start=round(
-                            word_info["start"] + start_time + timestamp_offset, 2
-                        ),
-                        end=round(word_info["end"] + start_time + timestamp_offset, 2),
-                    )
-                    for word_info in output.timestamp["word"]
-                ]
-
-                yield TranscriptResult(text, words)
-
-        upload_volume.reload()
-
-        file_path = f"{UPLOADS_PATH}/{filename}"
-        if not os.path.exists(file_path):
-            raise FileNotFoundError(f"File not found: {file_path}")
-
-        audio_array = load_and_convert_audio(file_path)
-        total_duration = len(audio_array) / float(SAMPLERATE)
-
-        all_text_parts: list[str] = []
-        all_words: list[WordTiming] = []
-
-        raw_segments = vad_segment_generator(audio_array)
-        speech_segments = batch_speech_segments(
-            raw_segments,
-            VAD_CONFIG["batch_max_duration"],
-        )
-        audio_segments = batch_segment_to_audio_segment(speech_segments, audio_array)
-
-        for batch in audio_segments:
-            audio_segment = batch.audio
-            results = transcribe_batch(self.model, [audio_segment])
-
-            for result in emit_results(
-                results,
-                [batch],
-            ):
-                if not result.text:
-                    continue
-                all_text_parts.append(result.text)
-                all_words.extend(result.words)
-
-        all_words = enforce_word_timing_constraints(all_words)
-
-        combined_text = " ".join(all_text_parts)
-        return {"text": combined_text, "words": all_words}
-
-
-@app.function(
-    scaledown_window=60,
-    timeout=600,
-    secrets=[
-        modal.Secret.from_name("reflector-gpu"),
-    ],
-    volumes={CACHE_PATH: model_cache, UPLOADS_PATH: upload_volume},
-    image=image,
-)
-@modal.concurrent(max_inputs=40)
-@modal.asgi_app()
-def web():
-    import os
-    import uuid
-
-    from fastapi import (
-        Body,
-        Depends,
-        FastAPI,
-        Form,
-        HTTPException,
-        UploadFile,
-        status,
-    )
-    from fastapi.security import OAuth2PasswordBearer
-    from pydantic import BaseModel
-
-    transcriber_live = TranscriberParakeetLive()
-    transcriber_file = TranscriberParakeetFile()
-
-    app = FastAPI()
-
-    oauth2_scheme = OAuth2PasswordBearer(tokenUrl="token")
-
-    def apikey_auth(apikey: str = Depends(oauth2_scheme)):
-        if apikey == os.environ["REFLECTOR_GPU_APIKEY"]:
-            return
-        raise HTTPException(
-            status_code=status.HTTP_401_UNAUTHORIZED,
-            detail="Invalid API key",
-            headers={"WWW-Authenticate": "Bearer"},
-        )
-
-    class TranscriptResponse(BaseModel):
-        result: dict
-
-    @app.post("/v1/audio/transcriptions", dependencies=[Depends(apikey_auth)])
-    def transcribe(
-        file: UploadFile = None,
-        files: list[UploadFile] | None = None,
-        model: str = Form(MODEL_NAME),
-        language: str = Form("en"),
-        batch: bool = Form(False),
-    ):
-        # Parakeet only supports English
-        if language != "en":
-            raise HTTPException(
-                status_code=400,
-                detail=f"Parakeet model only supports English. Got language='{language}'",
-            )
-        # Handle both single file and multiple files
-        if not file and not files:
-            raise HTTPException(
-                status_code=400, detail="Either 'file' or 'files' parameter is required"
-            )
-        if batch and not files:
-            raise HTTPException(
-                status_code=400, detail="Batch transcription requires 'files'"
-            )
-
-        upload_files = [file] if file else files
-
-        # Upload files to volume
-        uploaded_filenames = []
-        for upload_file in upload_files:
-            audio_suffix = upload_file.filename.split(".")[-1]
-            assert audio_suffix in SUPPORTED_FILE_EXTENSIONS
-
-            # Generate unique filename
-            unique_filename = f"{uuid.uuid4()}.{audio_suffix}"
-            file_path = f"{UPLOADS_PATH}/{unique_filename}"
-
-            print(f"Writing file to: {file_path}")
-            with open(file_path, "wb") as f:
-                content = upload_file.file.read()
-                f.write(content)
-
-            uploaded_filenames.append(unique_filename)
-
-        upload_volume.commit()
-
-        try:
-            # Use A10G live transcriber for per-file transcription
-            if batch and len(upload_files) > 1:
-                # Use batch transcription
-                func = transcriber_live.transcribe_batch.spawn(
-                    filenames=uploaded_filenames,
-                )
-                results = func.get()
-                return {"results": results}
-
-            # Per-file transcription
-            results = []
-            for filename in uploaded_filenames:
-                func = transcriber_live.transcribe_segment.spawn(
-                    filename=filename,
-                )
-                result = func.get()
-                result["filename"] = filename
-                results.append(result)
-
-            return {"results": results} if len(results) > 1 else results[0]
-
-        finally:
-            for filename in uploaded_filenames:
-                try:
-                    file_path = f"{UPLOADS_PATH}/{filename}"
-                    print(f"Deleting file: {file_path}")
-                    os.remove(file_path)
-                except Exception as e:
-                    print(f"Error deleting {filename}: {e}")
-
-            upload_volume.commit()
-
-    @app.post("/v1/audio/transcriptions-from-url", dependencies=[Depends(apikey_auth)])
-    def transcribe_from_url(
-        audio_file_url: str = Body(
-            ..., description="URL of the audio file to transcribe"
-        ),
-        model: str = Body(MODEL_NAME),
-        language: str = Body("en", description="Language code (only 'en' supported)"),
-        timestamp_offset: float = Body(0.0),
-    ):
-        # Parakeet only supports English
-        if language != "en":
-            raise HTTPException(
-                status_code=400,
-                detail=f"Parakeet model only supports English. Got language='{language}'",
-            )
-        unique_filename, audio_suffix = download_audio_to_volume(audio_file_url)
-
-        try:
-            func = transcriber_file.transcribe_segment.spawn(
-                filename=unique_filename,
-                timestamp_offset=timestamp_offset,
-            )
-            result = func.get()
-            return result
-        finally:
-            try:
-                file_path = f"{UPLOADS_PATH}/{unique_filename}"
-                print(f"Deleting file: {file_path}")
-                os.remove(file_path)
-                upload_volume.commit()
-            except Exception as e:
-                print(f"Error cleaning up {unique_filename}: {e}")
-
-    return app
-
-
-class NoStdStreams:
-    def __init__(self):
-        self.devnull = open(os.devnull, "w")
-
-    def __enter__(self):
-        self._stdout, self._stderr = sys.stdout, sys.stderr
-        self._stdout.flush()
-        self._stderr.flush()
-        sys.stdout, sys.stderr = self.devnull, self.devnull
-
-    def __exit__(self, exc_type, exc_value, traceback):
-        sys.stdout, sys.stderr = self._stdout, self._stderr
-        self.devnull.close()
--- a/server/migrations/versions/0194f65cd6d3_add_webhook_fields_to_rooms.py
+++ b/server/migrations/versions/0194f65cd6d3_add_webhook_fields_to_rooms.py
@@ -1,36 +0,0 @@
-"""Add webhook fields to rooms
-
-Revision ID: 0194f65cd6d3
-Revises: 5a8907fd1d78
-Create Date: 2025-08-27 09:03:19.610995
-
-"""
-
-from typing import Sequence, Union
-
-import sqlalchemy as sa
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "0194f65cd6d3"
-down_revision: Union[str, None] = "5a8907fd1d78"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("room", schema=None) as batch_op:
-        batch_op.add_column(sa.Column("webhook_url", sa.String(), nullable=True))
-        batch_op.add_column(sa.Column("webhook_secret", sa.String(), nullable=True))
-
-    # ### end Alembic commands ###
-
-
-def downgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("room", schema=None) as batch_op:
-        batch_op.drop_column("webhook_secret")
-        batch_op.drop_column("webhook_url")
-
-    # ### end Alembic commands ###
--- a/server/migrations/versions/0ab2d7ffaa16_add_long_summary_to_search_vector.py
+++ b/server/migrations/versions/0ab2d7ffaa16_add_long_summary_to_search_vector.py
@@ -1,64 +0,0 @@
-"""add_long_summary_to_search_vector
-
-Revision ID: 0ab2d7ffaa16
-Revises: b1c33bd09963
-Create Date: 2025-08-15 13:27:52.680211
-
-"""
-
-from typing import Sequence, Union
-
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "0ab2d7ffaa16"
-down_revision: Union[str, None] = "b1c33bd09963"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # Drop the existing search vector column and index
-    op.drop_index("idx_transcript_search_vector_en", table_name="transcript")
-    op.drop_column("transcript", "search_vector_en")
-
-    # Recreate the search vector column with long_summary included
-    op.execute("""
-        ALTER TABLE transcript ADD COLUMN search_vector_en tsvector
-        GENERATED ALWAYS AS (
-            setweight(to_tsvector('english', coalesce(title, '')), 'A') ||
-            setweight(to_tsvector('english', coalesce(long_summary, '')), 'B') ||
-            setweight(to_tsvector('english', coalesce(webvtt, '')), 'C')
-        ) STORED
-    """)
-
-    # Recreate the GIN index for the search vector
-    op.create_index(
-        "idx_transcript_search_vector_en",
-        "transcript",
-        ["search_vector_en"],
-        postgresql_using="gin",
-    )
-
-
-def downgrade() -> None:
-    # Drop the updated search vector column and index
-    op.drop_index("idx_transcript_search_vector_en", table_name="transcript")
-    op.drop_column("transcript", "search_vector_en")
-
-    # Recreate the original search vector column without long_summary
-    op.execute("""
-        ALTER TABLE transcript ADD COLUMN search_vector_en tsvector
-        GENERATED ALWAYS AS (
-            setweight(to_tsvector('english', coalesce(title, '')), 'A') ||
-            setweight(to_tsvector('english', coalesce(webvtt, '')), 'B')
-        ) STORED
-    """)
-
-    # Recreate the GIN index for the search vector
-    op.create_index(
-        "idx_transcript_search_vector_en",
-        "transcript",
-        ["search_vector_en"],
-        postgresql_using="gin",
-    )
--- a/server/migrations/versions/0ce521cda2ee_remove_user_id_from_meeting_table.py
+++ b/server/migrations/versions/0ce521cda2ee_remove_user_id_from_meeting_table.py
@@ -1,36 +0,0 @@
-"""remove user_id from meeting table
-
-Revision ID: 0ce521cda2ee
-Revises: 6dec9fb5b46c
-Create Date: 2025-09-10 12:40:55.688899
-
-"""
-
-from typing import Sequence, Union
-
-import sqlalchemy as sa
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "0ce521cda2ee"
-down_revision: Union[str, None] = "6dec9fb5b46c"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.drop_column("user_id")
-
-    # ### end Alembic commands ###
-
-
-def downgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.add_column(
-            sa.Column("user_id", sa.VARCHAR(), autoincrement=False, nullable=True)
-        )
-
-    # ### end Alembic commands ###
--- a/server/migrations/versions/2890b5104577_add_events_column_to_meetings_table.py
+++ b/server/migrations/versions/2890b5104577_add_events_column_to_meetings_table.py
@@ -1,38 +0,0 @@
-"""Add events column to meetings table
-
-Revision ID: 2890b5104577
-Revises: 6e6ea8e607c5
-Create Date: 2025-09-02 17:51:41.620777
-
-"""
-
-from typing import Sequence, Union
-
-import sqlalchemy as sa
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "2890b5104577"
-down_revision: Union[str, None] = "6e6ea8e607c5"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.add_column(
-            sa.Column(
-                "events", sa.JSON(), server_default=sa.text("'[]'"), nullable=False
-            )
-        )
-
-    # ### end Alembic commands ###
-
-
-def downgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.drop_column("events")
-
-    # ### end Alembic commands ###
--- a/server/migrations/versions/2ae3db106d4e_clean_up_orphaned_room_id_references_in_.py
+++ b/server/migrations/versions/2ae3db106d4e_clean_up_orphaned_room_id_references_in_.py
@@ -1,32 +0,0 @@
-"""clean up orphaned room_id references in meeting table
-
-Revision ID: 2ae3db106d4e
-Revises: def1b5867d4c
-Create Date: 2025-09-11 10:35:15.759967
-
-"""
-
-from typing import Sequence, Union
-
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "2ae3db106d4e"
-down_revision: Union[str, None] = "def1b5867d4c"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # Set room_id to NULL for meetings that reference non-existent rooms
-    op.execute("""
-        UPDATE meeting
-        SET room_id = NULL
-        WHERE room_id IS NOT NULL
-          AND room_id NOT IN (SELECT id FROM room WHERE id IS NOT NULL)
-    """)
-
-
-def downgrade() -> None:
-    # Cannot restore orphaned references - no operation needed
-    pass
--- a/server/migrations/versions/5a8907fd1d78_add_cascade_delete_to_meeting_consent_.py
+++ b/server/migrations/versions/5a8907fd1d78_add_cascade_delete_to_meeting_consent_.py
@@ -1,50 +0,0 @@
-"""add cascade delete to meeting consent foreign key
-
-Revision ID: 5a8907fd1d78
-Revises: 0ab2d7ffaa16
-Create Date: 2025-08-26 17:26:50.945491
-
-"""
-
-from typing import Sequence, Union
-
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "5a8907fd1d78"
-down_revision: Union[str, None] = "0ab2d7ffaa16"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting_consent", schema=None) as batch_op:
-        batch_op.drop_constraint(
-            batch_op.f("meeting_consent_meeting_id_fkey"), type_="foreignkey"
-        )
-        batch_op.create_foreign_key(
-            batch_op.f("meeting_consent_meeting_id_fkey"),
-            "meeting",
-            ["meeting_id"],
-            ["id"],
-            ondelete="CASCADE",
-        )
-
-    # ### end Alembic commands ###
-
-
-def downgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting_consent", schema=None) as batch_op:
-        batch_op.drop_constraint(
-            batch_op.f("meeting_consent_meeting_id_fkey"), type_="foreignkey"
-        )
-        batch_op.create_foreign_key(
-            batch_op.f("meeting_consent_meeting_id_fkey"),
-            "meeting",
-            ["meeting_id"],
-            ["id"],
-        )
-
-    # ### end Alembic commands ###
--- a/server/migrations/versions/6025e9b2bef2_remove_one_active_meeting_per_room_.py
+++ b/server/migrations/versions/6025e9b2bef2_remove_one_active_meeting_per_room_.py
@@ -0,0 +1,53 @@
+"""remove_one_active_meeting_per_room_constraint
+
+Revision ID: 6025e9b2bef2
+Revises: 9f5c78d352d6
+Create Date: 2025-08-18 18:45:44.418392
+
+"""
+
+from typing import Sequence, Union
+
+import sqlalchemy as sa
+from alembic import op
+
+# revision identifiers, used by Alembic.
+revision: str = "6025e9b2bef2"
+down_revision: Union[str, None] = "9f5c78d352d6"
+branch_labels: Union[str, Sequence[str], None] = None
+depends_on: Union[str, Sequence[str], None] = None
+
+
+def upgrade() -> None:
+    # Remove the unique constraint that prevents multiple active meetings per room
+    # This is needed to support calendar integration with overlapping meetings
+    # Check if index exists before trying to drop it
+    from alembic import context
+
+    if context.get_context().dialect.name == "postgresql":
+        conn = op.get_bind()
+        result = conn.execute(
+            sa.text(
+                "SELECT 1 FROM pg_indexes WHERE indexname = 'idx_one_active_meeting_per_room'"
+            )
+        )
+        if result.fetchone():
+            op.drop_index("idx_one_active_meeting_per_room", table_name="meeting")
+    else:
+        # For SQLite, just try to drop it
+        try:
+            op.drop_index("idx_one_active_meeting_per_room", table_name="meeting")
+        except:
+            pass
+
+
+def downgrade() -> None:
+    # Restore the unique constraint
+    op.create_index(
+        "idx_one_active_meeting_per_room",
+        "meeting",
+        ["room_id"],
+        unique=True,
+        postgresql_where=sa.text("is_active = true"),
+        sqlite_where=sa.text("is_active = 1"),
+    )
--- a/server/migrations/versions/61882a919591_webhook_url_and_secret_null_by_default.py
+++ b/server/migrations/versions/61882a919591_webhook_url_and_secret_null_by_default.py
@@ -1,28 +0,0 @@
-"""webhook url and secret null by default
-
-
-Revision ID: 61882a919591
-Revises: 0194f65cd6d3
-Create Date: 2025-08-29 11:46:36.738091
-
-"""
-
-from typing import Sequence, Union
-
-# revision identifiers, used by Alembic.
-revision: str = "61882a919591"
-down_revision: Union[str, None] = "0194f65cd6d3"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    pass
-    # ### end Alembic commands ###
-
-
-def downgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    pass
-    # ### end Alembic commands ###
--- a/server/migrations/versions/6dec9fb5b46c_make_meeting_room_id_required_and_add_.py
+++ b/server/migrations/versions/6dec9fb5b46c_make_meeting_room_id_required_and_add_.py
@@ -1,38 +0,0 @@
-"""make meeting room_id required and add foreign key
-
-Revision ID: 6dec9fb5b46c
-Revises: 61882a919591
-Create Date: 2025-09-10 10:47:06.006819
-
-"""
-
-from typing import Sequence, Union
-
-import sqlalchemy as sa
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "6dec9fb5b46c"
-down_revision: Union[str, None] = "61882a919591"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.alter_column("room_id", existing_type=sa.VARCHAR(), nullable=False)
-        batch_op.create_foreign_key(
-            None, "room", ["room_id"], ["id"], ondelete="CASCADE"
-        )
-
-    # ### end Alembic commands ###
-
-
-def downgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.drop_constraint("meeting_room_id_fkey", type_="foreignkey")
-        batch_op.alter_column("room_id", existing_type=sa.VARCHAR(), nullable=True)
-
-    # ### end Alembic commands ###
--- a/server/migrations/versions/6e6ea8e607c5_add_videoplatform_enum_for_rooms_and_.py
+++ b/server/migrations/versions/6e6ea8e607c5_add_videoplatform_enum_for_rooms_and_.py
@@ -1,44 +0,0 @@
-"""Add VideoPlatform enum for rooms and meetings
-
-Revision ID: 6e6ea8e607c5
-Revises: 61882a919591
-Create Date: 2025-09-02 17:33:21.022214
-
-"""
-
-from typing import Sequence, Union
-
-import sqlalchemy as sa
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "6e6ea8e607c5"
-down_revision: Union[str, None] = "61882a919591"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.add_column(
-            sa.Column("platform", sa.String(), server_default="whereby", nullable=False)
-        )
-
-    with op.batch_alter_table("room", schema=None) as batch_op:
-        batch_op.add_column(
-            sa.Column("platform", sa.String(), server_default="whereby", nullable=False)
-        )
-
-    # ### end Alembic commands ###
-
-
-def downgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("room", schema=None) as batch_op:
-        batch_op.drop_column("platform")
-
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.drop_column("platform")
-
-    # ### end Alembic commands ###
--- a/server/migrations/versions/b1c33bd09963_add_search_optimization_indexes.py
+++ b/server/migrations/versions/b1c33bd09963_add_search_optimization_indexes.py
@@ -1,41 +0,0 @@
-"""add_search_optimization_indexes
-
-Revision ID: b1c33bd09963
-Revises: 9f5c78d352d6
-Create Date: 2025-08-14 17:26:02.117408
-
-"""
-
-from typing import Sequence, Union
-
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "b1c33bd09963"
-down_revision: Union[str, None] = "9f5c78d352d6"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # Add indexes for actual search filtering patterns used in frontend
-    # Based on /browse page filters: room_id and source_kind
-
-    # Index for room_id + created_at (for room-specific searches with date ordering)
-    op.create_index(
-        "idx_transcript_room_id_created_at",
-        "transcript",
-        ["room_id", "created_at"],
-        if_not_exists=True,
-    )
-
-    # Index for source_kind alone (actively used filter in frontend)
-    op.create_index(
-        "idx_transcript_source_kind", "transcript", ["source_kind"], if_not_exists=True
-    )
-
-
-def downgrade() -> None:
-    # Remove the indexes in reverse order
-    op.drop_index("idx_transcript_source_kind", "transcript", if_exists=True)
-    op.drop_index("idx_transcript_room_id_created_at", "transcript", if_exists=True)
--- a/server/migrations/versions/d4a1c446458c_add_grace_period_fields_to_meeting.py
+++ b/server/migrations/versions/d4a1c446458c_add_grace_period_fields_to_meeting.py
@@ -0,0 +1,34 @@
+"""add_grace_period_fields_to_meeting
+
+Revision ID: d4a1c446458c
+Revises: 6025e9b2bef2
+Create Date: 2025-08-18 18:50:37.768052
+
+"""
+
+from typing import Sequence, Union
+
+import sqlalchemy as sa
+from alembic import op
+
+# revision identifiers, used by Alembic.
+revision: str = "d4a1c446458c"
+down_revision: Union[str, None] = "6025e9b2bef2"
+branch_labels: Union[str, Sequence[str], None] = None
+depends_on: Union[str, Sequence[str], None] = None
+
+
+def upgrade() -> None:
+    # Add fields to track when participants left for grace period logic
+    op.add_column(
+        "meeting", sa.Column("last_participant_left_at", sa.DateTime(timezone=True))
+    )
+    op.add_column(
+        "meeting",
+        sa.Column("grace_period_minutes", sa.Integer, server_default=sa.text("15")),
+    )
+
+
+def downgrade() -> None:
+    op.drop_column("meeting", "grace_period_minutes")
+    op.drop_column("meeting", "last_participant_left_at")
--- a/server/migrations/versions/def1b5867d4c_make_meeting_room_id_nullable_but_keep_.py
+++ b/server/migrations/versions/def1b5867d4c_make_meeting_room_id_nullable_but_keep_.py
@@ -1,34 +0,0 @@
-"""make meeting room_id nullable but keep foreign key
-
-Revision ID: def1b5867d4c
-Revises: 0ce521cda2ee
-Create Date: 2025-09-11 09:42:18.697264
-
-"""
-
-from typing import Sequence, Union
-
-import sqlalchemy as sa
-from alembic import op
-
-# revision identifiers, used by Alembic.
-revision: str = "def1b5867d4c"
-down_revision: Union[str, None] = "0ce521cda2ee"
-branch_labels: Union[str, Sequence[str], None] = None
-depends_on: Union[str, Sequence[str], None] = None
-
-
-def upgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.alter_column("room_id", existing_type=sa.VARCHAR(), nullable=True)
-
-    # ### end Alembic commands ###
-
-
-def downgrade() -> None:
-    # ### commands auto generated by Alembic - please adjust! ###
-    with op.batch_alter_table("meeting", schema=None) as batch_op:
-        batch_op.alter_column("room_id", existing_type=sa.VARCHAR(), nullable=False)
-
-    # ### end Alembic commands ###
--- a/server/pyproject.toml
+++ b/server/pyproject.toml
@@ -32,6 +32,7 @@ dependencies = [
    "redis>=5.0.1",
    "python-jose[cryptography]>=3.3.0",
    "python-multipart>=0.0.6",
+    "faster-whisper>=0.10.0",
    "transformers>=4.36.2",
    "jsonschema>=4.23.0",
    "openai>=1.59.7",
@@ -40,7 +41,7 @@ dependencies = [
    "llama-index-llms-openai-like>=0.4.0",
    "pytest-env>=1.1.5",
    "webvtt-py>=0.5.0",
-    "PyJWT>=2.8.0",
+    "icalendar>=6.0.0",
 ]

 [dependency-groups]
@@ -57,7 +58,6 @@ tests = [
    "httpx-ws>=0.4.1",
    "pytest-httpx>=0.23.1",
    "pytest-celery>=0.0.0",
-    "pytest-recording>=0.13.4",
    "pytest-docker>=3.2.3",
    "asgi-lifespan>=2.1.0",
 ]
@@ -68,15 +68,6 @@ evaluation = [
    "tqdm>=4.66.0",
    "pydantic>=2.1.1",
 ]
-local = [
-    "pyannote-audio>=3.3.2",
-    "faster-whisper>=0.10.0",
-]
-silero-vad = [
-    "silero-vad>=5.1.2",
-    "torch>=2.8.0",
-    "torchaudio>=2.8.0",
-]

 [tool.uv]
 default-groups = [
@@ -84,21 +75,6 @@ default-groups = [
    "tests",
    "aws",
    "evaluation",
-    "local",
-    "silero-vad"
-]
-
-[[tool.uv.index]]
-name = "pytorch-cpu"
-url = "https://download.pytorch.org/whl/cpu"
-explicit = true
-
-[tool.uv.sources]
-torch = [
-  { index = "pytorch-cpu" },
-]
-torchaudio = [
-  { index = "pytorch-cpu" },
 ]

 [build-system]
@@ -119,9 +95,6 @@ DATABASE_URL = "postgresql://test_user:test_password@localhost:15432/reflector_t
 addopts = "-ra -q --disable-pytest-warnings --cov --cov-report html -v"
 testpaths = ["tests"]
 asyncio_mode = "auto"
-markers = [
-    "gpu_modal: mark test to run only with GPU Modal endpoints (deselect with '-m \"not gpu_modal\"')",
-]

 [tool.ruff.lint]
 select = [
--- a/server/reflector/app.py
+++ b/server/reflector/app.py
@@ -12,9 +12,6 @@ from reflector.events import subscribers_shutdown, subscribers_startup
 from reflector.logger import logger
 from reflector.metrics import metrics_init
 from reflector.settings import settings
-from reflector.video_platforms.jitsi import router as jitsi_router
-from reflector.video_platforms.whereby import router as whereby_router
-from reflector.views.jibri_webhook import router as jibri_webhook_router
 from reflector.views.meetings import router as meetings_router
 from reflector.views.rooms import router as rooms_router
 from reflector.views.rtc_offer import router as rtc_offer_router
@@ -29,6 +26,7 @@ from reflector.views.transcripts_upload import router as transcripts_upload_rout
 from reflector.views.transcripts_webrtc import router as transcripts_webrtc_router
 from reflector.views.transcripts_websocket import router as transcripts_websocket_router
 from reflector.views.user import router as user_router
+from reflector.views.whereby import router as whereby_router
 from reflector.views.zulip import router as zulip_router

 try:
@@ -88,8 +86,6 @@ app.include_router(transcripts_process_router, prefix="/v1")
 app.include_router(user_router, prefix="/v1")
 app.include_router(zulip_router, prefix="/v1")
 app.include_router(whereby_router, prefix="/v1")
-app.include_router(jitsi_router, prefix="/v1")
-app.include_router(jibri_webhook_router)  # No /v1 prefix, uses /api/v1/jibri
 add_pagination(app)

 # prepare celery
--- a/server/reflector/asynctask.py
+++ b/server/reflector/asynctask.py
@@ -1,27 +0,0 @@
-import asyncio
-import functools
-
-from reflector.db import get_database
-
-
-def asynctask(f):
-    @functools.wraps(f)
-    def wrapper(*args, **kwargs):
-        async def run_with_db():
-            database = get_database()
-            await database.connect()
-            try:
-                return await f(*args, **kwargs)
-            finally:
-                await database.disconnect()
-
-        coro = run_with_db()
-        try:
-            loop = asyncio.get_running_loop()
-        except RuntimeError:
-            loop = None
-        if loop and loop.is_running():
-            return loop.run_until_complete(coro)
-        return asyncio.run(coro)
-
-    return wrapper
--- a/server/reflector/db/init.py
+++ b/server/reflector/db/init.py
@@ -24,6 +24,7 @@ def get_database() -> databases.Database:


 # import models
+import reflector.db.calendar_events  # noqa
 import reflector.db.meetings  # noqa
 import reflector.db.recordings  # noqa
 import reflector.db.rooms  # noqa
--- a/server/reflector/db/calendar_events.py
+++ b/server/reflector/db/calendar_events.py
@@ -0,0 +1,193 @@
+from datetime import datetime, timezone
+from typing import Any
+
+import sqlalchemy as sa
+from pydantic import BaseModel, Field
+from sqlalchemy.dialects.postgresql import JSONB
+
+from reflector.db import get_database, metadata
+from reflector.utils import generate_uuid4
+
+calendar_events = sa.Table(
+    "calendar_event",
+    metadata,
+    sa.Column("id", sa.String, primary_key=True),
+    sa.Column(
+        "room_id",
+        sa.String,
+        sa.ForeignKey("room.id", ondelete="CASCADE"),
+        nullable=False,
+    ),
+    sa.Column("ics_uid", sa.Text, nullable=False),
+    sa.Column("title", sa.Text),
+    sa.Column("description", sa.Text),
+    sa.Column("start_time", sa.DateTime(timezone=True), nullable=False),
+    sa.Column("end_time", sa.DateTime(timezone=True), nullable=False),
+    sa.Column("attendees", JSONB),
+    sa.Column("location", sa.Text),
+    sa.Column("ics_raw_data", sa.Text),
+    sa.Column("last_synced", sa.DateTime(timezone=True), nullable=False),
+    sa.Column("is_deleted", sa.Boolean, nullable=False, server_default=sa.false()),
+    sa.Column("created_at", sa.DateTime(timezone=True), nullable=False),
+    sa.Column("updated_at", sa.DateTime(timezone=True), nullable=False),
+    sa.UniqueConstraint("room_id", "ics_uid", name="uq_room_calendar_event"),
+    sa.Index("idx_calendar_event_room_start", "room_id", "start_time"),
+    sa.Index(
+        "idx_calendar_event_deleted",
+        "is_deleted",
+        postgresql_where=sa.text("NOT is_deleted"),
+    ),
+)
+
+
+class CalendarEvent(BaseModel):
+    id: str = Field(default_factory=generate_uuid4)
+    room_id: str
+    ics_uid: str
+    title: str | None = None
+    description: str | None = None
+    start_time: datetime
+    end_time: datetime
+    attendees: list[dict[str, Any]] | None = None
+    location: str | None = None
+    ics_raw_data: str | None = None
+    last_synced: datetime = Field(default_factory=lambda: datetime.now(timezone.utc))
+    is_deleted: bool = False
+    created_at: datetime = Field(default_factory=lambda: datetime.now(timezone.utc))
+    updated_at: datetime = Field(default_factory=lambda: datetime.now(timezone.utc))
+
+
+class CalendarEventController:
+    async def get_by_room(
+        self,
+        room_id: str,
+        include_deleted: bool = False,
+        start_after: datetime | None = None,
+        end_before: datetime | None = None,
+    ) -> list[CalendarEvent]:
+        """Get calendar events for a room."""
+        query = calendar_events.select().where(calendar_events.c.room_id == room_id)
+
+        if not include_deleted:
+            query = query.where(calendar_events.c.is_deleted == False)
+
+        if start_after:
+            query = query.where(calendar_events.c.start_time >= start_after)
+
+        if end_before:
+            query = query.where(calendar_events.c.end_time <= end_before)
+
+        query = query.order_by(calendar_events.c.start_time.asc())
+
+        results = await get_database().fetch_all(query)
+        return [CalendarEvent(**result) for result in results]
+
+    async def get_upcoming(
+        self, room_id: str, minutes_ahead: int = 30
+    ) -> list[CalendarEvent]:
+        """Get upcoming events for a room within the specified minutes."""
+        now = datetime.now(timezone.utc)
+        future_time = now + timedelta(minutes=minutes_ahead)
+
+        query = (
+            calendar_events.select()
+            .where(
+                sa.and_(
+                    calendar_events.c.room_id == room_id,
+                    calendar_events.c.is_deleted == False,
+                    calendar_events.c.start_time >= now,
+                    calendar_events.c.start_time <= future_time,
+                )
+            )
+            .order_by(calendar_events.c.start_time.asc())
+        )
+
+        results = await get_database().fetch_all(query)
+        return [CalendarEvent(**result) for result in results]
+
+    async def get_by_ics_uid(self, room_id: str, ics_uid: str) -> CalendarEvent | None:
+        """Get a calendar event by its ICS UID."""
+        query = calendar_events.select().where(
+            sa.and_(
+                calendar_events.c.room_id == room_id,
+                calendar_events.c.ics_uid == ics_uid,
+            )
+        )
+        result = await get_database().fetch_one(query)
+        return CalendarEvent(**result) if result else None
+
+    async def upsert(self, event: CalendarEvent) -> CalendarEvent:
+        """Create or update a calendar event."""
+        existing = await self.get_by_ics_uid(event.room_id, event.ics_uid)
+
+        if existing:
+            # Update existing event
+            event.id = existing.id
+            event.created_at = existing.created_at
+            event.updated_at = datetime.now(timezone.utc)
+
+            query = (
+                calendar_events.update()
+                .where(calendar_events.c.id == existing.id)
+                .values(**event.model_dump())
+            )
+        else:
+            # Insert new event
+            query = calendar_events.insert().values(**event.model_dump())
+
+        await get_database().execute(query)
+        return event
+
+    async def soft_delete_missing(
+        self, room_id: str, current_ics_uids: list[str]
+    ) -> int:
+        """Soft delete future events that are no longer in the calendar."""
+        now = datetime.now(timezone.utc)
+
+        # First, get the IDs of events to delete
+        select_query = calendar_events.select().where(
+            sa.and_(
+                calendar_events.c.room_id == room_id,
+                calendar_events.c.start_time > now,
+                calendar_events.c.is_deleted == False,
+                calendar_events.c.ics_uid.notin_(current_ics_uids)
+                if current_ics_uids
+                else True,
+            )
+        )
+
+        to_delete = await get_database().fetch_all(select_query)
+        delete_count = len(to_delete)
+
+        if delete_count > 0:
+            # Now update them
+            update_query = (
+                calendar_events.update()
+                .where(
+                    sa.and_(
+                        calendar_events.c.room_id == room_id,
+                        calendar_events.c.start_time > now,
+                        calendar_events.c.is_deleted == False,
+                        calendar_events.c.ics_uid.notin_(current_ics_uids)
+                        if current_ics_uids
+                        else True,
+                    )
+                )
+                .values(is_deleted=True, updated_at=now)
+            )
+
+            await get_database().execute(update_query)
+
+        return delete_count
+
+    async def delete_by_room(self, room_id: str) -> int:
+        """Hard delete all events for a room (used when room is deleted)."""
+        query = calendar_events.delete().where(calendar_events.c.room_id == room_id)
+        result = await get_database().execute(query)
+        return result.rowcount
+
+
+# Add missing import
+from datetime import timedelta
+
+calendar_events_controller = CalendarEventController()
--- a/server/reflector/db/meetings.py
+++ b/server/reflector/db/meetings.py
@@ -1,11 +1,13 @@
-from datetime import datetime, timezone
-from typing import Any, Dict, List, Literal
+from datetime import datetime
+from typing import Any, Literal

 import sqlalchemy as sa
+from fastapi import HTTPException
 from pydantic import BaseModel, Field
+from sqlalchemy.dialects.postgresql import JSONB

 from reflector.db import get_database, metadata
-from reflector.db.rooms import Room, VideoPlatform
+from reflector.db.rooms import Room
 from reflector.utils import generate_uuid4

 meetings = sa.Table(
@@ -17,12 +19,8 @@ meetings = sa.Table(
    sa.Column("host_room_url", sa.String),
    sa.Column("start_date", sa.DateTime(timezone=True)),
    sa.Column("end_date", sa.DateTime(timezone=True)),
-    sa.Column(
-        "room_id",
-        sa.String,
-        sa.ForeignKey("room.id", ondelete="CASCADE"),
-        nullable=True,
-    ),
+    sa.Column("user_id", sa.String),
+    sa.Column("room_id", sa.String),
    sa.Column("is_locked", sa.Boolean, nullable=False, server_default=sa.false()),
    sa.Column("room_mode", sa.String, nullable=False, server_default="normal"),
    sa.Column("recording_type", sa.String, nullable=False, server_default="cloud"),
@@ -44,27 +42,23 @@ meetings = sa.Table(
        nullable=False,
        server_default=sa.true(),
    ),
-    sa.Column("platform", sa.String, nullable=False, server_default="whereby"),
-    sa.Column("events", sa.JSON, nullable=False, server_default=sa.text("'[]'")),
-    sa.Index("idx_meeting_room_id", "room_id"),
-    sa.Index(
-        "idx_one_active_meeting_per_room",
-        "room_id",
-        unique=True,
-        postgresql_where=sa.text("is_active = true"),
+    sa.Column(
+        "calendar_event_id",
+        sa.String,
+        sa.ForeignKey("calendar_event.id", ondelete="SET NULL"),
    ),
+    sa.Column("calendar_metadata", JSONB),
+    sa.Column("last_participant_left_at", sa.DateTime(timezone=True)),
+    sa.Column("grace_period_minutes", sa.Integer, server_default=sa.text("15")),
+    sa.Index("idx_meeting_room_id", "room_id"),
+    sa.Index("idx_meeting_calendar_event", "calendar_event_id"),
 )

 meeting_consent = sa.Table(
    "meeting_consent",
    metadata,
    sa.Column("id", sa.String, primary_key=True),
-    sa.Column(
-        "meeting_id",
-        sa.String,
-        sa.ForeignKey("meeting.id", ondelete="CASCADE"),
-        nullable=False,
-    ),
+    sa.Column("meeting_id", sa.String, sa.ForeignKey("meeting.id"), nullable=False),
    sa.Column("user_id", sa.String),
    sa.Column("consent_given", sa.Boolean, nullable=False),
    sa.Column("consent_timestamp", sa.DateTime(timezone=True), nullable=False),
@@ -86,7 +80,8 @@ class Meeting(BaseModel):
    host_room_url: str
    start_date: datetime
    end_date: datetime
-    room_id: str | None
+    user_id: str | None = None
+    room_id: str | None = None
    is_locked: bool = False
    room_mode: Literal["normal", "group"] = "normal"
    recording_type: Literal["none", "local", "cloud"] = "cloud"
@@ -94,8 +89,11 @@ class Meeting(BaseModel):
        "none", "prompt", "automatic", "automatic-2nd-participant"
    ] = "automatic-2nd-participant"
    num_clients: int = 0
-    platform: VideoPlatform = VideoPlatform.WHEREBY
-    events: List[Dict[str, Any]] = Field(default_factory=list)
+    is_active: bool = True
+    calendar_event_id: str | None = None
+    calendar_metadata: dict[str, Any] | None = None
+    last_participant_left_at: datetime | None = None
+    grace_period_minutes: int = 15


 class MeetingController:
@@ -107,8 +105,14 @@ class MeetingController:
        host_room_url: str,
        start_date: datetime,
        end_date: datetime,
+        user_id: str,
        room: Room,
+        calendar_event_id: str | None = None,
+        calendar_metadata: dict[str, Any] | None = None,
    ):
+        """
+        Create a new meeting
+        """
        meeting = Meeting(
            id=id,
            room_name=room_name,
@@ -116,25 +120,33 @@ class MeetingController:
            host_room_url=host_room_url,
            start_date=start_date,
            end_date=end_date,
+            user_id=user_id,
            room_id=room.id,
            is_locked=room.is_locked,
            room_mode=room.room_mode,
            recording_type=room.recording_type,
            recording_trigger=room.recording_trigger,
-            platform=room.platform,
+            calendar_event_id=calendar_event_id,
+            calendar_metadata=calendar_metadata,
        )
        query = meetings.insert().values(**meeting.model_dump())
        await get_database().execute(query)
        return meeting

    async def get_all_active(self) -> list[Meeting]:
+        """
+        Get active meetings.
+        """
        query = meetings.select().where(meetings.c.is_active)
        return await get_database().fetch_all(query)

    async def get_by_room_name(
        self,
        room_name: str,
-    ) -> Meeting | None:
+    ) -> Meeting:
+        """
+        Get a meeting by room name.
+        """
        query = meetings.select().where(meetings.c.room_name == room_name)
        result = await get_database().fetch_one(query)
        if not result:
@@ -142,7 +154,11 @@ class MeetingController:

        return Meeting(**result)

-    async def get_active(self, room: Room, current_time: datetime) -> Meeting | None:
+    async def get_active(self, room: Room, current_time: datetime) -> Meeting:
+        """
+        Get latest active meeting for a room.
+        For backward compatibility, returns the most recent active meeting.
+        """
        end_date = getattr(meetings.c, "end_date")
        query = (
            meetings.select()
@@ -161,79 +177,87 @@ class MeetingController:

        return Meeting(**result)

+    async def get_all_active_for_room(
+        self, room: Room, current_time: datetime
+    ) -> list[Meeting]:
+        """
+        Get all active meetings for a room.
+        This supports multiple concurrent meetings per room.
+        """
+        end_date = getattr(meetings.c, "end_date")
+        query = (
+            meetings.select()
+            .where(
+                sa.and_(
+                    meetings.c.room_id == room.id,
+                    meetings.c.end_date > current_time,
+                    meetings.c.is_active,
+                )
+            )
+            .order_by(end_date.desc())
+        )
+        results = await get_database().fetch_all(query)
+        return [Meeting(**result) for result in results]
+
+    async def get_active_by_calendar_event(
+        self, room: Room, calendar_event_id: str, current_time: datetime
+    ) -> Meeting | None:
+        """
+        Get active meeting for a specific calendar event.
+        """
+        query = meetings.select().where(
+            sa.and_(
+                meetings.c.room_id == room.id,
+                meetings.c.calendar_event_id == calendar_event_id,
+                meetings.c.end_date > current_time,
+                meetings.c.is_active,
+            )
+        )
+        result = await get_database().fetch_one(query)
+        if not result:
+            return None
+        return Meeting(**result)
+
    async def get_by_id(self, meeting_id: str, **kwargs) -> Meeting | None:
+        """
+        Get a meeting by id
+        """
        query = meetings.select().where(meetings.c.id == meeting_id)
        result = await get_database().fetch_one(query)
        if not result:
            return None
        return Meeting(**result)

+    async def get_by_id_for_http(self, meeting_id: str, user_id: str | None) -> Meeting:
+        """
+        Get a meeting by ID for HTTP request.
+
+        If not found, it will raise a 404 error.
+        """
+        query = meetings.select().where(meetings.c.id == meeting_id)
+        result = await get_database().fetch_one(query)
+        if not result:
+            raise HTTPException(status_code=404, detail="Meeting not found")
+
+        meeting = Meeting(**result)
+        if result["user_id"] != user_id:
+            meeting.host_room_url = ""
+
+        return meeting
+
+    async def get_by_calendar_event(self, calendar_event_id: str) -> Meeting | None:
+        query = meetings.select().where(
+            meetings.c.calendar_event_id == calendar_event_id
+        )
+        result = await get_database().fetch_one(query)
+        if not result:
+            return None
+        return Meeting(**result)
+
    async def update_meeting(self, meeting_id: str, **kwargs):
        query = meetings.update().where(meetings.c.id == meeting_id).values(**kwargs)
        await get_database().execute(query)

-    async def add_event(
-        self, meeting_id: str, event_type: str, event_data: Dict[str, Any] = None
-    ):
-        """Add an event to a meeting's events list."""
-        if event_data is None:
-            event_data = {}
-
-        event = {
-            "type": event_type,
-            "timestamp": datetime.now(tz=timezone.utc).isoformat(),
-            "data": event_data,
-        }
-
-        # Get current events
-        query = meetings.select().where(meetings.c.id == meeting_id)
-        result = await get_database().fetch_one(query)
-        if not result:
-            return
-
-        current_events = result["events"] or []
-        current_events.append(event)
-
-        # Update with new events list
-        update_query = (
-            meetings.update()
-            .where(meetings.c.id == meeting_id)
-            .values(events=current_events)
-        )
-        await get_database().execute(update_query)
-
-    async def participant_joined(
-        self, meeting_id: str, participant_data: Dict[str, Any] = None
-    ):
-        """Record a participant joined event."""
-        await self.add_event(meeting_id, "participant_joined", participant_data)
-
-    async def participant_left(
-        self, meeting_id: str, participant_data: Dict[str, Any] = None
-    ):
-        """Record a participant left event."""
-        await self.add_event(meeting_id, "participant_left", participant_data)
-
-    async def recording_started(
-        self, meeting_id: str, recording_data: Dict[str, Any] = None
-    ):
-        """Record a recording started event."""
-        await self.add_event(meeting_id, "recording_started", recording_data)
-
-    async def recording_stopped(
-        self, meeting_id: str, recording_data: Dict[str, Any] = None
-    ):
-        """Record a recording stopped event."""
-        await self.add_event(meeting_id, "recording_stopped", recording_data)
-
-    async def get_events(self, meeting_id: str) -> List[Dict[str, Any]]:
-        """Get all events for a meeting."""
-        query = meetings.select().where(meetings.c.id == meeting_id)
-        result = await get_database().fetch_one(query)
-        if not result:
-            return []
-        return result["events"] or []
-

 class MeetingConsentController:
    async def get_by_meeting_id(self, meeting_id: str) -> list[MeetingConsent]:
@@ -254,7 +278,7 @@ class MeetingConsentController:
        result = await get_database().fetch_one(query)
        if result is None:
            return None
-        return MeetingConsent(**result)
+        return MeetingConsent(**result) if result else None

    async def upsert(self, consent: MeetingConsent) -> MeetingConsent:
        """Create new consent or update existing one for authenticated users"""
--- a/server/reflector/db/rooms.py
+++ b/server/reflector/db/rooms.py
@@ -1,6 +1,4 @@
-import secrets
 from datetime import datetime, timezone
-from enum import StrEnum
 from sqlite3 import IntegrityError
 from typing import Literal

@@ -12,12 +10,6 @@ from sqlalchemy.sql import false, or_
 from reflector.db import get_database, metadata
 from reflector.utils import generate_uuid4

-
-class VideoPlatform(StrEnum):
-    WHEREBY = "whereby"
-    JITSI = "jitsi"
-
-
 rooms = sqlalchemy.Table(
    "room",
    metadata,
@@ -48,12 +40,15 @@ rooms = sqlalchemy.Table(
    sqlalchemy.Column(
        "is_shared", sqlalchemy.Boolean, nullable=False, server_default=false()
    ),
-    sqlalchemy.Column("webhook_url", sqlalchemy.String, nullable=True),
-    sqlalchemy.Column("webhook_secret", sqlalchemy.String, nullable=True),
+    sqlalchemy.Column("ics_url", sqlalchemy.Text),
+    sqlalchemy.Column("ics_fetch_interval", sqlalchemy.Integer, server_default="300"),
    sqlalchemy.Column(
-        "platform", sqlalchemy.String, nullable=False, server_default="whereby"
+        "ics_enabled", sqlalchemy.Boolean, nullable=False, server_default=false()
    ),
+    sqlalchemy.Column("ics_last_sync", sqlalchemy.DateTime(timezone=True)),
+    sqlalchemy.Column("ics_last_etag", sqlalchemy.Text),
    sqlalchemy.Index("idx_room_is_shared", "is_shared"),
+    sqlalchemy.Index("idx_room_ics_enabled", "ics_enabled"),
 )


@@ -72,9 +67,11 @@ class Room(BaseModel):
        "none", "prompt", "automatic", "automatic-2nd-participant"
    ] = "automatic-2nd-participant"
    is_shared: bool = False
-    webhook_url: str | None = None
-    webhook_secret: str | None = None
-    platform: VideoPlatform = VideoPlatform.WHEREBY
+    ics_url: str | None = None
+    ics_fetch_interval: int = 300
+    ics_enabled: bool = False
+    ics_last_sync: datetime | None = None
+    ics_last_etag: str | None = None


 class RoomController:
@@ -123,16 +120,13 @@ class RoomController:
        recording_type: str,
        recording_trigger: str,
        is_shared: bool,
-        webhook_url: str = "",
-        webhook_secret: str = "",
-        platform: str = "whereby",
+        ics_url: str | None = None,
+        ics_fetch_interval: int = 300,
+        ics_enabled: bool = False,
    ):
        """
        Add a new room
        """
-        if webhook_url and not webhook_secret:
-            webhook_secret = secrets.token_urlsafe(32)
-
        room = Room(
            name=name,
            user_id=user_id,
@@ -144,9 +138,9 @@ class RoomController:
            recording_type=recording_type,
            recording_trigger=recording_trigger,
            is_shared=is_shared,
-            webhook_url=webhook_url,
-            webhook_secret=webhook_secret,
-            platform=platform,
+            ics_url=ics_url,
+            ics_fetch_interval=ics_fetch_interval,
+            ics_enabled=ics_enabled,
        )
        query = rooms.insert().values(**room.model_dump())
        try:
@@ -159,9 +153,6 @@ class RoomController:
        """
        Update a room fields with key/values in values
        """
-        if values.get("webhook_url") and not values.get("webhook_secret"):
-            values["webhook_secret"] = secrets.token_urlsafe(32)
-
        query = rooms.update().where(rooms.c.id == room.id).values(**values)
        try:
            await get_database().execute(query)
--- a/server/reflector/db/search.py
+++ b/server/reflector/db/search.py
@@ -1,38 +1,22 @@
 """Search functionality for transcripts and other entities."""

-import itertools
-from dataclasses import dataclass
 from datetime import datetime
 from io import StringIO
-from typing import Annotated, Any, Dict, Iterator
+from typing import Annotated, Any, Dict

 import sqlalchemy
 import webvtt
-from databases.interfaces import Record as DbRecord
-from fastapi import HTTPException
-from pydantic import (
-    BaseModel,
-    Field,
-    NonNegativeFloat,
-    NonNegativeInt,
-    TypeAdapter,
-    ValidationError,
-    constr,
-    field_serializer,
-)
+from pydantic import BaseModel, Field, constr, field_serializer

 from reflector.db import get_database
-from reflector.db.rooms import rooms
-from reflector.db.transcripts import SourceKind, TranscriptStatus, transcripts
+from reflector.db.transcripts import SourceKind, transcripts
 from reflector.db.utils import is_postgresql
 from reflector.logger import logger
-from reflector.utils.string import NonEmptyString, try_parse_non_empty_string

 DEFAULT_SEARCH_LIMIT = 20
 SNIPPET_CONTEXT_LENGTH = 50  # Characters before/after match to include
-DEFAULT_SNIPPET_MAX_LENGTH = NonNegativeInt(150)
-DEFAULT_MAX_SNIPPETS = NonNegativeInt(3)
-LONG_SUMMARY_MAX_SNIPPETS = 2
+DEFAULT_SNIPPET_MAX_LENGTH = 150
+DEFAULT_MAX_SNIPPETS = 3

 SearchQueryBase = constr(min_length=1, strip_whitespace=True)
 SearchLimitBase = Annotated[int, Field(ge=1, le=100)]
@@ -40,7 +24,6 @@ SearchOffsetBase = Annotated[int, Field(ge=0)]
 SearchTotalBase = Annotated[int, Field(ge=0)]

 SearchQuery = Annotated[SearchQueryBase, Field(description="Search query text")]
-search_query_adapter = TypeAdapter(SearchQuery)
 SearchLimit = Annotated[SearchLimitBase, Field(description="Results per page")]
 SearchOffset = Annotated[
    SearchOffsetBase, Field(description="Number of results to skip")
@@ -49,92 +32,15 @@ SearchTotal = Annotated[
    SearchTotalBase, Field(description="Total number of search results")
 ]

-WEBVTT_SPEC_HEADER = "WEBVTT"
-
-WebVTTContent = Annotated[
-    str,
-    Field(min_length=len(WEBVTT_SPEC_HEADER), description="WebVTT content"),
-]
-
-
-class WebVTTProcessor:
-    """Stateless processor for WebVTT content operations."""
-
-    @staticmethod
-    def parse(raw_content: str) -> WebVTTContent:
-        """Parse WebVTT content and return it as a string."""
-        if not raw_content.startswith(WEBVTT_SPEC_HEADER):
-            raise ValueError(f"Invalid WebVTT content, no header {WEBVTT_SPEC_HEADER}")
-        return raw_content
-
-    @staticmethod
-    def extract_text(webvtt_content: WebVTTContent) -> str:
-        """Extract plain text from WebVTT content using webvtt library."""
-        try:
-            buffer = StringIO(webvtt_content)
-            vtt = webvtt.read_buffer(buffer)
-            return " ".join(caption.text for caption in vtt if caption.text)
-        except webvtt.errors.MalformedFileError as e:
-            logger.warning(f"Malformed WebVTT content: {e}")
-            return ""
-        except (UnicodeDecodeError, ValueError) as e:
-            logger.warning(f"Failed to decode WebVTT content: {e}")
-            return ""
-        except AttributeError as e:
-            logger.error(
-                f"WebVTT parsing error - unexpected format: {e}", exc_info=True
-            )
-            return ""
-        except Exception as e:
-            logger.error(f"Unexpected error parsing WebVTT: {e}", exc_info=True)
-            return ""
-
-    @staticmethod
-    def generate_snippets(
-        webvtt_content: WebVTTContent,
-        query: SearchQuery,
-        max_snippets: NonNegativeInt = DEFAULT_MAX_SNIPPETS,
-    ) -> list[str]:
-        """Generate snippets from WebVTT content."""
-        return SnippetGenerator.generate(
-            WebVTTProcessor.extract_text(webvtt_content),
-            query,
-            max_snippets=max_snippets,
-        )
-
-
-@dataclass(frozen=True)
-class SnippetCandidate:
-    """Represents a candidate snippet with its position."""
-
-    _text: str
-    start: NonNegativeInt
-    _original_text_length: int
-
-    @property
-    def end(self) -> NonNegativeInt:
-        """Calculate end position from start and raw text length."""
-        return self.start + len(self._text)
-
-    def text(self) -> str:
-        """Get display text with ellipses added if needed."""
-        result = self._text.strip()
-        if self.start > 0:
-            result = "..." + result
-        if self.end < self._original_text_length:
-            result = result + "..."
-        return result
-

 class SearchParameters(BaseModel):
    """Validated search parameters for full-text search."""

-    query_text: SearchQuery | None = None
+    query_text: SearchQuery
    limit: SearchLimit = DEFAULT_SEARCH_LIMIT
    offset: SearchOffset = 0
    user_id: str | None = None
    room_id: str | None = None
-    source_kind: SourceKind | None = None


 class SearchResultDB(BaseModel):
@@ -158,18 +64,13 @@ class SearchResult(BaseModel):
    title: str | None = None
    user_id: str | None = None
    room_id: str | None = None
-    room_name: str | None = None
-    source_kind: SourceKind
    created_at: datetime
-    status: TranscriptStatus = Field(..., min_length=1)
+    status: str = Field(..., min_length=1)
    rank: float = Field(..., ge=0, le=1)
-    duration: NonNegativeFloat | None = Field(..., description="Duration in seconds")
+    duration: float | None = Field(..., ge=0, description="Duration in seconds")
    search_snippets: list[str] = Field(
        description="Text snippets around search matches"
    )
-    total_match_count: NonNegativeInt = Field(
-        default=0, description="Total number of matches found in the transcript"
-    )

    @field_serializer("created_at", when_used="json")
    def serialize_datetime(self, dt: datetime) -> str:
@@ -178,157 +79,84 @@ class SearchResult(BaseModel):
        return dt.isoformat()


-class SnippetGenerator:
-    """Stateless generator for text snippets and match operations."""
-
-    @staticmethod
-    def find_all_matches(text: str, query: str) -> Iterator[int]:
-        """Generate all match positions for a query in text."""
-        if not text:
-            logger.warning("Empty text for search query in find_all_matches")
-            return
-        if not query:
-            logger.warning("Empty query for search text in find_all_matches")
-            return
-
-        text_lower = text.lower()
-        query_lower = query.lower()
-        start = 0
-        prev_start = start
-        while (pos := text_lower.find(query_lower, start)) != -1:
-            yield pos
-            start = pos + len(query_lower)
-            if start <= prev_start:
-                raise ValueError("panic! find_all_matches is not incremental")
-            prev_start = start
-
-    @staticmethod
-    def count_matches(text: str, query: SearchQuery) -> NonNegativeInt:
-        """Count total number of matches for a query in text."""
-        ZERO = NonNegativeInt(0)
-        if not text:
-            logger.warning("Empty text for search query in count_matches")
-            return ZERO
-        assert query is not None
-        return NonNegativeInt(
-            sum(1 for _ in SnippetGenerator.find_all_matches(text, query))
-        )
-
-    @staticmethod
-    def create_snippet(
-        text: str, match_pos: int, max_length: int = DEFAULT_SNIPPET_MAX_LENGTH
-    ) -> SnippetCandidate:
-        """Create a snippet from a match position."""
-        snippet_start = NonNegativeInt(max(0, match_pos - SNIPPET_CONTEXT_LENGTH))
-        snippet_end = min(len(text), match_pos + max_length - SNIPPET_CONTEXT_LENGTH)
-
-        snippet_text = text[snippet_start:snippet_end]
-
-        return SnippetCandidate(
-            _text=snippet_text, start=snippet_start, _original_text_length=len(text)
-        )
-
-    @staticmethod
-    def filter_non_overlapping(
-        candidates: Iterator[SnippetCandidate],
-    ) -> Iterator[str]:
-        """Filter out overlapping snippets and return only display text."""
-        last_end = 0
-        for candidate in candidates:
-            display_text = candidate.text()
-            # it means that next overlapping snippets simply don't get included
-            # it's fine as simplistic logic and users probably won't care much because they already have their search results just fin
-            if candidate.start >= last_end and display_text:
-                yield display_text
-                last_end = candidate.end
-
-    @staticmethod
-    def generate(
-        text: str,
-        query: SearchQuery,
-        max_length: NonNegativeInt = DEFAULT_SNIPPET_MAX_LENGTH,
-        max_snippets: NonNegativeInt = DEFAULT_MAX_SNIPPETS,
-    ) -> list[str]:
-        """Generate snippets from text."""
-        assert query is not None
-        if not text:
-            logger.warning("Empty text for generate_snippets")
-            return []
-
-        candidates = (
-            SnippetGenerator.create_snippet(text, pos, max_length)
-            for pos in SnippetGenerator.find_all_matches(text, query)
-        )
-        filtered = SnippetGenerator.filter_non_overlapping(candidates)
-        snippets = list(itertools.islice(filtered, max_snippets))
-
-        # Fallback to first word search if no full matches
-        # it's another assumption: proper snippet logic generation is quite complicated and tied to db logic, so simplification is used here
-        if not snippets and " " in query:
-            first_word = query.split()[0]
-            return SnippetGenerator.generate(text, first_word, max_length, max_snippets)
-
-        return snippets
-
-    @staticmethod
-    def from_summary(
-        summary: str,
-        query: SearchQuery,
-        max_snippets: NonNegativeInt = LONG_SUMMARY_MAX_SNIPPETS,
-    ) -> list[str]:
-        """Generate snippets from summary text."""
-        return SnippetGenerator.generate(summary, query, max_snippets=max_snippets)
-
-    @staticmethod
-    def combine_sources(
-        summary: NonEmptyString | None,
-        webvtt: WebVTTContent | None,
-        query: SearchQuery,
-        max_total: NonNegativeInt = DEFAULT_MAX_SNIPPETS,
-    ) -> tuple[list[str], NonNegativeInt]:
-        """Combine snippets from multiple sources and return total match count.
-
-        Returns (snippets, total_match_count) tuple.
-
-        snippets can be empty for real in case of e.g. title match
-        """
-
-        assert (
-            summary is not None or webvtt is not None
-        ), "At least one source must be present"
-
-        webvtt_matches = 0
-        summary_matches = 0
-
-        if webvtt:
-            webvtt_text = WebVTTProcessor.extract_text(webvtt)
-            webvtt_matches = SnippetGenerator.count_matches(webvtt_text, query)
-
-        if summary:
-            summary_matches = SnippetGenerator.count_matches(summary, query)
-
-        total_matches = NonNegativeInt(webvtt_matches + summary_matches)
-
-        summary_snippets = (
-            SnippetGenerator.from_summary(summary, query) if summary else []
-        )
-
-        if len(summary_snippets) >= max_total:
-            return summary_snippets[:max_total], total_matches
-
-        remaining = max_total - len(summary_snippets)
-        webvtt_snippets = (
-            WebVTTProcessor.generate_snippets(webvtt, query, remaining)
-            if webvtt
-            else []
-        )
-
-        return summary_snippets + webvtt_snippets, total_matches
-
-
 class SearchController:
    """Controller for search operations across different entities."""

+    @staticmethod
+    def _extract_webvtt_text(webvtt_content: str) -> str:
+        """Extract plain text from WebVTT content using webvtt library."""
+        if not webvtt_content:
+            return ""
+
+        try:
+            buffer = StringIO(webvtt_content)
+            vtt = webvtt.read_buffer(buffer)
+            return " ".join(caption.text for caption in vtt if caption.text)
+        except (webvtt.errors.MalformedFileError, UnicodeDecodeError, ValueError) as e:
+            logger.warning(f"Failed to parse WebVTT content: {e}", exc_info=e)
+            return ""
+        except AttributeError as e:
+            logger.warning(f"WebVTT parsing error - unexpected format: {e}", exc_info=e)
+            return ""
+
+    @staticmethod
+    def _generate_snippets(
+        text: str,
+        q: SearchQuery,
+        max_length: int = DEFAULT_SNIPPET_MAX_LENGTH,
+        max_snippets: int = DEFAULT_MAX_SNIPPETS,
+    ) -> list[str]:
+        """Generate multiple snippets around all occurrences of search term."""
+        if not text or not q:
+            return []
+
+        snippets = []
+        lower_text = text.lower()
+        search_lower = q.lower()
+
+        last_snippet_end = 0
+        start_pos = 0
+
+        while len(snippets) < max_snippets:
+            match_pos = lower_text.find(search_lower, start_pos)
+
+            if match_pos == -1:
+                if not snippets and search_lower.split():
+                    first_word = search_lower.split()[0]
+                    match_pos = lower_text.find(first_word, start_pos)
+                    if match_pos == -1:
+                        break
+                else:
+                    break
+
+            snippet_start = max(0, match_pos - SNIPPET_CONTEXT_LENGTH)
+            snippet_end = min(
+                len(text), match_pos + max_length - SNIPPET_CONTEXT_LENGTH
+            )
+
+            if snippet_start < last_snippet_end:
+                start_pos = match_pos + len(search_lower)
+                continue
+
+            snippet = text[snippet_start:snippet_end]
+
+            if snippet_start > 0:
+                snippet = "..." + snippet
+            if snippet_end < len(text):
+                snippet = snippet + "..."
+
+            snippet = snippet.strip()
+
+            if snippet:
+                snippets.append(snippet)
+                last_snippet_end = snippet_end
+
+            start_pos = match_pos + len(search_lower)
+            if start_pos >= len(text):
+                break
+
+        return snippets
+
    @classmethod
    async def search_transcripts(
        cls, params: SearchParameters
@@ -344,72 +172,39 @@ class SearchController:
            )
            return [], 0

-        base_columns = [
-            transcripts.c.id,
-            transcripts.c.title,
-            transcripts.c.created_at,
-            transcripts.c.duration,
-            transcripts.c.status,
-            transcripts.c.user_id,
-            transcripts.c.room_id,
-            transcripts.c.source_kind,
-            transcripts.c.webvtt,
-            transcripts.c.long_summary,
-            sqlalchemy.case(
-                (
-                    transcripts.c.room_id.isnot(None) & rooms.c.id.is_(None),
-                    "Deleted Room",
-                ),
-                else_=rooms.c.name,
-            ).label("room_name"),
-        ]
-        search_query = None
-        if params.query_text is not None:
-            search_query = sqlalchemy.func.websearch_to_tsquery(
-                "english", params.query_text
-            )
-            rank_column = sqlalchemy.func.ts_rank(
-                transcripts.c.search_vector_en,
-                search_query,
-                32,  # normalization flag: rank/(rank+1) for 0-1 range
-            ).label("rank")
-        else:
-            rank_column = sqlalchemy.cast(1.0, sqlalchemy.Float).label("rank")
-
-        columns = base_columns + [rank_column]
-        base_query = sqlalchemy.select(columns).select_from(
-            transcripts.join(rooms, transcripts.c.room_id == rooms.c.id, isouter=True)
+        search_query = sqlalchemy.func.websearch_to_tsquery(
+            "english", params.query_text
        )

-        if params.query_text is not None:
-            # because already initialized based on params.query_text presence above
-            assert search_query is not None
-            base_query = base_query.where(
-                transcripts.c.search_vector_en.op("@@")(search_query)
-            )
+        base_query = sqlalchemy.select(
+            [
+                transcripts.c.id,
+                transcripts.c.title,
+                transcripts.c.created_at,
+                transcripts.c.duration,
+                transcripts.c.status,
+                transcripts.c.user_id,
+                transcripts.c.room_id,
+                transcripts.c.source_kind,
+                transcripts.c.webvtt,
+                sqlalchemy.func.ts_rank(
+                    transcripts.c.search_vector_en,
+                    search_query,
+                    32,  # normalization flag: rank/(rank+1) for 0-1 range
+                ).label("rank"),
+            ]
+        ).where(transcripts.c.search_vector_en.op("@@")(search_query))

        if params.user_id:
-            base_query = base_query.where(
-                sqlalchemy.or_(
-                    transcripts.c.user_id == params.user_id, rooms.c.is_shared
-                )
-            )
-        else:
-            base_query = base_query.where(rooms.c.is_shared)
+            base_query = base_query.where(transcripts.c.user_id == params.user_id)
        if params.room_id:
            base_query = base_query.where(transcripts.c.room_id == params.room_id)
-        if params.source_kind:
-            base_query = base_query.where(
-                transcripts.c.source_kind == params.source_kind
-            )
-
-        if params.query_text is not None:
-            order_by = sqlalchemy.desc(sqlalchemy.text("rank"))
-        else:
-            order_by = sqlalchemy.desc(transcripts.c.created_at)
-
-        query = base_query.order_by(order_by).limit(params.limit).offset(params.offset)

+        query = (
+            base_query.order_by(sqlalchemy.desc(sqlalchemy.text("rank")))
+            .limit(params.limit)
+            .offset(params.offset)
+        )
        rs = await get_database().fetch_all(query)

        count_query = sqlalchemy.select([sqlalchemy.func.count()]).select_from(
@@ -417,52 +212,20 @@ class SearchController:
        )
        total = await get_database().fetch_val(count_query)

-        def _process_result(r: DbRecord) -> SearchResult:
+        def _process_result(r) -> SearchResult:
            r_dict: Dict[str, Any] = dict(r)
-
-            webvtt_raw: str | None = r_dict.pop("webvtt", None)
-            webvtt: WebVTTContent | None
-            if webvtt_raw:
-                webvtt = WebVTTProcessor.parse(webvtt_raw)
-            else:
-                webvtt = None
-
-            long_summary_r: str | None = r_dict.pop("long_summary", None)
-            long_summary: NonEmptyString = try_parse_non_empty_string(long_summary_r)
-            room_name: str | None = r_dict.pop("room_name", None)
+            webvtt: str | None = r_dict.pop("webvtt", None)
            db_result = SearchResultDB.model_validate(r_dict)

-            at_least_one_source = webvtt is not None or long_summary is not None
-            has_query = params.query_text is not None
-            snippets, total_match_count = (
-                SnippetGenerator.combine_sources(
-                    long_summary, webvtt, params.query_text, DEFAULT_MAX_SNIPPETS
-                )
-                if has_query and at_least_one_source
-                else ([], 0)
-            )
+            snippets = []
+            if webvtt:
+                plain_text = cls._extract_webvtt_text(webvtt)
+                snippets = cls._generate_snippets(plain_text, params.query_text)

-            return SearchResult(
-                **db_result.model_dump(),
-                room_name=room_name,
-                search_snippets=snippets,
-                total_match_count=total_match_count,
-            )
-
-        try:
-            results = [_process_result(r) for r in rs]
-        except ValidationError as e:
-            logger.error(f"Invalid search result data: {e}", exc_info=True)
-            raise HTTPException(
-                status_code=500, detail="Internal search result data consistency error"
-            )
-        except Exception as e:
-            logger.error(f"Error processing search results: {e}", exc_info=True)
-            raise
+            return SearchResult(**db_result.model_dump(), search_snippets=snippets)

+        results = [_process_result(r) for r in rs]
        return results, total


 search_controller = SearchController()
-webvtt_processor = WebVTTProcessor()
-snippet_generator = SnippetGenerator()
--- a/server/reflector/db/transcripts.py
+++ b/server/reflector/db/transcripts.py
@@ -88,8 +88,6 @@ transcripts = sqlalchemy.Table(
    sqlalchemy.Index("idx_transcript_created_at", "created_at"),
    sqlalchemy.Index("idx_transcript_user_id_recording_id", "user_id", "recording_id"),
    sqlalchemy.Index("idx_transcript_room_id", "room_id"),
-    sqlalchemy.Index("idx_transcript_source_kind", "source_kind"),
-    sqlalchemy.Index("idx_transcript_room_id_created_at", "room_id", "created_at"),
 )

 # Add PostgreSQL-specific full-text search column
@@ -101,8 +99,7 @@ if is_postgresql():
            TSVECTOR,
            sqlalchemy.Computed(
                "setweight(to_tsvector('english', coalesce(title, '')), 'A') || "
-                "setweight(to_tsvector('english', coalesce(long_summary, '')), 'B') || "
-                "setweight(to_tsvector('english', coalesce(webvtt, '')), 'C')",
+                "setweight(to_tsvector('english', coalesce(webvtt, '')), 'B')",
                persisted=True,
            ),
        )
@@ -122,15 +119,6 @@ def generate_transcript_name() -> str:
    return f"Transcript {now.strftime('%Y-%m-%d %H:%M:%S')}"


-TranscriptStatus = Literal[
-    "idle", "uploaded", "recording", "processing", "error", "ended"
-]
-
-
-class StrValue(BaseModel):
-    value: str
-
-
 class AudioWaveform(BaseModel):
    data: list[float]

@@ -194,7 +182,7 @@ class Transcript(BaseModel):
    id: str = Field(default_factory=generate_uuid4)
    user_id: str | None = None
    name: str = Field(default_factory=generate_transcript_name)
-    status: TranscriptStatus = "idle"
+    status: str = "idle"
    duration: float = 0
    created_at: datetime = Field(default_factory=lambda: datetime.now(timezone.utc))
    title: str | None = None
@@ -741,27 +729,5 @@ class TranscriptController:
        transcript.delete_participant(participant_id)
        await self.update(transcript, {"participants": transcript.participants_dump()})

-    async def set_status(
-        self, transcript_id: str, status: TranscriptStatus
-    ) -> TranscriptEvent | None:
-        """
-        Update the status of a transcript
-
-        Will add an event STATUS + update the status field of transcript
-        """
-        async with self.transaction():
-            transcript = await self.get_by_id(transcript_id)
-            if not transcript:
-                raise Exception(f"Transcript {transcript_id} not found")
-            if transcript.status == status:
-                return
-            resp = await self.append_event(
-                transcript=transcript,
-                event="STATUS",
-                data=StrValue(value=status),
-            )
-            await self.update(transcript, {"status": status})
-        return resp
-

 transcripts_controller = TranscriptController()
--- a/server/reflector/jibri_events.py
+++ b/server/reflector/jibri_events.py
@@ -1,227 +0,0 @@
-import json
-from pathlib import Path
-from typing import Any, Dict, List, Literal, Optional, Union
-
-from pydantic import BaseModel
-from typing_extensions import TypedDict
-
-
-class ParticipantInfo(BaseModel):
-    jid: str
-    nick: str
-    id: str
-    is_moderator: bool = False
-
-
-class ParticipantLeftInfo(BaseModel):
-    jid: str
-    nick: Optional[str] = None
-    duration_seconds: Optional[int] = None
-
-
-class RoomCreatedEvent(BaseModel):
-    type: Literal["room_created"]
-    timestamp: int
-    room_name: str
-    room_jid: str
-    meeting_url: str
-
-
-class RecordingStartedEvent(BaseModel):
-    type: Literal["recording_started"]
-    timestamp: int
-    room_name: str
-    session_id: str
-    jibri_jid: str
-
-
-class RecordingStoppedEvent(BaseModel):
-    type: Literal["recording_stopped"]
-    timestamp: int
-    room_name: str
-    session_id: str
-    meeting_url: str
-
-
-class ParticipantJoinedEvent(BaseModel):
-    type: Literal["participant_joined"]
-    timestamp: int
-    room_name: str
-    participant: ParticipantInfo
-
-
-class ParticipantLeftEvent(BaseModel):
-    type: Literal["participant_left"]
-    timestamp: int
-    room_name: str
-    participant: ParticipantLeftInfo
-
-
-class SpeakerActiveEvent(BaseModel):
-    type: Literal["speaker_active"]
-    timestamp: int
-    room_name: str
-    speaker_jid: str
-    speaker_nick: str
-    duration: int
-
-
-class DominantSpeakerChangedEvent(BaseModel):
-    type: Literal["dominant_speaker_changed"]
-    timestamp: int
-    room_name: str
-    previous: str
-    current: str
-
-
-JitsiEvent = Union[
-    RoomCreatedEvent,
-    RecordingStartedEvent,
-    RecordingStoppedEvent,
-    ParticipantJoinedEvent,
-    ParticipantLeftEvent,
-    SpeakerActiveEvent,
-    DominantSpeakerChangedEvent,
-]
-
-
-class RoomInfo(TypedDict):
-    name: str
-    jid: str
-    created_at: int
-    meeting_url: str
-    recording_stopped_at: Optional[int]
-
-
-class ParticipantData(TypedDict):
-    jid: str
-    nick: str
-    id: str
-    is_moderator: bool
-    joined_at: int
-    left_at: Optional[int]
-    duration: Optional[int]
-    events: List[str]
-
-
-class SpeakerStats(TypedDict):
-    total_time: int
-    nick: str
-
-
-class ParsedMetadata(TypedDict):
-    room: RoomInfo
-    participants: List[ParticipantData]
-    speaker_stats: Dict[str, SpeakerStats]
-    event_count: int
-
-
-class JitsiEventParser:
-    def parse_event(self, event_data: Dict[str, Any]) -> Optional[JitsiEvent]:
-        event_type = event_data.get("type")
-
-        try:
-            if event_type == "room_created":
-                return RoomCreatedEvent(**event_data)
-            elif event_type == "recording_started":
-                return RecordingStartedEvent(**event_data)
-            elif event_type == "recording_stopped":
-                return RecordingStoppedEvent(**event_data)
-            elif event_type == "participant_joined":
-                return ParticipantJoinedEvent(**event_data)
-            elif event_type == "participant_left":
-                return ParticipantLeftEvent(**event_data)
-            elif event_type == "speaker_active":
-                return SpeakerActiveEvent(**event_data)
-            elif event_type == "dominant_speaker_changed":
-                return DominantSpeakerChangedEvent(**event_data)
-            else:
-                return None
-        except Exception:
-            return None
-
-    def parse_events_file(self, recording_path: str) -> ParsedMetadata:
-        events_file = Path(recording_path) / "events.jsonl"
-
-        room_info: RoomInfo = {
-            "name": "",
-            "jid": "",
-            "created_at": 0,
-            "meeting_url": "",
-            "recording_stopped_at": None,
-        }
-
-        if not events_file.exists():
-            return ParsedMetadata(
-                room=room_info, participants=[], speaker_stats={}, event_count=0
-            )
-
-        events: List[JitsiEvent] = []
-        participants: Dict[str, ParticipantData] = {}
-        speaker_stats: Dict[str, SpeakerStats] = {}
-
-        with open(events_file, "r") as f:
-            for line in f:
-                if not line.strip():
-                    continue
-
-                try:
-                    event_data = json.loads(line)
-                    event = self.parse_event(event_data)
-
-                    if event is None:
-                        continue
-
-                    events.append(event)
-
-                    if isinstance(event, RoomCreatedEvent):
-                        room_info = {
-                            "name": event.room_name,
-                            "jid": event.room_jid,
-                            "created_at": event.timestamp,
-                            "meeting_url": event.meeting_url,
-                            "recording_stopped_at": None,
-                        }
-
-                    elif isinstance(event, ParticipantJoinedEvent):
-                        participants[event.participant.id] = {
-                            "jid": event.participant.jid,
-                            "nick": event.participant.nick,
-                            "id": event.participant.id,
-                            "is_moderator": event.participant.is_moderator,
-                            "joined_at": event.timestamp,
-                            "left_at": None,
-                            "duration": None,
-                            "events": ["joined"],
-                        }
-
-                    elif isinstance(event, ParticipantLeftEvent):
-                        participant_id = event.participant.jid.split("/")[0]
-                        if participant_id in participants:
-                            participants[participant_id]["left_at"] = event.timestamp
-                            participants[participant_id]["duration"] = (
-                                event.participant.duration_seconds
-                            )
-                            participants[participant_id]["events"].append("left")
-
-                    elif isinstance(event, SpeakerActiveEvent):
-                        if event.speaker_jid not in speaker_stats:
-                            speaker_stats[event.speaker_jid] = {
-                                "total_time": 0,
-                                "nick": event.speaker_nick,
-                            }
-                        speaker_stats[event.speaker_jid]["total_time"] += event.duration
-
-                    elif isinstance(event, RecordingStoppedEvent):
-                        room_info["recording_stopped_at"] = event.timestamp
-                        room_info["meeting_url"] = event.meeting_url
-
-                except (json.JSONDecodeError, Exception):
-                    continue
-
-        return ParsedMetadata(
-            room=room_info,
-            participants=list(participants.values()),
-            speaker_stats=speaker_stats,
-            event_count=len(events),
-        )
--- a/server/reflector/pipelines/main_file_pipeline.py
+++ b/server/reflector/pipelines/main_file_pipeline.py
@@ -1,439 +0,0 @@
-"""
-File-based processing pipeline
-==============================
-
-Optimized pipeline for processing complete audio/video files.
-Uses parallel processing for transcription, diarization, and waveform generation.
-"""
-
-import asyncio
-import uuid
-from pathlib import Path
-
-import av
-import structlog
-from celery import chain, shared_task
-
-from reflector.asynctask import asynctask
-from reflector.db.rooms import rooms_controller
-from reflector.db.transcripts import (
-    SourceKind,
-    Transcript,
-    TranscriptStatus,
-    transcripts_controller,
-)
-from reflector.logger import logger
-from reflector.pipelines.main_live_pipeline import (
-    PipelineMainBase,
-    broadcast_to_sockets,
-    task_cleanup_consent,
-    task_pipeline_post_to_zulip,
-)
-from reflector.processors import (
-    AudioFileWriterProcessor,
-    TranscriptFinalSummaryProcessor,
-    TranscriptFinalTitleProcessor,
-    TranscriptTopicDetectorProcessor,
-)
-from reflector.processors.audio_waveform_processor import AudioWaveformProcessor
-from reflector.processors.file_diarization import FileDiarizationInput
-from reflector.processors.file_diarization_auto import FileDiarizationAutoProcessor
-from reflector.processors.file_transcript import FileTranscriptInput
-from reflector.processors.file_transcript_auto import FileTranscriptAutoProcessor
-from reflector.processors.transcript_diarization_assembler import (
-    TranscriptDiarizationAssemblerInput,
-    TranscriptDiarizationAssemblerProcessor,
-)
-from reflector.processors.types import (
-    DiarizationSegment,
-    TitleSummary,
-)
-from reflector.processors.types import (
-    Transcript as TranscriptType,
-)
-from reflector.settings import settings
-from reflector.storage import get_transcripts_storage
-from reflector.worker.webhook import send_transcript_webhook
-
-
-class EmptyPipeline:
-    """Empty pipeline for processors that need a pipeline reference"""
-
-    def __init__(self, logger: structlog.BoundLogger):
-        self.logger = logger
-
-    def get_pref(self, k, d=None):
-        return d
-
-    async def emit(self, event):
-        pass
-
-
-class PipelineMainFile(PipelineMainBase):
-    """
-    Optimized file processing pipeline.
-    Processes complete audio/video files with parallel execution.
-    """
-
-    logger: structlog.BoundLogger = None
-    empty_pipeline = None
-
-    def __init__(self, transcript_id: str):
-        super().__init__(transcript_id=transcript_id)
-        self.logger = logger.bind(transcript_id=self.transcript_id)
-        self.empty_pipeline = EmptyPipeline(logger=self.logger)
-
-    def _handle_gather_exceptions(self, results: list, operation: str) -> None:
-        """Handle exceptions from asyncio.gather with return_exceptions=True"""
-        for i, result in enumerate(results):
-            if not isinstance(result, Exception):
-                continue
-            self.logger.error(
-                f"Error in {operation} (task {i}): {result}",
-                transcript_id=self.transcript_id,
-                exc_info=result,
-            )
-
-    @broadcast_to_sockets
-    async def set_status(self, transcript_id: str, status: TranscriptStatus):
-        async with self.lock_transaction():
-            return await transcripts_controller.set_status(transcript_id, status)
-
-    async def process(self, file_path: Path):
-        """Main entry point for file processing"""
-        self.logger.info(f"Starting file pipeline for {file_path}")
-
-        transcript = await self.get_transcript()
-
-        # Clear transcript as we're going to regenerate everything
-        async with self.transaction():
-            await transcripts_controller.update(
-                transcript,
-                {
-                    "events": [],
-                    "topics": [],
-                },
-            )
-
-        # Extract audio and write to transcript location
-        audio_path = await self.extract_and_write_audio(file_path, transcript)
-
-        # Upload for processing
-        audio_url = await self.upload_audio(audio_path, transcript)
-
-        # Run parallel processing
-        await self.run_parallel_processing(
-            audio_path,
-            audio_url,
-            transcript.source_language,
-            transcript.target_language,
-        )
-
-        self.logger.info("File pipeline complete")
-
-        await transcripts_controller.set_status(transcript.id, "ended")
-
-    async def extract_and_write_audio(
-        self, file_path: Path, transcript: Transcript
-    ) -> Path:
-        """Extract audio from video if needed and write to transcript location as MP3"""
-        self.logger.info(f"Processing audio file: {file_path}")
-
-        # Check if it's already audio-only
-        container = av.open(str(file_path))
-        has_video = len(container.streams.video) > 0
-        container.close()
-
-        # Use AudioFileWriterProcessor to write MP3 to transcript location
-        mp3_writer = AudioFileWriterProcessor(
-            path=transcript.audio_mp3_filename,
-            on_duration=self.on_duration,
-        )
-
-        # Process audio frames and write to transcript location
-        input_container = av.open(str(file_path))
-        for frame in input_container.decode(audio=0):
-            await mp3_writer.push(frame)
-
-        await mp3_writer.flush()
-        input_container.close()
-
-        if has_video:
-            self.logger.info(
-                f"Extracted audio from video and saved to {transcript.audio_mp3_filename}"
-            )
-        else:
-            self.logger.info(
-                f"Converted audio file and saved to {transcript.audio_mp3_filename}"
-            )
-
-        return transcript.audio_mp3_filename
-
-    async def upload_audio(self, audio_path: Path, transcript: Transcript) -> str:
-        """Upload audio to storage for processing"""
-        storage = get_transcripts_storage()
-
-        if not storage:
-            raise Exception(
-                "Storage backend required for file processing. Configure TRANSCRIPT_STORAGE_* settings."
-            )
-
-        self.logger.info("Uploading audio to storage")
-
-        with open(audio_path, "rb") as f:
-            audio_data = f.read()
-
-        storage_path = f"file_pipeline/{transcript.id}/audio.mp3"
-        await storage.put_file(storage_path, audio_data)
-
-        audio_url = await storage.get_file_url(storage_path)
-
-        self.logger.info(f"Audio uploaded to {audio_url}")
-        return audio_url
-
-    async def run_parallel_processing(
-        self,
-        audio_path: Path,
-        audio_url: str,
-        source_language: str,
-        target_language: str,
-    ):
-        """Coordinate parallel processing of transcription, diarization, and waveform"""
-        self.logger.info(
-            "Starting parallel processing", transcript_id=self.transcript_id
-        )
-
-        # Phase 1: Parallel processing of independent tasks
-        transcription_task = self.transcribe_file(audio_url, source_language)
-        diarization_task = self.diarize_file(audio_url)
-        waveform_task = self.generate_waveform(audio_path)
-
-        results = await asyncio.gather(
-            transcription_task, diarization_task, waveform_task, return_exceptions=True
-        )
-
-        transcript_result = results[0]
-        diarization_result = results[1]
-
-        # Handle errors - raise any exception that occurred
-        self._handle_gather_exceptions(results, "parallel processing")
-        for result in results:
-            if isinstance(result, Exception):
-                raise result
-
-        # Phase 2: Assemble transcript with diarization
-        self.logger.info(
-            "Assembling transcript with diarization", transcript_id=self.transcript_id
-        )
-        processor = TranscriptDiarizationAssemblerProcessor()
-        input_data = TranscriptDiarizationAssemblerInput(
-            transcript=transcript_result, diarization=diarization_result or []
-        )
-
-        # Store result for retrieval
-        diarized_transcript: Transcript | None = None
-
-        async def capture_result(transcript):
-            nonlocal diarized_transcript
-            diarized_transcript = transcript
-
-        processor.on(capture_result)
-        await processor.push(input_data)
-        await processor.flush()
-
-        if not diarized_transcript:
-            raise ValueError("No diarized transcript captured")
-
-        # Phase 3: Generate topics from diarized transcript
-        self.logger.info("Generating topics", transcript_id=self.transcript_id)
-        topics = await self.detect_topics(diarized_transcript, target_language)
-
-        # Phase 4: Generate title and summaries in parallel
-        self.logger.info(
-            "Generating title and summaries", transcript_id=self.transcript_id
-        )
-        results = await asyncio.gather(
-            self.generate_title(topics),
-            self.generate_summaries(topics),
-            return_exceptions=True,
-        )
-
-        self._handle_gather_exceptions(results, "title and summary generation")
-
-    async def transcribe_file(self, audio_url: str, language: str) -> TranscriptType:
-        """Transcribe complete file"""
-        processor = FileTranscriptAutoProcessor()
-        input_data = FileTranscriptInput(audio_url=audio_url, language=language)
-
-        # Store result for retrieval
-        result: TranscriptType | None = None
-
-        async def capture_result(transcript):
-            nonlocal result
-            result = transcript
-
-        processor.on(capture_result)
-        await processor.push(input_data)
-        await processor.flush()
-
-        if not result:
-            raise ValueError("No transcript captured")
-
-        return result
-
-    async def diarize_file(self, audio_url: str) -> list[DiarizationSegment] | None:
-        """Get diarization for file"""
-        if not settings.DIARIZATION_BACKEND:
-            self.logger.info("Diarization disabled")
-            return None
-
-        processor = FileDiarizationAutoProcessor()
-        input_data = FileDiarizationInput(audio_url=audio_url)
-
-        # Store result for retrieval
-        result = None
-
-        async def capture_result(diarization_output):
-            nonlocal result
-            result = diarization_output.diarization
-
-        try:
-            processor.on(capture_result)
-            await processor.push(input_data)
-            await processor.flush()
-            return result
-        except Exception as e:
-            self.logger.error(f"Diarization failed: {e}")
-            return None
-
-    async def generate_waveform(self, audio_path: Path):
-        """Generate and save waveform"""
-        transcript = await self.get_transcript()
-
-        processor = AudioWaveformProcessor(
-            audio_path=audio_path,
-            waveform_path=transcript.audio_waveform_filename,
-            on_waveform=self.on_waveform,
-        )
-        processor.set_pipeline(self.empty_pipeline)
-
-        await processor.flush()
-
-    async def detect_topics(
-        self, transcript: TranscriptType, target_language: str
-    ) -> list[TitleSummary]:
-        """Detect topics from complete transcript"""
-        chunk_size = 300
-        topics: list[TitleSummary] = []
-
-        async def on_topic(topic: TitleSummary):
-            topics.append(topic)
-            return await self.on_topic(topic)
-
-        topic_detector = TranscriptTopicDetectorProcessor(callback=on_topic)
-        topic_detector.set_pipeline(self.empty_pipeline)
-
-        for i in range(0, len(transcript.words), chunk_size):
-            chunk_words = transcript.words[i : i + chunk_size]
-            if not chunk_words:
-                continue
-
-            chunk_transcript = TranscriptType(
-                words=chunk_words, translation=transcript.translation
-            )
-
-            await topic_detector.push(chunk_transcript)
-
-        await topic_detector.flush()
-        return topics
-
-    async def generate_title(self, topics: list[TitleSummary]):
-        """Generate title from topics"""
-        if not topics:
-            self.logger.warning("No topics for title generation")
-            return
-
-        processor = TranscriptFinalTitleProcessor(callback=self.on_title)
-        processor.set_pipeline(self.empty_pipeline)
-
-        for topic in topics:
-            await processor.push(topic)
-
-        await processor.flush()
-
-    async def generate_summaries(self, topics: list[TitleSummary]):
-        """Generate long and short summaries from topics"""
-        if not topics:
-            self.logger.warning("No topics for summary generation")
-            return
-
-        transcript = await self.get_transcript()
-        processor = TranscriptFinalSummaryProcessor(
-            transcript=transcript,
-            callback=self.on_long_summary,
-            on_short_summary=self.on_short_summary,
-        )
-        processor.set_pipeline(self.empty_pipeline)
-
-        for topic in topics:
-            await processor.push(topic)
-
-        await processor.flush()
-
-
-@shared_task
-@asynctask
-async def task_send_webhook_if_needed(*, transcript_id: str):
-    """Send webhook if this is a room recording with webhook configured"""
-    transcript = await transcripts_controller.get_by_id(transcript_id)
-    if not transcript:
-        return
-
-    if transcript.source_kind == SourceKind.ROOM and transcript.room_id:
-        room = await rooms_controller.get_by_id(transcript.room_id)
-        if room and room.webhook_url:
-            logger.info(
-                "Dispatching webhook",
-                transcript_id=transcript_id,
-                room_id=room.id,
-                webhook_url=room.webhook_url,
-            )
-            send_transcript_webhook.delay(
-                transcript_id, room.id, event_id=uuid.uuid4().hex
-            )
-
-
-@shared_task
-@asynctask
-async def task_pipeline_file_process(*, transcript_id: str):
-    """Celery task for file pipeline processing"""
-
-    transcript = await transcripts_controller.get_by_id(transcript_id)
-    if not transcript:
-        raise Exception(f"Transcript {transcript_id} not found")
-
-    pipeline = PipelineMainFile(transcript_id=transcript_id)
-    try:
-        await pipeline.set_status(transcript_id, "processing")
-
-        # Find the file to process
-        audio_file = next(transcript.data_path.glob("upload.*"), None)
-        if not audio_file:
-            audio_file = next(transcript.data_path.glob("audio.*"), None)
-
-        if not audio_file:
-            raise Exception("No audio file found to process")
-
-        await pipeline.process(audio_file)
-
-    except Exception:
-        await pipeline.set_status(transcript_id, "error")
-        raise
-
-    # Run post-processing chain: consent cleanup -> zulip -> webhook
-    post_chain = chain(
-        task_cleanup_consent.si(transcript_id=transcript_id),
-        task_pipeline_post_to_zulip.si(transcript_id=transcript_id),
-        task_send_webhook_if_needed.si(transcript_id=transcript_id),
-    )
-    post_chain.delay()
--- a/server/reflector/pipelines/main_live_pipeline.py
+++ b/server/reflector/pipelines/main_live_pipeline.py
@@ -22,7 +22,7 @@ from celery import chord, current_task, group, shared_task
 from pydantic import BaseModel
 from structlog import BoundLogger as Logger

-from reflector.asynctask import asynctask
+from reflector.db import get_database
 from reflector.db.meetings import meeting_consent_controller, meetings_controller
 from reflector.db.recordings import recordings_controller
 from reflector.db.rooms import rooms_controller
@@ -32,7 +32,6 @@ from reflector.db.transcripts import (
    TranscriptFinalLongSummary,
    TranscriptFinalShortSummary,
    TranscriptFinalTitle,
-    TranscriptStatus,
    TranscriptText,
    TranscriptTopic,
    TranscriptWaveform,
@@ -41,9 +40,8 @@ from reflector.db.transcripts import (
 from reflector.logger import logger
 from reflector.pipelines.runner import PipelineMessage, PipelineRunner
 from reflector.processors import (
-    AudioChunkerAutoProcessor,
+    AudioChunkerProcessor,
    AudioDiarizationAutoProcessor,
-    AudioDownscaleProcessor,
    AudioFileWriterProcessor,
    AudioMergeProcessor,
    AudioTranscriptAutoProcessor,
@@ -70,6 +68,29 @@ from reflector.zulip import (
 )


+def asynctask(f):
+    @functools.wraps(f)
+    def wrapper(*args, **kwargs):
+        async def run_with_db():
+            database = get_database()
+            await database.connect()
+            try:
+                return await f(*args, **kwargs)
+            finally:
+                await database.disconnect()
+
+        coro = run_with_db()
+        try:
+            loop = asyncio.get_running_loop()
+        except RuntimeError:
+            loop = None
+        if loop and loop.is_running():
+            return loop.run_until_complete(coro)
+        return asyncio.run(coro)
+
+    return wrapper
+
+
 def broadcast_to_sockets(func):
    """
    Decorator to broadcast transcript event to websockets
@@ -126,18 +147,15 @@ class StrValue(BaseModel):


 class PipelineMainBase(PipelineRunner[PipelineMessage], Generic[PipelineMessage]):
-    def __init__(self, transcript_id: str):
-        super().__init__()
-        self._lock = asyncio.Lock()
-        self.transcript_id = transcript_id
-        self.ws_room_id = f"ts:{self.transcript_id}"
-        self._ws_manager = None
+    transcript_id: str
+    ws_room_id: str | None = None
+    ws_manager: WebsocketManager | None = None

-    @property
-    def ws_manager(self) -> WebsocketManager:
-        if self._ws_manager is None:
-            self._ws_manager = get_ws_manager()
-        return self._ws_manager
+    def prepare(self):
+        # prepare websocket
+        self._lock = asyncio.Lock()
+        self.ws_room_id = f"ts:{self.transcript_id}"
+        self.ws_manager = get_ws_manager()

    async def get_transcript(self) -> Transcript:
        # fetch the transcript
@@ -165,16 +183,9 @@ class PipelineMainBase(PipelineRunner[PipelineMessage], Generic[PipelineMessage]
            for topic in topics
        ]

-    @asynccontextmanager
-    async def lock_transaction(self):
-        # This lock is to prevent multiple processor starting adding
-        # into event array at the same time
-        async with self._lock:
-            yield
-
    @asynccontextmanager
    async def transaction(self):
-        async with self.lock_transaction():
+        async with self._lock:
            async with transcripts_controller.transaction():
                yield

@@ -183,14 +194,14 @@ class PipelineMainBase(PipelineRunner[PipelineMessage], Generic[PipelineMessage]
        # if it's the first part, update the status of the transcript
        # but do not set the ended status yet.
        if isinstance(self, PipelineMainLive):
-            status_mapping: dict[str, TranscriptStatus] = {
+            status_mapping = {
                "started": "recording",
                "push": "recording",
                "flush": "processing",
                "error": "error",
            }
        elif isinstance(self, PipelineMainFinalSummaries):
-            status_mapping: dict[str, TranscriptStatus] = {
+            status_mapping = {
                "push": "processing",
                "flush": "processing",
                "error": "error",
@@ -206,8 +217,22 @@ class PipelineMainBase(PipelineRunner[PipelineMessage], Generic[PipelineMessage]
            return

        # when the status of the pipeline changes, update the transcript
-        async with self._lock:
-            return await transcripts_controller.set_status(self.transcript_id, status)
+        async with self.transaction():
+            transcript = await self.get_transcript()
+            if status == transcript.status:
+                return
+            resp = await transcripts_controller.append_event(
+                transcript=transcript,
+                event="STATUS",
+                data=StrValue(value=status),
+            )
+            await transcripts_controller.update(
+                transcript,
+                {
+                    "status": status,
+                },
+            )
+            return resp

    @broadcast_to_sockets
    async def on_transcript(self, data):
@@ -330,6 +355,7 @@ class PipelineMainLive(PipelineMainBase):
    async def create(self) -> Pipeline:
        # create a context for the whole rtc transaction
        # add a customised logger to the context
+        self.prepare()
        transcript = await self.get_transcript()

        processors = [
@@ -337,8 +363,7 @@ class PipelineMainLive(PipelineMainBase):
                path=transcript.audio_wav_filename,
                on_duration=self.on_duration,
            ),
-            AudioDownscaleProcessor(),
-            AudioChunkerAutoProcessor(),
+            AudioChunkerProcessor(),
            AudioMergeProcessor(),
            AudioTranscriptAutoProcessor.as_threaded(),
            TranscriptLinerProcessor(),
@@ -351,7 +376,6 @@ class PipelineMainLive(PipelineMainBase):
        pipeline.set_pref("audio:target_language", transcript.target_language)
        pipeline.logger.bind(transcript_id=transcript.id)
        pipeline.logger.info("Pipeline main live created")
-        pipeline.describe()

        return pipeline

@@ -370,6 +394,7 @@ class PipelineMainDiarization(PipelineMainBase[AudioDiarizationInput]):
    async def create(self) -> Pipeline:
        # create a context for the whole rtc transaction
        # add a customised logger to the context
+        self.prepare()
        pipeline = Pipeline(
            AudioDiarizationAutoProcessor(callback=self.on_topic),
        )
@@ -410,6 +435,8 @@ class PipelineMainFromTopics(PipelineMainBase[TitleSummaryWithIdProcessorType]):
        raise NotImplementedError

    async def create(self) -> Pipeline:
+        self.prepare()
+
        # get transcript
        self._transcript = transcript = await self.get_transcript()

@@ -765,7 +792,7 @@ def pipeline_post(*, transcript_id: str):
        chain_final_summaries,
    ) | task_pipeline_post_to_zulip.si(transcript_id=transcript_id)

-    return chain.delay()
+    chain.delay()


@get_transcript
--- a/server/reflector/pipelines/runner.py
+++ b/server/reflector/pipelines/runner.py
@@ -18,14 +18,22 @@ During its lifecycle, it will emit the following status:
 import asyncio
 from typing import Generic, TypeVar

+from pydantic import BaseModel, ConfigDict
+
 from reflector.logger import logger
 from reflector.processors import Pipeline

 PipelineMessage = TypeVar("PipelineMessage")


-class PipelineRunner(Generic[PipelineMessage]):
-    def __init__(self):
+class PipelineRunner(BaseModel, Generic[PipelineMessage]):
+    model_config = ConfigDict(arbitrary_types_allowed=True)
+
+    status: str = "idle"
+    pipeline: Pipeline | None = None
+
+    def __init__(self, **kwargs):
+        super().__init__(**kwargs)
        self._task = None
        self._q_cmd = asyncio.Queue(maxsize=4096)
        self._ev_done = asyncio.Event()
@@ -34,8 +42,6 @@ class PipelineRunner(Generic[PipelineMessage]):
            runner=id(self),
            runner_cls=self.__class__.__name__,
        )
-        self.status = "idle"
-        self.pipeline: Pipeline | None = None

    async def create(self) -> Pipeline:
        """
--- a/server/reflector/processors/init.py
+++ b/server/reflector/processors/init.py
@@ -1,7 +1,5 @@
 from .audio_chunker import AudioChunkerProcessor  # noqa: F401
-from .audio_chunker_auto import AudioChunkerAutoProcessor  # noqa: F401
 from .audio_diarization_auto import AudioDiarizationAutoProcessor  # noqa: F401
-from .audio_downscale import AudioDownscaleProcessor  # noqa: F401
 from .audio_file_writer import AudioFileWriterProcessor  # noqa: F401
 from .audio_merge import AudioMergeProcessor  # noqa: F401
 from .audio_transcript import AudioTranscriptProcessor  # noqa: F401
@@ -13,13 +11,6 @@ from .base import (  # noqa: F401
    Processor,
    ThreadedProcessor,
 )
-from .file_diarization import FileDiarizationProcessor  # noqa: F401
-from .file_diarization_auto import FileDiarizationAutoProcessor  # noqa: F401
-from .file_transcript import FileTranscriptProcessor  # noqa: F401
-from .file_transcript_auto import FileTranscriptAutoProcessor  # noqa: F401
-from .transcript_diarization_assembler import (
-    TranscriptDiarizationAssemblerProcessor,  # noqa: F401
-)
 from .transcript_final_summary import TranscriptFinalSummaryProcessor  # noqa: F401
 from .transcript_final_title import TranscriptFinalTitleProcessor  # noqa: F401
 from .transcript_liner import TranscriptLinerProcessor  # noqa: F401
--- a/server/reflector/processors/audio_chunker.py
+++ b/server/reflector/processors/audio_chunker.py
@@ -1,78 +1,28 @@
-from typing import Optional
-
 import av
-from prometheus_client import Counter, Histogram

 from reflector.processors.base import Processor


 class AudioChunkerProcessor(Processor):
    """
-    Base class for assembling audio frames into chunks
+    Assemble audio frames into chunks
    """

    INPUT_TYPE = av.AudioFrame
    OUTPUT_TYPE = list[av.AudioFrame]

-    m_chunk = Histogram(
-        "audio_chunker",
-        "Time spent in AudioChunker.chunk",
-        ["backend"],
-    )
-    m_chunk_call = Counter(
-        "audio_chunker_call",
-        "Number of calls to AudioChunker.chunk",
-        ["backend"],
-    )
-    m_chunk_success = Counter(
-        "audio_chunker_success",
-        "Number of successful calls to AudioChunker.chunk",
-        ["backend"],
-    )
-    m_chunk_failure = Counter(
-        "audio_chunker_failure",
-        "Number of failed calls to AudioChunker.chunk",
-        ["backend"],
-    )
-
-    def __init__(self, *args, **kwargs):
-        name = self.__class__.__name__
-        self.m_chunk = self.m_chunk.labels(name)
-        self.m_chunk_call = self.m_chunk_call.labels(name)
-        self.m_chunk_success = self.m_chunk_success.labels(name)
-        self.m_chunk_failure = self.m_chunk_failure.labels(name)
-        super().__init__(*args, **kwargs)
+    def __init__(self, max_frames=256):
+        super().__init__()
        self.frames: list[av.AudioFrame] = []
+        self.max_frames = max_frames

    async def _push(self, data: av.AudioFrame):
-        """Process incoming audio frame"""
-        # Validate audio format on first frame
-        if len(self.frames) == 0:
-            if data.sample_rate != 16000 or len(data.layout.channels) != 1:
-                raise ValueError(
-                    f"AudioChunkerProcessor expects 16kHz mono audio, got {data.sample_rate}Hz "
-                    f"with {len(data.layout.channels)} channel(s). "
-                    f"Use AudioDownscaleProcessor before this processor."
-                )
-
-        try:
-            self.m_chunk_call.inc()
-            with self.m_chunk.time():
-                result = await self._chunk(data)
-            self.m_chunk_success.inc()
-            if result:
-                await self.emit(result)
-        except Exception:
-            self.m_chunk_failure.inc()
-            raise
-
-    async def _chunk(self, data: av.AudioFrame) -> Optional[list[av.AudioFrame]]:
-        """
-        Process audio frame and return chunk when ready.
-        Subclasses should implement their chunking logic here.
-        """
-        raise NotImplementedError
+        self.frames.append(data)
+        if len(self.frames) >= self.max_frames:
+            await self.flush()

    async def _flush(self):
-        """Flush any remaining frames when processing ends"""
-        raise NotImplementedError
+        frames = self.frames[:]
+        self.frames = []
+        if frames:
+            await self.emit(frames)
--- a/server/reflector/processors/audio_chunker_auto.py
+++ b/server/reflector/processors/audio_chunker_auto.py
@@ -1,32 +0,0 @@
-import importlib
-
-from reflector.processors.audio_chunker import AudioChunkerProcessor
-from reflector.settings import settings
-
-
-class AudioChunkerAutoProcessor(AudioChunkerProcessor):
-    _registry = {}
-
-    @classmethod
-    def register(cls, name, kclass):
-        cls._registry[name] = kclass
-
-    def __new__(cls, name: str | None = None, **kwargs):
-        if name is None:
-            name = settings.AUDIO_CHUNKER_BACKEND
-        if name not in cls._registry:
-            module_name = f"reflector.processors.audio_chunker_{name}"
-            importlib.import_module(module_name)
-
-        # gather specific configuration for the processor
-        # search `AUDIO_CHUNKER_BACKEND_XXX_YYY`, push to constructor as `backend_xxx_yyy`
-        config = {}
-        name_upper = name.upper()
-        settings_prefix = "AUDIO_CHUNKER_"
-        config_prefix = f"{settings_prefix}{name_upper}_"
-        for key, value in settings:
-            if key.startswith(config_prefix):
-                config_name = key[len(settings_prefix) :].lower()
-                config[config_name] = value
-
-        return cls._registry[name](**config | kwargs)
--- a/server/reflector/processors/audio_chunker_frames.py
+++ b/server/reflector/processors/audio_chunker_frames.py
@@ -1,34 +0,0 @@
-from typing import Optional
-
-import av
-
-from reflector.processors.audio_chunker import AudioChunkerProcessor
-from reflector.processors.audio_chunker_auto import AudioChunkerAutoProcessor
-
-
-class AudioChunkerFramesProcessor(AudioChunkerProcessor):
-    """
-    Simple frame-based audio chunker that emits chunks after a fixed number of frames
-    """
-
-    def __init__(self, max_frames=256, **kwargs):
-        super().__init__(**kwargs)
-        self.max_frames = max_frames
-
-    async def _chunk(self, data: av.AudioFrame) -> Optional[list[av.AudioFrame]]:
-        self.frames.append(data)
-        if len(self.frames) >= self.max_frames:
-            frames_to_emit = self.frames[:]
-            self.frames = []
-            return frames_to_emit
-
-        return None
-
-    async def _flush(self):
-        frames = self.frames[:]
-        self.frames = []
-        if frames:
-            await self.emit(frames)
-
-
-AudioChunkerAutoProcessor.register("frames", AudioChunkerFramesProcessor)
--- a/server/reflector/processors/audio_chunker_silero.py
+++ b/server/reflector/processors/audio_chunker_silero.py
@@ -1,298 +0,0 @@
-from typing import Optional
-
-import av
-import numpy as np
-import torch
-from silero_vad import VADIterator, load_silero_vad
-
-from reflector.processors.audio_chunker import AudioChunkerProcessor
-from reflector.processors.audio_chunker_auto import AudioChunkerAutoProcessor
-
-
-class AudioChunkerSileroProcessor(AudioChunkerProcessor):
-    """
-    Assemble audio frames into chunks with VAD-based speech detection using Silero VAD
-    """
-
-    def __init__(
-        self,
-        block_frames=256,
-        max_frames=1024,
-        use_onnx=True,
-        min_frames=2,
-        **kwargs,
-    ):
-        super().__init__(**kwargs)
-        self.block_frames = block_frames
-        self.max_frames = max_frames
-        self.min_frames = min_frames
-
-        # Initialize Silero VAD
-        self._init_vad(use_onnx)
-
-    def _init_vad(self, use_onnx=False):
-        """Initialize Silero VAD model"""
-        try:
-            torch.set_num_threads(1)
-            self.vad_model = load_silero_vad(onnx=use_onnx)
-            self.vad_iterator = VADIterator(self.vad_model, sampling_rate=16000)
-            self.logger.info("Silero VAD initialized successfully")
-
-        except Exception as e:
-            self.logger.error(f"Failed to initialize Silero VAD: {e}")
-            self.vad_model = None
-            self.vad_iterator = None
-
-    async def _chunk(self, data: av.AudioFrame) -> Optional[list[av.AudioFrame]]:
-        """Process audio frame and return chunk when ready"""
-        self.frames.append(data)
-
-        # Check for speech segments every 32 frames (~1 second)
-        if len(self.frames) >= 32 and len(self.frames) % 32 == 0:
-            return await self._process_block()
-
-        # Safety fallback - emit if we hit max frames
-        elif len(self.frames) >= self.max_frames:
-            self.logger.warning(
-                f"AudioChunkerSileroProcessor: Reached max frames ({self.max_frames}), "
-                f"emitting first {self.max_frames // 2} frames"
-            )
-            frames_to_emit = self.frames[: self.max_frames // 2]
-            self.frames = self.frames[self.max_frames // 2 :]
-            if len(frames_to_emit) >= self.min_frames:
-                return frames_to_emit
-            else:
-                self.logger.debug(
-                    f"Ignoring fallback segment with {len(frames_to_emit)} frames "
-                    f"(< {self.min_frames} minimum)"
-                )
-
-        return None
-
-    async def _process_block(self) -> Optional[list[av.AudioFrame]]:
-        # Need at least 32 frames for VAD detection (~1 second)
-        if len(self.frames) < 32 or self.vad_iterator is None:
-            return None
-
-        # Processing block with current buffer size
-        print(f"Processing block: {len(self.frames)} frames in buffer")
-
-        try:
-            # Convert frames to numpy array for VAD
-            audio_array = self._frames_to_numpy(self.frames)
-
-            if audio_array is None:
-                # Fallback: emit all frames if conversion failed
-                frames_to_emit = self.frames[:]
-                self.frames = []
-                if len(frames_to_emit) >= self.min_frames:
-                    return frames_to_emit
-                else:
-                    self.logger.debug(
-                        f"Ignoring conversion-failed segment with {len(frames_to_emit)} frames "
-                        f"(< {self.min_frames} minimum)"
-                    )
-                return None
-
-            # Find complete speech segments in the buffer
-            speech_end_frame = self._find_speech_segment_end(audio_array)
-
-            if speech_end_frame is None or speech_end_frame <= 0:
-                # No speech found but buffer is getting large
-                if len(self.frames) > 512:
-                    # Check if it's all silence and can be discarded
-                    # No speech segment found, buffer at {len(self.frames)} frames
-
-                    # Could emit silence or discard old frames here
-                    # For now, keep first 256 frames and discard older silence
-                    if len(self.frames) > 768:
-                        self.logger.debug(
-                            f"Discarding {len(self.frames) - 256} old frames (likely silence)"
-                        )
-                        self.frames = self.frames[-256:]
-                return None
-
-            # Calculate segment timing information
-            frames_to_emit = self.frames[:speech_end_frame]
-
-            # Get timing from av.AudioFrame
-            if frames_to_emit:
-                first_frame = frames_to_emit[0]
-                last_frame = frames_to_emit[-1]
-                sample_rate = first_frame.sample_rate
-
-                # Calculate duration
-                total_samples = sum(f.samples for f in frames_to_emit)
-                duration_seconds = total_samples / sample_rate if sample_rate > 0 else 0
-
-                # Get timestamps if available
-                start_time = (
-                    first_frame.pts * first_frame.time_base if first_frame.pts else 0
-                )
-                end_time = (
-                    last_frame.pts * last_frame.time_base if last_frame.pts else 0
-                )
-
-                # Convert to HH:MM:SS format for logging
-                def format_time(seconds):
-                    if not seconds:
-                        return "00:00:00"
-                    total_seconds = int(float(seconds))
-                    hours = total_seconds // 3600
-                    minutes = (total_seconds % 3600) // 60
-                    secs = total_seconds % 60
-                    return f"{hours:02d}:{minutes:02d}:{secs:02d}"
-
-                start_formatted = format_time(start_time)
-                end_formatted = format_time(end_time)
-
-                # Keep remaining frames for next processing
-                remaining_after = len(self.frames) - speech_end_frame
-
-                # Single structured log line
-                self.logger.info(
-                    "Speech segment found",
-                    start=start_formatted,
-                    end=end_formatted,
-                    frames=speech_end_frame,
-                    duration=round(duration_seconds, 2),
-                    buffer_before=len(self.frames),
-                    remaining=remaining_after,
-                )
-
-            # Keep remaining frames for next processing
-            self.frames = self.frames[speech_end_frame:]
-
-            # Filter out segments with too few frames
-            if len(frames_to_emit) >= self.min_frames:
-                return frames_to_emit
-            else:
-                self.logger.debug(
-                    f"Ignoring segment with {len(frames_to_emit)} frames "
-                    f"(< {self.min_frames} minimum)"
-                )
-
-        except Exception as e:
-            self.logger.error(f"Error in VAD processing: {e}")
-            # Fallback to simple chunking
-            if len(self.frames) >= self.block_frames:
-                frames_to_emit = self.frames[: self.block_frames]
-                self.frames = self.frames[self.block_frames :]
-                if len(frames_to_emit) >= self.min_frames:
-                    return frames_to_emit
-                else:
-                    self.logger.debug(
-                        f"Ignoring exception-fallback segment with {len(frames_to_emit)} frames "
-                        f"(< {self.min_frames} minimum)"
-                    )
-
-        return None
-
-    def _frames_to_numpy(self, frames: list[av.AudioFrame]) -> Optional[np.ndarray]:
-        """Convert av.AudioFrame list to numpy array for VAD processing"""
-        if not frames:
-            return None
-
-        try:
-            audio_data = []
-            for frame in frames:
-                frame_array = frame.to_ndarray()
-
-                if len(frame_array.shape) == 2:
-                    frame_array = frame_array.flatten()
-
-                audio_data.append(frame_array)
-
-            if not audio_data:
-                return None
-
-            combined_audio = np.concatenate(audio_data)
-
-            # Ensure float32 format
-            if combined_audio.dtype == np.int16:
-                # Normalize int16 audio to float32 in range [-1.0, 1.0]
-                combined_audio = combined_audio.astype(np.float32) / 32768.0
-            elif combined_audio.dtype != np.float32:
-                combined_audio = combined_audio.astype(np.float32)
-
-            return combined_audio
-
-        except Exception as e:
-            self.logger.error(f"Error converting frames to numpy: {e}")
-
-        return None
-
-    def _find_speech_segment_end(self, audio_array: np.ndarray) -> Optional[int]:
-        """Find complete speech segments and return frame index at segment end"""
-        if self.vad_iterator is None or len(audio_array) == 0:
-            return None
-
-        try:
-            # Process audio in 512-sample windows for VAD
-            window_size = 512
-            min_silence_windows = 3  # Require 3 windows of silence after speech
-
-            # Track speech state
-            in_speech = False
-            speech_start = None
-            speech_end = None
-            silence_count = 0
-
-            for i in range(0, len(audio_array), window_size):
-                chunk = audio_array[i : i + window_size]
-                if len(chunk) < window_size:
-                    chunk = np.pad(chunk, (0, window_size - len(chunk)))
-
-                # Detect if this window has speech
-                speech_dict = self.vad_iterator(chunk, return_seconds=True)
-
-                # VADIterator returns dict with 'start' and 'end' when speech segments are detected
-                if speech_dict:
-                    if not in_speech:
-                        # Speech started
-                        speech_start = i
-                        in_speech = True
-                        # Debug: print(f"Speech START at sample {i}, VAD: {speech_dict}")
-                    silence_count = 0  # Reset silence counter
-                    continue
-
-                if not in_speech:
-                    continue
-
-                # We're in speech but found silence
-                silence_count += 1
-                if silence_count < min_silence_windows:
-                    continue
-
-                # Found end of speech segment
-                speech_end = i - (min_silence_windows - 1) * window_size
-                # Debug: print(f"Speech END at sample {speech_end}")
-
-                # Convert sample position to frame index
-                samples_per_frame = self.frames[0].samples if self.frames else 1024
-                frame_index = speech_end // samples_per_frame
-
-                # Ensure we don't exceed buffer
-                frame_index = min(frame_index, len(self.frames))
-                return frame_index
-
-            return None
-
-        except Exception as e:
-            self.logger.error(f"Error finding speech segment: {e}")
-            return None
-
-    async def _flush(self):
-        frames = self.frames[:]
-        self.frames = []
-        if frames:
-            if len(frames) >= self.min_frames:
-                await self.emit(frames)
-            else:
-                self.logger.debug(
-                    f"Ignoring flush segment with {len(frames)} frames "
-                    f"(< {self.min_frames} minimum)"
-                )
-
-
-AudioChunkerAutoProcessor.register("silero", AudioChunkerSileroProcessor)
--- a/server/reflector/processors/audio_diarization.py
+++ b/server/reflector/processors/audio_diarization.py
@@ -1,7 +1,6 @@
 from reflector.processors.base import Processor
 from reflector.processors.types import (
    AudioDiarizationInput,
-    DiarizationSegment,
    TitleSummary,
    Word,
 )
@@ -38,21 +37,18 @@ class AudioDiarizationProcessor(Processor):
    async def _diarize(self, data: AudioDiarizationInput):
        raise NotImplementedError

-    @classmethod
-    def assign_speaker(cls, words: list[Word], diarization: list[DiarizationSegment]):
-        cls._diarization_remove_overlap(diarization)
-        cls._diarization_remove_segment_without_words(words, diarization)
-        cls._diarization_merge_same_speaker(diarization)
-        cls._diarization_assign_speaker(words, diarization)
+    def assign_speaker(self, words: list[Word], diarization: list[dict]):
+        self._diarization_remove_overlap(diarization)
+        self._diarization_remove_segment_without_words(words, diarization)
+        self._diarization_merge_same_speaker(words, diarization)
+        self._diarization_assign_speaker(words, diarization)

-    @staticmethod
-    def iter_words_from_topics(topics: list[TitleSummary]):
+    def iter_words_from_topics(self, topics: TitleSummary):
        for topic in topics:
            for word in topic.transcript.words:
                yield word

-    @staticmethod
-    def is_word_continuation(word_prev, word):
+    def is_word_continuation(self, word_prev, word):
        """
        Return True if the word is a continuation of the previous word
        by checking if the previous word is ending with a punctuation
@@ -65,8 +61,7 @@ class AudioDiarizationProcessor(Processor):
            return False
        return True

-    @staticmethod
-    def _diarization_remove_overlap(diarization: list[DiarizationSegment]):
+    def _diarization_remove_overlap(self, diarization: list[dict]):
        """
        Remove overlap in diarization results

@@ -91,9 +86,8 @@ class AudioDiarizationProcessor(Processor):
            else:
                diarization_idx += 1

-    @staticmethod
    def _diarization_remove_segment_without_words(
-        words: list[Word], diarization: list[DiarizationSegment]
+        self, words: list[Word], diarization: list[dict]
    ):
        """
        Remove diarization segments without words
@@ -122,8 +116,9 @@ class AudioDiarizationProcessor(Processor):
            else:
                diarization_idx += 1

-    @staticmethod
-    def _diarization_merge_same_speaker(diarization: list[DiarizationSegment]):
+    def _diarization_merge_same_speaker(
+        self, words: list[Word], diarization: list[dict]
+    ):
        """
        Merge diarization contigous segments with the same speaker

@@ -140,10 +135,7 @@ class AudioDiarizationProcessor(Processor):
            else:
                diarization_idx += 1

-    @classmethod
-    def _diarization_assign_speaker(
-        cls, words: list[Word], diarization: list[DiarizationSegment]
-    ):
+    def _diarization_assign_speaker(self, words: list[Word], diarization: list[dict]):
        """
        Assign speaker to words based on diarization

@@ -151,7 +143,7 @@ class AudioDiarizationProcessor(Processor):
        """

        word_idx = 0
-        last_speaker = 0
+        last_speaker = None
        for d in diarization:
            start = d["start"]
            end = d["end"]
@@ -166,7 +158,7 @@ class AudioDiarizationProcessor(Processor):
                    # If it's a continuation, assign with the last speaker
                    is_continuation = False
                    if word_idx > 0 and word_idx < len(words) - 1:
-                        is_continuation = cls.is_word_continuation(
+                        is_continuation = self.is_word_continuation(
                            *words[word_idx - 1 : word_idx + 1]
                        )
                    if is_continuation:
--- a/server/reflector/processors/audio_diarization_pyannote.py
+++ b/server/reflector/processors/audio_diarization_pyannote.py
@@ -1,74 +0,0 @@
-import os
-
-import torch
-import torchaudio
-from pyannote.audio import Pipeline
-
-from reflector.processors.audio_diarization import AudioDiarizationProcessor
-from reflector.processors.audio_diarization_auto import AudioDiarizationAutoProcessor
-from reflector.processors.types import AudioDiarizationInput, DiarizationSegment
-
-
-class AudioDiarizationPyannoteProcessor(AudioDiarizationProcessor):
-    """Local diarization processor using pyannote.audio library"""
-
-    def __init__(
-        self,
-        model_name: str = "pyannote/speaker-diarization-3.1",
-        pyannote_auth_token: str | None = None,
-        device: str | None = None,
-        **kwargs,
-    ):
-        super().__init__(**kwargs)
-        self.model_name = model_name
-        self.auth_token = pyannote_auth_token or os.environ.get("HF_TOKEN")
-        self.device = device
-
-        if device is None:
-            self.device = "cuda" if torch.cuda.is_available() else "cpu"
-
-        self.logger.info(f"Loading pyannote diarization model: {self.model_name}")
-        self.diarization_pipeline = Pipeline.from_pretrained(
-            self.model_name, use_auth_token=self.auth_token
-        )
-        self.diarization_pipeline.to(torch.device(self.device))
-        self.logger.info(f"Diarization model loaded on device: {self.device}")
-
-    async def _diarize(self, data: AudioDiarizationInput) -> list[DiarizationSegment]:
-        try:
-            # Load audio file (audio_url is assumed to be a local file path)
-            self.logger.info(f"Loading local audio file: {data.audio_url}")
-            waveform, sample_rate = torchaudio.load(data.audio_url)
-            audio_input = {"waveform": waveform, "sample_rate": sample_rate}
-            self.logger.info("Running speaker diarization")
-            diarization = self.diarization_pipeline(audio_input)
-
-            # Convert pyannote diarization output to our format
-            segments = []
-            for segment, _, speaker in diarization.itertracks(yield_label=True):
-                # Extract speaker number from label (e.g., "SPEAKER_00" -> 0)
-                speaker_id = 0
-                if speaker.startswith("SPEAKER_"):
-                    try:
-                        speaker_id = int(speaker.split("_")[-1])
-                    except (ValueError, IndexError):
-                        # Fallback to hash-based ID if parsing fails
-                        speaker_id = hash(speaker) % 1000
-
-                segments.append(
-                    {
-                        "start": round(segment.start, 3),
-                        "end": round(segment.end, 3),
-                        "speaker": speaker_id,
-                    }
-                )
-
-            self.logger.info(f"Diarization completed with {len(segments)} segments")
-            return segments
-
-        except Exception as e:
-            self.logger.exception(f"Diarization failed: {e}")
-            raise
-
-
-AudioDiarizationAutoProcessor.register("pyannote", AudioDiarizationPyannoteProcessor)
--- a/server/reflector/processors/audio_downscale.py
+++ b/server/reflector/processors/audio_downscale.py
@@ -1,60 +0,0 @@
-from typing import Optional
-
-import av
-from av.audio.resampler import AudioResampler
-
-from reflector.processors.base import Processor
-
-
-def copy_frame(frame: av.AudioFrame) -> av.AudioFrame:
-    frame_copy = frame.from_ndarray(
-        frame.to_ndarray(),
-        format=frame.format.name,
-        layout=frame.layout.name,
-    )
-    frame_copy.sample_rate = frame.sample_rate
-    frame_copy.pts = frame.pts
-    frame_copy.time_base = frame.time_base
-    return frame_copy
-
-
-class AudioDownscaleProcessor(Processor):
-    """
-    Downscale audio frames to 16kHz mono format
-    """
-
-    INPUT_TYPE = av.AudioFrame
-    OUTPUT_TYPE = av.AudioFrame
-
-    def __init__(self, target_rate: int = 16000, target_layout: str = "mono", **kwargs):
-        super().__init__(**kwargs)
-        self.target_rate = target_rate
-        self.target_layout = target_layout
-        self.resampler: Optional[AudioResampler] = None
-        self.needs_resampling: Optional[bool] = None
-
-    async def _push(self, data: av.AudioFrame):
-        if self.needs_resampling is None:
-            self.needs_resampling = (
-                data.sample_rate != self.target_rate
-                or data.layout.name != self.target_layout
-            )
-
-            if self.needs_resampling:
-                self.resampler = AudioResampler(
-                    format="s16", layout=self.target_layout, rate=self.target_rate
-                )
-
-        if not self.needs_resampling or not self.resampler:
-            await self.emit(data)
-            return
-
-        resampled_frames = self.resampler.resample(copy_frame(data))
-        for resampled_frame in resampled_frames:
-            await self.emit(resampled_frame)
-
-    async def _flush(self):
-        if self.needs_resampling and self.resampler:
-            final_frames = self.resampler.resample(None)
-            for frame in final_frames:
-                await self.emit(frame)
--- a/server/reflector/processors/audio_merge.py
+++ b/server/reflector/processors/audio_merge.py
@@ -16,46 +16,37 @@ class AudioMergeProcessor(Processor):
    INPUT_TYPE = list[av.AudioFrame]
    OUTPUT_TYPE = AudioFile

-    def __init__(self, **kwargs):
-        super().__init__(**kwargs)
-
    async def _push(self, data: list[av.AudioFrame]):
        if not data:
            return

        # get audio information from first frame
        frame = data[0]
-        output_channels = len(frame.layout.channels)
-        output_sample_rate = frame.sample_rate
-        output_sample_width = frame.format.bytes
+        channels = len(frame.layout.channels)
+        sample_rate = frame.sample_rate
+        sample_width = frame.format.bytes

        # create audio file
        uu = uuid4().hex
        fd = io.BytesIO()

-        # Use PyAV to write frames
        out_container = av.open(fd, "w", format="wav")
-        out_stream = out_container.add_stream("pcm_s16le", rate=output_sample_rate)
-        out_stream.layout = frame.layout.name
-
+        out_stream = out_container.add_stream("pcm_s16le", rate=sample_rate)
        for frame in data:
            for packet in out_stream.encode(frame):
                out_container.mux(packet)
-
-        # Flush the encoder
        for packet in out_stream.encode(None):
            out_container.mux(packet)
        out_container.close()
-
        fd.seek(0)

        # emit audio file
        audiofile = AudioFile(
            name=f"{monotonic_ns()}-{uu}.wav",
            fd=fd,
-            sample_rate=output_sample_rate,
-            channels=output_channels,
-            sample_width=output_sample_width,
+            sample_rate=sample_rate,
+            channels=channels,
+            sample_width=sample_width,
            timestamp=data[0].pts * data[0].time_base,
        )

--- a/server/reflector/processors/audio_transcript_modal.py
+++ b/server/reflector/processors/audio_transcript_modal.py
@@ -21,11 +21,7 @@ from reflector.settings import settings


 class AudioTranscriptModalProcessor(AudioTranscriptProcessor):
-    def __init__(
-        self,
-        modal_api_key: str | None = None,
-        **kwargs,
-    ):
+    def __init__(self, modal_api_key: str | None = None, **kwargs):
        super().__init__()
        if not settings.TRANSCRIPT_URL:
            raise Exception(
--- a/server/reflector/processors/base.py
+++ b/server/reflector/processors/base.py
@@ -173,7 +173,6 @@ class Processor(Emitter):
        except Exception:
            self.m_processor_failure.inc()
            self.logger.exception("Error in push")
-            raise

    async def flush(self):
        """
@@ -241,45 +240,33 @@ class ThreadedProcessor(Processor):
        self.INPUT_TYPE = processor.INPUT_TYPE
        self.OUTPUT_TYPE = processor.OUTPUT_TYPE
        self.executor = ThreadPoolExecutor(max_workers=max_workers)
-        self.queue = asyncio.Queue(maxsize=50)
-        self.task: asyncio.Task | None = None
+        self.queue = asyncio.Queue()
+        self.task = asyncio.get_running_loop().create_task(self.loop())

    def set_pipeline(self, pipeline: "Pipeline"):
        super().set_pipeline(pipeline)
        self.processor.set_pipeline(pipeline)

    async def loop(self):
-        try:
-            while True:
-                data = await self.queue.get()
-                self.m_processor_queue.set(self.queue.qsize())
-                with self.m_processor_queue_in_progress.track_inprogress():
+        while True:
+            data = await self.queue.get()
+            self.m_processor_queue.set(self.queue.qsize())
+            with self.m_processor_queue_in_progress.track_inprogress():
+                try:
+                    if data is None:
+                        await self.processor.flush()
+                        break
                    try:
-                        if data is None:
-                            await self.processor.flush()
-                            break
-                        try:
-                            await self.processor.push(data)
-                        except Exception:
-                            self.logger.error(
-                                f"Error in push {self.processor.__class__.__name__}"
-                                ", continue"
-                            )
-                    finally:
-                        self.queue.task_done()
-        except Exception as e:
-            logger.error(f"Crash in {self.__class__.__name__}: {e}", exc_info=e)
-
-    async def _ensure_task(self):
-        if self.task is None:
-            self.task = asyncio.get_running_loop().create_task(self.loop())
-
-        # XXX not doing a sleep here make the whole pipeline prior the thread
-        # to be running without having a chance to work on the task here.
-        await asyncio.sleep(0)
+                        await self.processor.push(data)
+                    except Exception:
+                        self.logger.error(
+                            f"Error in push {self.processor.__class__.__name__}"
+                            ", continue"
+                        )
+                finally:
+                    self.queue.task_done()

    async def _push(self, data):
-        await self._ensure_task()
        await self.queue.put(data)

    async def _flush(self):
--- a/server/reflector/processors/file_diarization.py
+++ b/server/reflector/processors/file_diarization.py
@@ -1,33 +0,0 @@
-from pydantic import BaseModel
-
-from reflector.processors.base import Processor
-from reflector.processors.types import DiarizationSegment
-
-
-class FileDiarizationInput(BaseModel):
-    """Input for file diarization containing audio URL"""
-
-    audio_url: str
-
-
-class FileDiarizationOutput(BaseModel):
-    """Output for file diarization containing speaker segments"""
-
-    diarization: list[DiarizationSegment]
-
-
-class FileDiarizationProcessor(Processor):
-    """
-    Diarize complete audio files from URL
-    """
-
-    INPUT_TYPE = FileDiarizationInput
-    OUTPUT_TYPE = FileDiarizationOutput
-
-    async def _push(self, data: FileDiarizationInput):
-        result = await self._diarize(data)
-        if result:
-            await self.emit(result)
-
-    async def _diarize(self, data: FileDiarizationInput):
-        raise NotImplementedError
--- a/server/reflector/processors/file_diarization_auto.py
+++ b/server/reflector/processors/file_diarization_auto.py
@@ -1,33 +0,0 @@
-import importlib
-
-from reflector.processors.file_diarization import FileDiarizationProcessor
-from reflector.settings import settings
-
-
-class FileDiarizationAutoProcessor(FileDiarizationProcessor):
-    _registry = {}
-
-    @classmethod
-    def register(cls, name, kclass):
-        cls._registry[name] = kclass
-
-    def __new__(cls, name: str | None = None, **kwargs):
-        if name is None:
-            name = settings.DIARIZATION_BACKEND
-
-        if name not in cls._registry:
-            module_name = f"reflector.processors.file_diarization_{name}"
-            importlib.import_module(module_name)
-
-        # gather specific configuration for the processor
-        # search `DIARIZATION_BACKEND_XXX_YYY`, push to constructor as `backend_xxx_yyy`
-        config = {}
-        name_upper = name.upper()
-        settings_prefix = "DIARIZATION_"
-        config_prefix = f"{settings_prefix}{name_upper}_"
-        for key, value in settings:
-            if key.startswith(config_prefix):
-                config_name = key[len(settings_prefix) :].lower()
-                config[config_name] = value
-
-        return cls._registry[name](**config | kwargs)
--- a/server/reflector/processors/file_diarization_modal.py
+++ b/server/reflector/processors/file_diarization_modal.py
@@ -1,58 +0,0 @@
-"""
-File diarization implementation using the GPU service from modal.com
-
-API will be a POST request to DIARIZATION_URL:
-
-```
-POST /diarize?audio_file_url=...&timestamp=0
-Authorization: Bearer <modal_api_key>
-```
-"""
-
-import httpx
-
-from reflector.processors.file_diarization import (
-    FileDiarizationInput,
-    FileDiarizationOutput,
-    FileDiarizationProcessor,
-)
-from reflector.processors.file_diarization_auto import FileDiarizationAutoProcessor
-from reflector.settings import settings
-
-
-class FileDiarizationModalProcessor(FileDiarizationProcessor):
-    def __init__(self, modal_api_key: str | None = None, **kwargs):
-        super().__init__(**kwargs)
-        if not settings.DIARIZATION_URL:
-            raise Exception(
-                "DIARIZATION_URL required to use FileDiarizationModalProcessor"
-            )
-        self.diarization_url = settings.DIARIZATION_URL + "/diarize"
-        self.file_timeout = settings.DIARIZATION_FILE_TIMEOUT
-        self.modal_api_key = modal_api_key
-
-    async def _diarize(self, data: FileDiarizationInput):
-        """Get speaker diarization for file"""
-        self.logger.info(f"Starting diarization from {data.audio_url}")
-
-        headers = {}
-        if self.modal_api_key:
-            headers["Authorization"] = f"Bearer {self.modal_api_key}"
-
-        async with httpx.AsyncClient(timeout=self.file_timeout) as client:
-            response = await client.post(
-                self.diarization_url,
-                headers=headers,
-                params={
-                    "audio_file_url": data.audio_url,
-                    "timestamp": 0,
-                },
-                follow_redirects=True,
-            )
-            response.raise_for_status()
-            diarization_data = response.json()["diarization"]
-
-        return FileDiarizationOutput(diarization=diarization_data)
-
-
-FileDiarizationAutoProcessor.register("modal", FileDiarizationModalProcessor)
--- a/server/reflector/processors/file_transcript.py
+++ b/server/reflector/processors/file_transcript.py
@@ -1,65 +0,0 @@
-from prometheus_client import Counter, Histogram
-
-from reflector.processors.base import Processor
-from reflector.processors.types import Transcript
-
-
-class FileTranscriptInput:
-    """Input for file transcription containing audio URL and language settings"""
-
-    def __init__(self, audio_url: str, language: str = "en"):
-        self.audio_url = audio_url
-        self.language = language
-
-
-class FileTranscriptProcessor(Processor):
-    """
-    Transcript complete audio files from URL
-    """
-
-    INPUT_TYPE = FileTranscriptInput
-    OUTPUT_TYPE = Transcript
-
-    m_transcript = Histogram(
-        "file_transcript",
-        "Time spent in FileTranscript.transcript",
-        ["backend"],
-    )
-    m_transcript_call = Counter(
-        "file_transcript_call",
-        "Number of calls to FileTranscript.transcript",
-        ["backend"],
-    )
-    m_transcript_success = Counter(
-        "file_transcript_success",
-        "Number of successful calls to FileTranscript.transcript",
-        ["backend"],
-    )
-    m_transcript_failure = Counter(
-        "file_transcript_failure",
-        "Number of failed calls to FileTranscript.transcript",
-        ["backend"],
-    )
-
-    def __init__(self, *args, **kwargs):
-        name = self.__class__.__name__
-        self.m_transcript = self.m_transcript.labels(name)
-        self.m_transcript_call = self.m_transcript_call.labels(name)
-        self.m_transcript_success = self.m_transcript_success.labels(name)
-        self.m_transcript_failure = self.m_transcript_failure.labels(name)
-        super().__init__(*args, **kwargs)
-
-    async def _push(self, data: FileTranscriptInput):
-        try:
-            self.m_transcript_call.inc()
-            with self.m_transcript.time():
-                result = await self._transcript(data)
-            self.m_transcript_success.inc()
-            if result:
-                await self.emit(result)
-        except Exception:
-            self.m_transcript_failure.inc()
-            raise
-
-    async def _transcript(self, data: FileTranscriptInput):
-        raise NotImplementedError
--- a/server/reflector/processors/file_transcript_auto.py
+++ b/server/reflector/processors/file_transcript_auto.py
@@ -1,32 +0,0 @@
-import importlib
-
-from reflector.processors.file_transcript import FileTranscriptProcessor
-from reflector.settings import settings
-
-
-class FileTranscriptAutoProcessor(FileTranscriptProcessor):
-    _registry = {}
-
-    @classmethod
-    def register(cls, name, kclass):
-        cls._registry[name] = kclass
-
-    def __new__(cls, name: str | None = None, **kwargs):
-        if name is None:
-            name = settings.TRANSCRIPT_BACKEND
-        if name not in cls._registry:
-            module_name = f"reflector.processors.file_transcript_{name}"
-            importlib.import_module(module_name)
-
-        # gather specific configuration for the processor
-        # search `TRANSCRIPT_BACKEND_XXX_YYY`, push to constructor as `backend_xxx_yyy`
-        config = {}
-        name_upper = name.upper()
-        settings_prefix = "TRANSCRIPT_"
-        config_prefix = f"{settings_prefix}{name_upper}_"
-        for key, value in settings:
-            if key.startswith(config_prefix):
-                config_name = key[len(settings_prefix) :].lower()
-                config[config_name] = value
-
-        return cls._registry[name](**config | kwargs)
--- a/server/reflector/processors/file_transcript_modal.py
+++ b/server/reflector/processors/file_transcript_modal.py
@@ -1,78 +0,0 @@
-"""
-File transcription implementation using the GPU service from modal.com
-
-API will be a POST request to TRANSCRIPT_URL:
-
-```json
-{
-    "audio_file_url": "https://...",
-    "language": "en",
-    "model": "parakeet-tdt-0.6b-v2",
-    "batch": true
-}
-```
-"""
-
-import httpx
-
-from reflector.processors.file_transcript import (
-    FileTranscriptInput,
-    FileTranscriptProcessor,
-)
-from reflector.processors.file_transcript_auto import FileTranscriptAutoProcessor
-from reflector.processors.types import Transcript, Word
-from reflector.settings import settings
-
-
-class FileTranscriptModalProcessor(FileTranscriptProcessor):
-    def __init__(self, modal_api_key: str | None = None, **kwargs):
-        super().__init__(**kwargs)
-        if not settings.TRANSCRIPT_URL:
-            raise Exception(
-                "TRANSCRIPT_URL required to use FileTranscriptModalProcessor"
-            )
-        self.transcript_url = settings.TRANSCRIPT_URL
-        self.file_timeout = settings.TRANSCRIPT_FILE_TIMEOUT
-        self.modal_api_key = modal_api_key
-
-    async def _transcript(self, data: FileTranscriptInput):
-        """Send full file to Modal for transcription"""
-        url = f"{self.transcript_url}/v1/audio/transcriptions-from-url"
-
-        self.logger.info(f"Starting file transcription from {data.audio_url}")
-
-        headers = {}
-        if self.modal_api_key:
-            headers["Authorization"] = f"Bearer {self.modal_api_key}"
-
-        async with httpx.AsyncClient(timeout=self.file_timeout) as client:
-            response = await client.post(
-                url,
-                headers=headers,
-                json={
-                    "audio_file_url": data.audio_url,
-                    "language": data.language,
-                    "batch": True,
-                },
-                follow_redirects=True,
-            )
-            response.raise_for_status()
-            result = response.json()
-
-        words = [
-            Word(
-                text=word_info["word"],
-                start=word_info["start"],
-                end=word_info["end"],
-            )
-            for word_info in result.get("words", [])
-        ]
-
-        # words come not in order
-        words.sort(key=lambda w: w.start)
-
-        return Transcript(words=words)
-
-
-# Register with the auto processor
-FileTranscriptAutoProcessor.register("modal", FileTranscriptModalProcessor)
--- a/server/reflector/processors/transcript_diarization_assembler.py
+++ b/server/reflector/processors/transcript_diarization_assembler.py
@@ -1,45 +0,0 @@
-"""
-Processor to assemble transcript with diarization results
-"""
-
-from reflector.processors.audio_diarization import AudioDiarizationProcessor
-from reflector.processors.base import Processor
-from reflector.processors.types import DiarizationSegment, Transcript
-
-
-class TranscriptDiarizationAssemblerInput:
-    """Input containing transcript and diarization data"""
-
-    def __init__(self, transcript: Transcript, diarization: list[DiarizationSegment]):
-        self.transcript = transcript
-        self.diarization = diarization
-
-
-class TranscriptDiarizationAssemblerProcessor(Processor):
-    """
-    Assemble transcript with diarization results by applying speaker assignments
-    """
-
-    INPUT_TYPE = TranscriptDiarizationAssemblerInput
-    OUTPUT_TYPE = Transcript
-
-    async def _push(self, data: TranscriptDiarizationAssemblerInput):
-        result = await self._assemble(data)
-        if result:
-            await self.emit(result)
-
-    async def _assemble(self, data: TranscriptDiarizationAssemblerInput):
-        """Apply diarization to transcript words"""
-        if not data.diarization:
-            self.logger.info(
-                "No diarization data provided, returning original transcript"
-            )
-            return data.transcript
-
-        # Reuse logic from AudioDiarizationProcessor
-        processor = AudioDiarizationProcessor()
-        words = data.transcript.words
-        processor.assign_speaker(words, data.diarization)
-
-        self.logger.info(f"Applied diarization to {len(words)} words")
-        return data.transcript
--- a/server/reflector/processors/types.py
+++ b/server/reflector/processors/types.py
@@ -2,22 +2,13 @@ import io
 import re
 import tempfile
 from pathlib import Path
-from typing import Annotated, TypedDict
+from typing import Annotated

 from profanityfilter import ProfanityFilter
 from pydantic import BaseModel, Field, PrivateAttr

 from reflector.redis_cache import redis_cache

-
-class DiarizationSegment(TypedDict):
-    """Type definition for diarization segment containing speaker information"""
-
-    start: float
-    end: float
-    speaker: int
-
-
 PUNC_RE = re.compile(r"[.;:?!…]")

 profanity_filter = ProfanityFilter()
--- a/server/reflector/services/ics_sync.py
+++ b/server/reflector/services/ics_sync.py
@@ -0,0 +1,296 @@
+import hashlib
+from datetime import date, datetime, timedelta, timezone
+from typing import TypedDict
+
+import httpx
+import pytz
+from icalendar import Calendar, Event
+from loguru import logger
+
+from reflector.db.calendar_events import CalendarEvent, calendar_events_controller
+from reflector.db.rooms import Room, rooms_controller
+from reflector.settings import settings
+
+
+class AttendeeData(TypedDict, total=False):
+    email: str | None
+    name: str | None
+    status: str | None
+    role: str | None
+
+
+class EventData(TypedDict):
+    ics_uid: str
+    title: str | None
+    description: str | None
+    location: str | None
+    start_time: datetime
+    end_time: datetime
+    attendees: list[AttendeeData]
+    ics_raw_data: str
+
+
+class SyncStats(TypedDict):
+    events_created: int
+    events_updated: int
+    events_deleted: int
+
+
+class ICSFetchService:
+    def __init__(self):
+        self.client = httpx.AsyncClient(
+            timeout=30.0, headers={"User-Agent": "Reflector/1.0"}
+        )
+
+    async def fetch_ics(self, url: str) -> str:
+        response = await self.client.get(url)
+        response.raise_for_status()
+
+        return response.text
+
+    def parse_ics(self, ics_content: str) -> Calendar:
+        return Calendar.from_ical(ics_content)
+
+    def extract_room_events(
+        self, calendar: Calendar, room_name: str, room_url: str
+    ) -> list[EventData]:
+        events = []
+        now = datetime.now(timezone.utc)
+        window_start = now - timedelta(hours=1)
+        window_end = now + timedelta(hours=24)
+
+        for component in calendar.walk():
+            if component.name == "VEVENT":
+                # Skip cancelled events
+                status = component.get("STATUS", "").upper()
+                if status == "CANCELLED":
+                    continue
+
+                # Check if event matches this room
+                if self._event_matches_room(component, room_name, room_url):
+                    event_data = self._parse_event(component)
+
+                    # Only include events in our time window
+                    if (
+                        event_data
+                        and window_start <= event_data["start_time"] <= window_end
+                    ):
+                        events.append(event_data)
+
+        return events
+
+    def _event_matches_room(self, event: Event, room_name: str, room_url: str) -> bool:
+        location = str(event.get("LOCATION", ""))
+        description = str(event.get("DESCRIPTION", ""))
+
+        # Only match full room URL (with or without protocol)
+        patterns = [
+            room_url,  # Full URL with protocol
+            room_url.replace("https://", ""),  # Without https protocol
+            room_url.replace("http://", ""),  # Without http protocol
+        ]
+
+        # Check location and description for patterns
+        text_to_check = f"{location} {description}".lower()
+
+        for pattern in patterns:
+            if pattern.lower() in text_to_check:
+                return True
+
+        return False
+
+    def _parse_event(self, event: Event) -> EventData | None:
+        # Extract basic fields
+        uid = str(event.get("UID", ""))
+        summary = str(event.get("SUMMARY", ""))
+        description = str(event.get("DESCRIPTION", ""))
+        location = str(event.get("LOCATION", ""))
+
+        # Parse dates
+        dtstart = event.get("DTSTART")
+        dtend = event.get("DTEND")
+
+        if not dtstart:
+            return None
+
+        # Convert to datetime
+        start_time = self._normalize_datetime(
+            dtstart.dt if hasattr(dtstart, "dt") else dtstart
+        )
+        end_time = (
+            self._normalize_datetime(dtend.dt if hasattr(dtend, "dt") else dtend)
+            if dtend
+            else start_time + timedelta(hours=1)
+        )
+
+        # Parse attendees
+        attendees = self._parse_attendees(event)
+
+        # Get raw event data for storage
+        raw_data = event.to_ical().decode("utf-8")
+
+        return {
+            "ics_uid": uid,
+            "title": summary,
+            "description": description,
+            "location": location,
+            "start_time": start_time,
+            "end_time": end_time,
+            "attendees": attendees,
+            "ics_raw_data": raw_data,
+        }
+
+    def _normalize_datetime(self, dt) -> datetime:
+        # Handle date objects (all-day events)
+        if isinstance(dt, date) and not isinstance(dt, datetime):
+            # Convert to datetime at start of day in UTC
+            dt = datetime.combine(dt, datetime.min.time())
+            dt = pytz.UTC.localize(dt)
+        elif isinstance(dt, datetime):
+            # Add UTC timezone if naive
+            if dt.tzinfo is None:
+                dt = pytz.UTC.localize(dt)
+            else:
+                # Convert to UTC
+                dt = dt.astimezone(pytz.UTC)
+
+        return dt
+
+    def _parse_attendees(self, event: Event) -> list[AttendeeData]:
+        attendees = []
+
+        # Parse ATTENDEE properties
+        for attendee in event.get("ATTENDEE", []):
+            if not isinstance(attendee, list):
+                attendee = [attendee]
+
+            for att in attendee:
+                att_data: AttendeeData = {
+                    "email": str(att).replace("mailto:", "") if att else None,
+                    "name": att.params.get("CN") if hasattr(att, "params") else None,
+                    "status": att.params.get("PARTSTAT")
+                    if hasattr(att, "params")
+                    else None,
+                    "role": att.params.get("ROLE") if hasattr(att, "params") else None,
+                }
+                attendees.append(att_data)
+
+        # Add organizer
+        organizer = event.get("ORGANIZER")
+        if organizer:
+            org_data: AttendeeData = {
+                "email": str(organizer).replace("mailto:", "") if organizer else None,
+                "name": organizer.params.get("CN")
+                if hasattr(organizer, "params")
+                else None,
+                "role": "ORGANIZER",
+            }
+            attendees.append(org_data)
+
+        return attendees
+
+
+class ICSSyncService:
+    def __init__(self):
+        self.fetch_service = ICSFetchService()
+
+    async def sync_room_calendar(self, room: Room) -> dict:
+        if not room.ics_enabled or not room.ics_url:
+            return {"status": "skipped", "reason": "ICS not configured"}
+
+        try:
+            # Check if it's time to sync
+            if not self._should_sync(room):
+                return {"status": "skipped", "reason": "Not time to sync yet"}
+
+            # Fetch ICS file
+            ics_content = await self.fetch_service.fetch_ics(room.ics_url)
+
+            # Check if content changed
+            content_hash = hashlib.md5(ics_content.encode()).hexdigest()
+            if room.ics_last_etag == content_hash:
+                logger.info(f"No changes in ICS for room {room.id}")
+                return {"status": "unchanged", "hash": content_hash}
+
+            # Parse calendar
+            calendar = self.fetch_service.parse_ics(ics_content)
+
+            # Build room URL
+            room_url = f"{settings.BASE_URL}/room/{room.name}"
+
+            # Extract matching events
+            events = self.fetch_service.extract_room_events(
+                calendar, room.name, room_url
+            )
+
+            # Sync events to database
+            sync_result = await self._sync_events_to_database(room.id, events)
+
+            # Update room sync metadata
+            await rooms_controller.update(
+                room,
+                {
+                    "ics_last_sync": datetime.now(timezone.utc),
+                    "ics_last_etag": content_hash,
+                },
+                mutate=False,
+            )
+
+            return {
+                "status": "success",
+                "hash": content_hash,
+                "events_found": len(events),
+                **sync_result,
+            }
+
+        except Exception as e:
+            logger.error(f"Failed to sync ICS for room {room.id}: {e}")
+            return {"status": "error", "error": str(e)}
+
+    def _should_sync(self, room: Room) -> bool:
+        if not room.ics_last_sync:
+            return True
+
+        time_since_sync = datetime.now(timezone.utc) - room.ics_last_sync
+        return time_since_sync.total_seconds() >= room.ics_fetch_interval
+
+    async def _sync_events_to_database(
+        self, room_id: str, events: list[EventData]
+    ) -> SyncStats:
+        created = 0
+        updated = 0
+
+        # Track current event IDs
+        current_ics_uids = []
+
+        for event_data in events:
+            # Create CalendarEvent object
+            calendar_event = CalendarEvent(room_id=room_id, **event_data)
+
+            # Upsert event
+            existing = await calendar_events_controller.get_by_ics_uid(
+                room_id, event_data["ics_uid"]
+            )
+
+            if existing:
+                updated += 1
+            else:
+                created += 1
+
+            await calendar_events_controller.upsert(calendar_event)
+            current_ics_uids.append(event_data["ics_uid"])
+
+        # Soft delete events that are no longer in calendar
+        deleted = await calendar_events_controller.soft_delete_missing(
+            room_id, current_ics_uids
+        )
+
+        return {
+            "events_created": created,
+            "events_updated": updated,
+            "events_deleted": deleted,
+        }
+
+
+# Global instance
+ics_sync_service = ICSSyncService()
--- a/server/reflector/settings.py
+++ b/server/reflector/settings.py
@@ -1,8 +1,5 @@
-from pydantic.types import PositiveInt
 from pydantic_settings import BaseSettings, SettingsConfigDict

-from reflector.utils.string import NonEmptyString
-

 class Settings(BaseSettings):
    model_config = SettingsConfigDict(
@@ -24,16 +21,11 @@ class Settings(BaseSettings):
    # local data directory
    DATA_DIR: str = "./data"

-    # Audio Chunking
-    # backends: silero, frames
-    AUDIO_CHUNKER_BACKEND: str = "frames"
-
    # Audio Transcription
    # backends: whisper, modal
    TRANSCRIPT_BACKEND: str = "whisper"
    TRANSCRIPT_URL: str | None = None
    TRANSCRIPT_TIMEOUT: int = 90
-    TRANSCRIPT_FILE_TIMEOUT: int = 600

    # Audio Transcription: modal backend
    TRANSCRIPT_MODAL_API_KEY: str | None = None
@@ -74,14 +66,10 @@ class Settings(BaseSettings):
    DIARIZATION_ENABLED: bool = True
    DIARIZATION_BACKEND: str = "modal"
    DIARIZATION_URL: str | None = None
-    DIARIZATION_FILE_TIMEOUT: int = 600

    # Diarization: modal backend
    DIARIZATION_MODAL_API_KEY: str | None = None

-    # Diarization: local pyannote.audio
-    DIARIZATION_PYANNOTE_AUTH_TOKEN: str | None = None
-
    # Sentry
    SENTRY_DSN: str | None = None

@@ -93,8 +81,9 @@ class Settings(BaseSettings):
    AUTH_JWT_PUBLIC_KEY: str | None = "authentik.monadical.com_public.pem"
    AUTH_JWT_AUDIENCE: str | None = None

+    # API public mode
+    # if set, all anonymous record will be public
    PUBLIC_MODE: bool = False
-    PUBLIC_DATA_RETENTION_DAYS: PositiveInt = 7

    # Min transcript length to generate topic + summary
    MIN_TRANSCRIPT_LENGTH: int = 750
@@ -122,24 +111,13 @@ class Settings(BaseSettings):

    # Whereby integration
    WHEREBY_API_URL: str = "https://api.whereby.dev/v1"
-    WHEREBY_API_KEY: NonEmptyString | None = None
-
-    # Jibri integration
-    JIBRI_RECORDINGS_PATH: str = "/recordings"
+    WHEREBY_API_KEY: str | None = None
    WHEREBY_WEBHOOK_SECRET: str | None = None
    AWS_WHEREBY_ACCESS_KEY_ID: str | None = None
    AWS_WHEREBY_ACCESS_KEY_SECRET: str | None = None
    AWS_PROCESS_RECORDING_QUEUE_URL: str | None = None
    SQS_POLLING_TIMEOUT_SECONDS: int = 60

-    # Jitsi Meet
-    JITSI_DOMAIN: str = "meet.jit.si"
-    JITSI_JWT_SECRET: str | None = None
-    JITSI_WEBHOOK_SECRET: str | None = None
-    JITSI_APP_ID: str = "reflector"
-    JITSI_JWT_ISSUER: str = "reflector"
-    JITSI_JWT_AUDIENCE: str = "jitsi"
-
    # Zulip integration
    ZULIP_REALM: str | None = None
    ZULIP_API_KEY: str | None = None
--- a/server/reflector/tools/cleanup_old_data.py
+++ b/server/reflector/tools/cleanup_old_data.py
@@ -1,72 +0,0 @@
-#!/usr/bin/env python
-"""
-Manual cleanup tool for old public data.
-Uses the same implementation as the Celery worker task.
-"""
-
-import argparse
-import asyncio
-import sys
-
-import structlog
-
-from reflector.settings import settings
-from reflector.worker.cleanup import _cleanup_old_public_data
-
-logger = structlog.get_logger(__name__)
-
-
-async def cleanup_old_data(days: int = 7):
-    logger.info(
-        "Starting manual cleanup",
-        retention_days=days,
-        public_mode=settings.PUBLIC_MODE,
-    )
-
-    if not settings.PUBLIC_MODE:
-        logger.critical(
-            "WARNING: PUBLIC_MODE is False. "
-            "This tool is intended for public instances only."
-        )
-        raise Exception("Tool intended for public instances only")
-
-    result = await _cleanup_old_public_data(days=days)
-
-    if result:
-        logger.info(
-            "Cleanup completed",
-            transcripts_deleted=result.get("transcripts_deleted", 0),
-            meetings_deleted=result.get("meetings_deleted", 0),
-            recordings_deleted=result.get("recordings_deleted", 0),
-            errors_count=len(result.get("errors", [])),
-        )
-        if result.get("errors"):
-            logger.warning(
-                "Errors encountered during cleanup:", errors=result["errors"][:10]
-            )
-    else:
-        logger.info("Cleanup skipped or completed without results")
-
-
-def main():
-    parser = argparse.ArgumentParser(
-        description="Clean up old transcripts and meetings"
-    )
-    parser.add_argument(
-        "--days",
-        type=int,
-        default=7,
-        help="Number of days to keep data (default: 7)",
-    )
-
-    args = parser.parse_args()
-
-    if args.days < 1:
-        logger.error("Days must be at least 1")
-        sys.exit(1)
-
-    asyncio.run(cleanup_old_data(days=args.days))
-
-
-if __name__ == "__main__":
-    main()
--- a/server/reflector/tools/process.py
+++ b/server/reflector/tools/process.py
@@ -1,220 +1,105 @@
-"""
-Process audio file with diarization support
-"""
-
-import argparse
 import asyncio
-import json
-import shutil
-import sys
-import time
-from pathlib import Path
-from typing import Any, Dict, List, Literal

-from reflector.db.transcripts import SourceKind, TranscriptTopic, transcripts_controller
+import av
+
 from reflector.logger import logger
-from reflector.pipelines.main_file_pipeline import (
-    task_pipeline_file_process as task_pipeline_file_process,
-)
-from reflector.pipelines.main_live_pipeline import pipeline_post as live_pipeline_post
-from reflector.pipelines.main_live_pipeline import (
-    pipeline_process as live_pipeline_process,
+from reflector.processors import (
+    AudioChunkerProcessor,
+    AudioMergeProcessor,
+    AudioTranscriptAutoProcessor,
+    Pipeline,
+    PipelineEvent,
+    TranscriptFinalSummaryProcessor,
+    TranscriptFinalTitleProcessor,
+    TranscriptLinerProcessor,
+    TranscriptTopicDetectorProcessor,
+    TranscriptTranslatorAutoProcessor,
 )
+from reflector.processors.base import BroadcastProcessor


-def serialize_topics(topics: List[TranscriptTopic]) -> List[Dict[str, Any]]:
-    """Convert TranscriptTopic objects to JSON-serializable dicts"""
-    serialized = []
-    for topic in topics:
-        topic_dict = topic.model_dump()
-        serialized.append(topic_dict)
-    return serialized
-
-
-def debug_print_speakers(serialized_topics: List[Dict[str, Any]]) -> None:
-    """Print debug info about speakers found in topics"""
-    all_speakers = set()
-    for topic_dict in serialized_topics:
-        for word in topic_dict.get("words", []):
-            all_speakers.add(word.get("speaker", 0))
-
-    print(
-        f"Found {len(serialized_topics)} topics with speakers: {all_speakers}",
-        file=sys.stderr,
-    )
-
-
-TranscriptId = str
-
-
-# common interface for every flow: it needs an Entry in db with specific ceremony (file path + status + actual file in file system)
-# ideally we want to get rid of it at some point
-async def prepare_entry(
-    source_path: str,
-    source_language: str,
-    target_language: str,
-) -> TranscriptId:
-    file_path = Path(source_path)
-
-    transcript = await transcripts_controller.add(
-        file_path.name,
-        # note that the real file upload has SourceKind: LIVE for the reason of it's an error
-        source_kind=SourceKind.FILE,
-        source_language=source_language,
-        target_language=target_language,
-        user_id=None,
-    )
-
-    logger.info(
-        f"Created empty transcript {transcript.id} for file {file_path.name} because technically we need an empty transcript before we start transcript"
-    )
-
-    # pipelines expect files as upload.*
-
-    extension = file_path.suffix
-    upload_path = transcript.data_path / f"upload{extension}"
-    upload_path.parent.mkdir(parents=True, exist_ok=True)
-    shutil.copy2(source_path, upload_path)
-    logger.info(f"Copied {source_path} to {upload_path}")
-
-    # pipelines expect entity status "uploaded"
-    await transcripts_controller.update(transcript, {"status": "uploaded"})
-
-    return transcript.id
-
-
-# same reason as prepare_entry
-async def extract_result_from_entry(
-    transcript_id: TranscriptId, output_path: str
-) -> None:
-    post_final_transcript = await transcripts_controller.get_by_id(transcript_id)
-
-    # assert post_final_transcript.status == "ended"
-    # File pipeline doesn't set status to "ended", only live pipeline does https://github.com/Monadical-SAS/reflector/issues/582
-    topics = post_final_transcript.topics
-    if not topics:
-        raise RuntimeError(
-            f"No topics found for transcript {transcript_id} after processing"
-        )
-
-    serialized_topics = serialize_topics(topics)
-
-    if output_path:
-        # Write to JSON file
-        with open(output_path, "w") as f:
-            for topic_dict in serialized_topics:
-                json.dump(topic_dict, f)
-                f.write("\n")
-        print(f"Results written to {output_path}", file=sys.stderr)
-    else:
-        # Write to stdout as JSONL
-        for topic_dict in serialized_topics:
-            print(json.dumps(topic_dict))
-
-    debug_print_speakers(serialized_topics)
-
-
-async def process_live_pipeline(
-    transcript_id: TranscriptId,
+async def process_audio_file(
+    filename,
+    event_callback,
+    only_transcript=False,
+    source_language="en",
+    target_language="en",
 ):
-    """Process transcript_id with transcription and diarization"""
+    # build pipeline for audio processing
+    processors = [
+        AudioChunkerProcessor(),
+        AudioMergeProcessor(),
+        AudioTranscriptAutoProcessor.as_threaded(),
+        TranscriptLinerProcessor(),
+        TranscriptTranslatorAutoProcessor.as_threaded(),
+    ]
+    if not only_transcript:
+        processors += [
+            TranscriptTopicDetectorProcessor.as_threaded(),
+            BroadcastProcessor(
+                processors=[
+                    TranscriptFinalTitleProcessor.as_threaded(),
+                    TranscriptFinalSummaryProcessor.as_threaded(),
+                ],
+            ),
+        ]

-    print(f"Processing transcript_id {transcript_id}...", file=sys.stderr)
-    await live_pipeline_process(transcript_id=transcript_id)
-    print(f"Processing complete for transcript {transcript_id}", file=sys.stderr)
-
-    pre_final_transcript = await transcripts_controller.get_by_id(transcript_id)
-
-    # assert documented behaviour: after process, the pipeline isn't ended. this is the reason of calling pipeline_post
-    assert pre_final_transcript.status != "ended"
-
-    # at this point, diarization is running but we have no access to it. run diarization in parallel - one will hopefully win after polling
-    result = live_pipeline_post(transcript_id=transcript_id)
-
-    # result.ready() blocks even without await; it mutates result also
-    while not result.ready():
-        print(f"Status: {result.state}")
-        time.sleep(2)
-
-
-async def process_file_pipeline(
-    transcript_id: TranscriptId,
-):
-    """Process audio/video file using the optimized file pipeline"""
-
-    # task_pipeline_file_process is a Celery task, need to use .delay() for async execution
-    result = task_pipeline_file_process.delay(transcript_id=transcript_id)
-
-    # Wait for the Celery task to complete
-    while not result.ready():
-        print(f"File pipeline status: {result.state}", file=sys.stderr)
-        time.sleep(2)
-
-    logger.info("File pipeline processing complete")
-
-
-async def process(
-    source_path: str,
-    source_language: str,
-    target_language: str,
-    pipeline: Literal["live", "file"],
-    output_path: str = None,
-):
-    from reflector.db import get_database
-
-    database = get_database()
-    # db connect is a part of ceremony
-    await database.connect()
+    # transcription output
+    pipeline = Pipeline(*processors)
+    pipeline.set_pref("audio:source_language", source_language)
+    pipeline.set_pref("audio:target_language", target_language)
+    pipeline.describe()
+    pipeline.on(event_callback)

+    # start processing audio
+    logger.info(f"Opening {filename}")
+    container = av.open(filename)
    try:
-        transcript_id = await prepare_entry(
-            source_path,
-            source_language,
-            target_language,
-        )
-
-        pipeline_handlers = {
-            "live": process_live_pipeline,
-            "file": process_file_pipeline,
-        }
-
-        handler = pipeline_handlers.get(pipeline)
-        if not handler:
-            raise ValueError(f"Unknown pipeline type: {pipeline}")
-
-        await handler(transcript_id)
-
-        await extract_result_from_entry(transcript_id, output_path)
+        logger.info("Start pushing audio into the pipeline")
+        for frame in container.decode(audio=0):
+            await pipeline.push(frame)
    finally:
-        await database.disconnect()
+        logger.info("Flushing the pipeline")
+        await pipeline.flush()
+
+    logger.info("All done !")


 if __name__ == "__main__":
-    parser = argparse.ArgumentParser(
-        description="Process audio files with speaker diarization"
-    )
+    import argparse
+
+    parser = argparse.ArgumentParser()
    parser.add_argument("source", help="Source file (mp3, wav, mp4...)")
-    parser.add_argument(
-        "--pipeline",
-        required=True,
-        choices=["live", "file"],
-        help="Pipeline type to use for processing (live: streaming/incremental, file: batch/parallel)",
-    )
-    parser.add_argument(
-        "--source-language", default="en", help="Source language code (default: en)"
-    )
-    parser.add_argument(
-        "--target-language", default="en", help="Target language code (default: en)"
-    )
+    parser.add_argument("--only-transcript", "-t", action="store_true")
+    parser.add_argument("--source-language", default="en")
+    parser.add_argument("--target-language", default="en")
    parser.add_argument("--output", "-o", help="Output file (output.jsonl)")
    args = parser.parse_args()

+    output_fd = None
+    if args.output:
+        output_fd = open(args.output, "w")
+
+    async def event_callback(event: PipelineEvent):
+        processor = event.processor
+        # ignore some processor
+        if processor in ("AudioChunkerProcessor", "AudioMergeProcessor"):
+            return
+        logger.info(f"Event: {event}")
+        if output_fd:
+            output_fd.write(event.model_dump_json())
+            output_fd.write("\n")
+
    asyncio.run(
-        process(
+        process_audio_file(
            args.source,
-            args.source_language,
-            args.target_language,
-            args.pipeline,
-            args.output,
+            event_callback,
+            only_transcript=args.only_transcript,
+            source_language=args.source_language,
+            target_language=args.target_language,
        )
    )
+
+    if output_fd:
+        output_fd.close()
+        logger.info(f"Output written to {args.output}")
--- a/server/reflector/tools/process_with_diarization.py
+++ b/server/reflector/tools/process_with_diarization.py
@@ -0,0 +1,315 @@
+"""
+@vibe-generated
+Process audio file with diarization support
+===========================================
+
+Extended version of process.py that includes speaker diarization.
+This tool processes audio files locally without requiring the full server infrastructure.
+"""
+
+import asyncio
+import tempfile
+import uuid
+from pathlib import Path
+from typing import List
+
+import av
+
+from reflector.logger import logger
+from reflector.processors import (
+    AudioChunkerProcessor,
+    AudioFileWriterProcessor,
+    AudioMergeProcessor,
+    AudioTranscriptAutoProcessor,
+    Pipeline,
+    PipelineEvent,
+    TranscriptFinalSummaryProcessor,
+    TranscriptFinalTitleProcessor,
+    TranscriptLinerProcessor,
+    TranscriptTopicDetectorProcessor,
+    TranscriptTranslatorAutoProcessor,
+)
+from reflector.processors.base import BroadcastProcessor, Processor
+from reflector.processors.types import (
+    AudioDiarizationInput,
+    TitleSummary,
+    TitleSummaryWithId,
+)
+
+
+class TopicCollectorProcessor(Processor):
+    """Collect topics for diarization"""
+
+    INPUT_TYPE = TitleSummary
+    OUTPUT_TYPE = TitleSummary
+
+    def __init__(self, **kwargs):
+        super().__init__(**kwargs)
+        self.topics: List[TitleSummaryWithId] = []
+        self._topic_id = 0
+
+    async def _push(self, data: TitleSummary):
+        # Convert to TitleSummaryWithId and collect
+        self._topic_id += 1
+        topic_with_id = TitleSummaryWithId(
+            id=str(self._topic_id),
+            title=data.title,
+            summary=data.summary,
+            timestamp=data.timestamp,
+            duration=data.duration,
+            transcript=data.transcript,
+        )
+        self.topics.append(topic_with_id)
+
+        # Pass through the original topic
+        await self.emit(data)
+
+    def get_topics(self) -> List[TitleSummaryWithId]:
+        return self.topics
+
+
+async def process_audio_file_with_diarization(
+    filename,
+    event_callback,
+    only_transcript=False,
+    source_language="en",
+    target_language="en",
+    enable_diarization=True,
+    diarization_backend="modal",
+):
+    # Create temp file for audio if diarization is enabled
+    audio_temp_path = None
+    if enable_diarization:
+        audio_temp_file = tempfile.NamedTemporaryFile(suffix=".wav", delete=False)
+        audio_temp_path = audio_temp_file.name
+        audio_temp_file.close()
+
+    # Create processor for collecting topics
+    topic_collector = TopicCollectorProcessor()
+
+    # Build pipeline for audio processing
+    processors = []
+
+    # Add audio file writer at the beginning if diarization is enabled
+    if enable_diarization:
+        processors.append(AudioFileWriterProcessor(audio_temp_path))
+
+    # Add the rest of the processors
+    processors += [
+        AudioChunkerProcessor(),
+        AudioMergeProcessor(),
+        AudioTranscriptAutoProcessor.as_threaded(),
+    ]
+
+    processors += [
+        TranscriptLinerProcessor(),
+        TranscriptTranslatorAutoProcessor.as_threaded(),
+    ]
+
+    if not only_transcript:
+        processors += [
+            TranscriptTopicDetectorProcessor.as_threaded(),
+            # Collect topics for diarization
+            topic_collector,
+            BroadcastProcessor(
+                processors=[
+                    TranscriptFinalTitleProcessor.as_threaded(),
+                    TranscriptFinalSummaryProcessor.as_threaded(),
+                ],
+            ),
+        ]
+
+    # Create main pipeline
+    pipeline = Pipeline(*processors)
+    pipeline.set_pref("audio:source_language", source_language)
+    pipeline.set_pref("audio:target_language", target_language)
+    pipeline.describe()
+    pipeline.on(event_callback)
+
+    # Start processing audio
+    logger.info(f"Opening {filename}")
+    container = av.open(filename)
+    try:
+        logger.info("Start pushing audio into the pipeline")
+        for frame in container.decode(audio=0):
+            await pipeline.push(frame)
+    finally:
+        logger.info("Flushing the pipeline")
+        await pipeline.flush()
+
+    # Run diarization if enabled and we have topics
+    if enable_diarization and not only_transcript and audio_temp_path:
+        topics = topic_collector.get_topics()
+
+        if topics:
+            logger.info(f"Starting diarization with {len(topics)} topics")
+
+            try:
+                from reflector.processors import AudioDiarizationAutoProcessor
+
+                diarization_processor = AudioDiarizationAutoProcessor(
+                    name=diarization_backend
+                )
+
+                diarization_processor.set_pipeline(pipeline)
+
+                # For Modal backend, we need to upload the file to S3 first
+                if diarization_backend == "modal":
+                    from datetime import datetime, timezone
+
+                    from reflector.storage import get_transcripts_storage
+                    from reflector.utils.s3_temp_file import S3TemporaryFile
+
+                    storage = get_transcripts_storage()
+
+                    # Generate a unique filename in evaluation folder
+                    timestamp = datetime.now(timezone.utc).strftime("%Y%m%d_%H%M%S")
+                    audio_filename = f"evaluation/diarization_temp/{timestamp}_{uuid.uuid4().hex}.wav"
+
+                    # Use context manager for automatic cleanup
+                    async with S3TemporaryFile(storage, audio_filename) as s3_file:
+                        # Read and upload the audio file
+                        with open(audio_temp_path, "rb") as f:
+                            audio_data = f.read()
+
+                        audio_url = await s3_file.upload(audio_data)
+                        logger.info(f"Uploaded audio to S3: {audio_filename}")
+
+                        # Create diarization input with S3 URL
+                        diarization_input = AudioDiarizationInput(
+                            audio_url=audio_url, topics=topics
+                        )
+
+                        # Run diarization
+                        await diarization_processor.push(diarization_input)
+                        await diarization_processor.flush()
+
+                        logger.info("Diarization complete")
+                        # File will be automatically cleaned up when exiting the context
+                else:
+                    # For local backend, use local file path
+                    audio_url = audio_temp_path
+
+                    # Create diarization input
+                    diarization_input = AudioDiarizationInput(
+                        audio_url=audio_url, topics=topics
+                    )
+
+                    # Run diarization
+                    await diarization_processor.push(diarization_input)
+                    await diarization_processor.flush()
+
+                    logger.info("Diarization complete")
+
+            except ImportError as e:
+                logger.error(f"Failed to import diarization dependencies: {e}")
+                logger.error(
+                    "Install with: uv pip install pyannote.audio torch torchaudio"
+                )
+                logger.error(
+                    "And set HF_TOKEN environment variable for pyannote models"
+                )
+                raise SystemExit(1)
+            except Exception as e:
+                logger.error(f"Diarization failed: {e}")
+                raise SystemExit(1)
+        else:
+            logger.warning("Skipping diarization: no topics available")
+
+    # Clean up temp file
+    if audio_temp_path:
+        try:
+            Path(audio_temp_path).unlink()
+        except Exception as e:
+            logger.warning(f"Failed to clean up temp file {audio_temp_path}: {e}")
+
+    logger.info("All done!")
+
+
+if __name__ == "__main__":
+    import argparse
+    import os
+
+    parser = argparse.ArgumentParser(
+        description="Process audio files with optional speaker diarization"
+    )
+    parser.add_argument("source", help="Source file (mp3, wav, mp4...)")
+    parser.add_argument(
+        "--only-transcript",
+        "-t",
+        action="store_true",
+        help="Only generate transcript without topics/summaries",
+    )
+    parser.add_argument(
+        "--source-language", default="en", help="Source language code (default: en)"
+    )
+    parser.add_argument(
+        "--target-language", default="en", help="Target language code (default: en)"
+    )
+    parser.add_argument("--output", "-o", help="Output file (output.jsonl)")
+    parser.add_argument(
+        "--enable-diarization",
+        "-d",
+        action="store_true",
+        help="Enable speaker diarization",
+    )
+    parser.add_argument(
+        "--diarization-backend",
+        default="modal",
+        choices=["modal"],
+        help="Diarization backend to use (default: modal)",
+    )
+    args = parser.parse_args()
+
+    # Set REDIS_HOST to localhost if not provided
+    if "REDIS_HOST" not in os.environ:
+        os.environ["REDIS_HOST"] = "localhost"
+        logger.info("REDIS_HOST not set, defaulting to localhost")
+
+    output_fd = None
+    if args.output:
+        output_fd = open(args.output, "w")
+
+    async def event_callback(event: PipelineEvent):
+        processor = event.processor
+        data = event.data
+
+        # Ignore internal processors
+        if processor in (
+            "AudioChunkerProcessor",
+            "AudioMergeProcessor",
+            "AudioFileWriterProcessor",
+            "TopicCollectorProcessor",
+            "BroadcastProcessor",
+        ):
+            return
+
+        # If diarization is enabled, skip the original topic events from the pipeline
+        # The diarization processor will emit the same topics but with speaker info
+        if processor == "TranscriptTopicDetectorProcessor" and args.enable_diarization:
+            return
+
+        # Log all events
+        logger.info(f"Event: {processor} - {type(data).__name__}")
+
+        # Write to output
+        if output_fd:
+            output_fd.write(event.model_dump_json())
+            output_fd.write("\n")
+            output_fd.flush()
+
+    asyncio.run(
+        process_audio_file_with_diarization(
+            args.source,
+            event_callback,
+            only_transcript=args.only_transcript,
+            source_language=args.source_language,
+            target_language=args.target_language,
+            enable_diarization=args.enable_diarization,
+            diarization_backend=args.diarization_backend,
+        )
+    )
+
+    if output_fd:
+        output_fd.close()
+        logger.info(f"Output written to {args.output}")
--- a/server/reflector/tools/runpipeline.py
+++ b/server/reflector/tools/runpipeline.py
@@ -53,7 +53,7 @@ async def run_single_processor(args):
    async def event_callback(event: PipelineEvent):
        processor = event.processor
        # ignore some processor
-        if processor in ("AudioChunkerAutoProcessor", "AudioMergeProcessor"):
+        if processor in ("AudioChunkerProcessor", "AudioMergeProcessor"):
            return
        print(f"Event: {event}")
        if output_fd:
--- a/server/reflector/tools/test_diarization.py
+++ b/server/reflector/tools/test_diarization.py
@@ -0,0 +1,96 @@
+#!/usr/bin/env python3
+"""
+@vibe-generated
+Test script for the diarization CLI tool
+=========================================
+
+This script helps test the diarization functionality with sample audio files.
+"""
+
+import asyncio
+import sys
+from pathlib import Path
+
+from reflector.logger import logger
+
+
+async def test_diarization(audio_file: str):
+    """Test the diarization functionality"""
+
+    # Import the processing function
+    from process_with_diarization import process_audio_file_with_diarization
+
+    # Collect events
+    events = []
+
+    async def event_callback(event):
+        events.append({"processor": event.processor, "data": event.data})
+        logger.info(f"Event from {event.processor}")
+
+    # Process the audio file
+    logger.info(f"Processing audio file: {audio_file}")
+
+    try:
+        await process_audio_file_with_diarization(
+            audio_file,
+            event_callback,
+            only_transcript=False,
+            source_language="en",
+            target_language="en",
+            enable_diarization=True,
+            diarization_backend="modal",
+        )
+
+        # Analyze results
+        logger.info(f"Processing complete. Received {len(events)} events")
+
+        # Look for diarization results
+        diarized_topics = []
+        for event in events:
+            if "TitleSummary" in event["processor"]:
+                # Check if words have speaker information
+                if hasattr(event["data"], "transcript") and event["data"].transcript:
+                    words = event["data"].transcript.words
+                    if words and hasattr(words[0], "speaker"):
+                        speakers = set(
+                            w.speaker for w in words if hasattr(w, "speaker")
+                        )
+                        logger.info(
+                            f"Found {len(speakers)} speakers in topic: {event['data'].title}"
+                        )
+                        diarized_topics.append(event["data"])
+
+        if diarized_topics:
+            logger.info(f"Successfully diarized {len(diarized_topics)} topics")
+
+            # Print sample output
+            sample_topic = diarized_topics[0]
+            logger.info("Sample diarized output:")
+            for i, word in enumerate(sample_topic.transcript.words[:10]):
+                logger.info(f"  Word {i}: '{word.text}' - Speaker {word.speaker}")
+        else:
+            logger.warning("No diarization results found in output")
+
+        return events
+
+    except Exception as e:
+        logger.error(f"Error during processing: {e}")
+        raise
+
+
+def main():
+    if len(sys.argv) < 2:
+        print("Usage: python test_diarization.py <audio_file>")
+        sys.exit(1)
+
+    audio_file = sys.argv[1]
+    if not Path(audio_file).exists():
+        print(f"Error: Audio file '{audio_file}' not found")
+        sys.exit(1)
+
+    # Run the test
+    asyncio.run(test_diarization(audio_file))
+
+
+if __name__ == "__main__":
+    main()
--- a/server/reflector/utils/string.py
+++ b/server/reflector/utils/string.py
@@ -1,23 +0,0 @@
-from typing import Annotated
-
-from pydantic import Field, TypeAdapter, constr
-
-NonEmptyStringBase = constr(min_length=1, strip_whitespace=False)
-NonEmptyString = Annotated[
-    NonEmptyStringBase,
-    Field(description="A non-empty string", min_length=1),
-]
-non_empty_string_adapter = TypeAdapter(NonEmptyString)
-
-
-def parse_non_empty_string(s: str, error: str | None = None) -> NonEmptyString:
-    try:
-        return non_empty_string_adapter.validate_python(s)
-    except Exception as e:
-        raise ValueError(f"{e}: {error}" if error else e) from e
-
-
-def try_parse_non_empty_string(s: str) -> NonEmptyString | None:
-    if not s:
-        return None
-    return parse_non_empty_string(s)
--- a/server/reflector/video_platforms/init.py
+++ b/server/reflector/video_platforms/init.py
@@ -1,17 +0,0 @@
-# Video Platform Abstraction Layer
-"""
-This module provides an abstraction layer for different video conferencing platforms.
-It allows seamless switching between providers (Whereby, Daily.co, etc.) without
-changing the core application logic.
-"""
-
-from .base import MeetingData, VideoPlatformClient, VideoPlatformConfig
-from .registry import get_platform_client, register_platform
-
-__all__ = [
-    "VideoPlatformClient",
-    "VideoPlatformConfig",
-    "MeetingData",
-    "get_platform_client",
-    "register_platform",
-]
--- a/server/reflector/video_platforms/base.py
+++ b/server/reflector/video_platforms/base.py
@@ -1,82 +0,0 @@
-from abc import ABC, abstractmethod
-from datetime import datetime
-from typing import Any, Dict, Optional
-
-from pydantic import BaseModel
-
-from reflector.db.rooms import Room
-
-
-class MeetingData(BaseModel):
-    """Standardized meeting data returned by all platforms."""
-
-    meeting_id: str
-    room_name: str
-    room_url: str
-    host_room_url: str
-    platform: str
-    extra_data: Dict[str, Any] = {}  # Platform-specific data
-
-
-class VideoPlatformConfig(BaseModel):
-    """Configuration for a video platform."""
-
-    api_key: str
-    webhook_secret: str
-    api_url: Optional[str] = None
-    subdomain: Optional[str] = None
-    s3_bucket: Optional[str] = None
-    s3_region: Optional[str] = None
-    aws_role_arn: Optional[str] = None
-    aws_access_key_id: Optional[str] = None
-    aws_access_key_secret: Optional[str] = None
-
-
-class VideoPlatformClient(ABC):
-    """Abstract base class for video platform integrations."""
-
-    PLATFORM_NAME: str = ""
-
-    def __init__(self, config: VideoPlatformConfig):
-        self.config = config
-
-    @abstractmethod
-    async def create_meeting(
-        self, room_name_prefix: str, end_date: datetime, room: Room
-    ) -> MeetingData:
-        """Create a new meeting room."""
-        pass
-
-    @abstractmethod
-    async def get_room_sessions(self, room_name: str) -> Dict[str, Any]:
-        """Get session information for a room."""
-        pass
-
-    @abstractmethod
-    async def delete_room(self, room_name: str) -> bool:
-        """Delete a room. Returns True if successful."""
-        pass
-
-    @abstractmethod
-    async def upload_logo(self, room_name: str, logo_path: str) -> bool:
-        """Upload a logo to the room. Returns True if successful."""
-        pass
-
-    @abstractmethod
-    def verify_webhook_signature(
-        self, body: bytes, signature: str, timestamp: Optional[str] = None
-    ) -> bool:
-        """Verify webhook signature for security."""
-        pass
-
-    def format_recording_config(self, room: Room) -> Dict[str, Any]:
-        """Format recording configuration for the platform.
-        Can be overridden by specific implementations."""
-        if room.recording_type == "cloud" and self.config.s3_bucket:
-            return {
-                "type": room.recording_type,
-                "bucket": self.config.s3_bucket,
-                "region": self.config.s3_region,
-                "trigger": room.recording_trigger,
-            }
-        return {"type": room.recording_type}
--- a/server/reflector/video_platforms/factory.py
+++ b/server/reflector/video_platforms/factory.py
@@ -1,54 +0,0 @@
-"""Factory for creating video platform clients based on configuration."""
-
-from typing import TYPE_CHECKING, Literal, Optional, overload
-
-from reflector.db.rooms import VideoPlatform
-from reflector.settings import settings
-
-from .base import VideoPlatformClient, VideoPlatformConfig
-from .registry import get_platform_client
-
-if TYPE_CHECKING:
-    from .jitsi import JitsiClient
-    from .whereby import WherebyClient
-
-
-def get_platform_config(platform: str) -> VideoPlatformConfig:
-    """Get configuration for a specific platform."""
-    if platform == VideoPlatform.WHEREBY:
-        return VideoPlatformConfig(
-            api_key=settings.WHEREBY_API_KEY or "",
-            webhook_secret=settings.WHEREBY_WEBHOOK_SECRET or "",
-            api_url=settings.WHEREBY_API_URL,
-            s3_bucket=settings.RECORDING_STORAGE_AWS_BUCKET_NAME,
-            aws_access_key_id=settings.AWS_WHEREBY_ACCESS_KEY_ID,
-            aws_access_key_secret=settings.AWS_WHEREBY_ACCESS_KEY_SECRET,
-        )
-    elif platform == VideoPlatform.JITSI:
-        return VideoPlatformConfig(
-            api_key="",  # Jitsi uses JWT, no API key
-            webhook_secret=settings.JITSI_WEBHOOK_SECRET or "",
-            api_url=f"https://{settings.JITSI_DOMAIN}",
-        )
-    else:
-        raise ValueError(f"Unknown platform: {platform}")
-
-
-@overload
-def create_platform_client(platform: Literal["jitsi"]) -> "JitsiClient": ...
-
-
-@overload
-def create_platform_client(platform: Literal["whereby"]) -> "WherebyClient": ...
-
-
-def create_platform_client(platform: str) -> VideoPlatformClient:
-    """Create a video platform client instance."""
-    config = get_platform_config(platform)
-    return get_platform_client(platform, config)
-
-
-def get_platform_for_room(room_id: Optional[str] = None) -> str:
-    """Determine which platform to use for a room based on feature flags."""
-    # For now, default to whereby since we don't have feature flags yet
-    return VideoPlatform.WHEREBY
--- a/server/reflector/video_platforms/jitsi/init.py
+++ b/server/reflector/video_platforms/jitsi/init.py
@@ -1,4 +0,0 @@
-from .client import JitsiClient, JitsiMeetingData
-from .router import router
-
-__all__ = ["JitsiClient", "JitsiMeetingData", "router"]
--- a/server/reflector/video_platforms/jitsi/client.py
+++ b/server/reflector/video_platforms/jitsi/client.py
@@ -1,111 +0,0 @@
-import hmac
-from datetime import datetime, timezone
-from hashlib import sha256
-from typing import Any, Dict, Optional
-
-import jwt
-
-from reflector.db.rooms import Room, VideoPlatform
-from reflector.settings import settings
-from reflector.utils import generate_uuid4
-
-from ..base import MeetingData, VideoPlatformClient
-
-
-class JitsiMeetingData(MeetingData):
-    @property
-    def user_jwt(self) -> str:
-        return self.extra_data.get("user_jwt", "")
-
-    @property
-    def host_jwt(self) -> str:
-        return self.extra_data.get("host_jwt", "")
-
-    @property
-    def domain(self) -> str:
-        return self.extra_data.get("domain", "")
-
-
-class JitsiClient(VideoPlatformClient):
-    PLATFORM_NAME = VideoPlatform.JITSI
-
-    def _generate_jwt(self, room: str, moderator: bool, exp: datetime) -> str:
-        if not settings.JITSI_JWT_SECRET:
-            raise ValueError("JITSI_JWT_SECRET is required for JWT generation")
-
-        payload = {
-            "aud": settings.JITSI_JWT_AUDIENCE,
-            "iss": settings.JITSI_JWT_ISSUER,
-            "sub": settings.JITSI_DOMAIN,
-            "room": room,
-            "exp": int(exp.timestamp()),
-            "context": {
-                "user": {
-                    "name": "Reflector User",
-                    "moderator": moderator,
-                },
-                "features": {
-                    "recording": True,
-                    "livestreaming": False,
-                },
-            },
-        }
-
-        return jwt.encode(payload, settings.JITSI_JWT_SECRET, algorithm="HS256")
-
-    async def create_meeting(
-        self, room_name_prefix: str, end_date: datetime, room: Room
-    ) -> JitsiMeetingData:
-        jitsi_room = f"reflector-{room.name}-{generate_uuid4()}"
-
-        user_jwt = self._generate_jwt(room=jitsi_room, moderator=False, exp=end_date)
-        host_jwt = self._generate_jwt(room=jitsi_room, moderator=True, exp=end_date)
-
-        room_url = f"https://{settings.JITSI_DOMAIN}/{jitsi_room}?jwt={user_jwt}"
-        host_room_url = f"https://{settings.JITSI_DOMAIN}/{jitsi_room}?jwt={host_jwt}"
-
-        return JitsiMeetingData(
-            meeting_id=generate_uuid4(),
-            room_name=jitsi_room,
-            room_url=room_url,
-            host_room_url=host_room_url,
-            platform=self.PLATFORM_NAME,
-            extra_data={
-                "user_jwt": user_jwt,
-                "host_jwt": host_jwt,
-                "domain": settings.JITSI_DOMAIN,
-            },
-        )
-
-    async def get_room_sessions(self, room_name: str) -> Dict[str, Any]:
-        return {
-            "roomName": room_name,
-            "sessions": [
-                {
-                    "sessionId": generate_uuid4(),
-                    "startTime": datetime.now(tz=timezone.utc).isoformat(),
-                    "participants": [],
-                    "isActive": True,
-                }
-            ],
-        }
-
-    async def delete_room(self, room_name: str) -> bool:
-        return True
-
-    async def upload_logo(self, room_name: str, logo_path: str) -> bool:
-        return True
-
-    def verify_webhook_signature(
-        self, body: bytes, signature: str, timestamp: Optional[str] = None
-    ) -> bool:
-        if not signature or not self.config.webhook_secret:
-            return False
-
-        try:
-            expected = hmac.new(
-                self.config.webhook_secret.encode(), body, sha256
-            ).hexdigest()
-            return hmac.compare_digest(expected, signature)
-        except Exception:
-            return False
--- a/server/reflector/video_platforms/jitsi/router.py
+++ b/server/reflector/video_platforms/jitsi/router.py
@@ -1,165 +0,0 @@
-import hmac
-from datetime import datetime
-from hashlib import sha256
-from typing import Any, Dict
-
-from fastapi import APIRouter, HTTPException, Request
-from pydantic import BaseModel
-
-from reflector.db.meetings import meetings_controller
-from reflector.settings import settings
-
-try:
-    from reflector.video_platforms import create_platform_client
-except ImportError:
-    # PyJWT not yet installed, will be added in final task
-    def create_platform_client(platform: str):
-        return None
-
-
-router = APIRouter()
-
-
-class JitsiWebhookEvent(BaseModel):
-    event: str
-    room: str
-    timestamp: datetime
-    data: Dict[str, Any] = {}
-
-
-class JibriRecordingEvent(BaseModel):
-    room_name: str
-    recording_file: str
-    recording_status: str
-    timestamp: datetime
-
-
-def verify_jitsi_webhook_signature(body: bytes, signature: str) -> bool:
-    """Verify Jitsi webhook signature using HMAC-SHA256."""
-    if not signature or not settings.JITSI_WEBHOOK_SECRET:
-        return False
-
-    try:
-        client = create_platform_client("jitsi")
-        if client is None:
-            # Fallback verification when platform client not available
-            expected = hmac.new(
-                settings.JITSI_WEBHOOK_SECRET.encode(), body, sha256
-            ).hexdigest()
-            return hmac.compare_digest(expected, signature)
-        return client.verify_webhook_signature(body, signature)
-    except Exception:
-        return False
-
-
-@router.post("/jitsi/events")
-async def jitsi_events_webhook(event: JitsiWebhookEvent, request: Request):
-    """
-    Handle Prosody event-sync webhooks from Jitsi Meet.
-
-    Expected event types:
-    - muc-occupant-joined: participant joined the room
-    - muc-occupant-left: participant left the room
-    - jibri-recording-on: recording started
-    - jibri-recording-off: recording stopped
-    """
-    # Verify webhook signature
-    body = await request.body()
-    signature = request.headers.get("x-jitsi-signature", "")
-
-    if not verify_jitsi_webhook_signature(body, signature):
-        raise HTTPException(status_code=401, detail="Invalid webhook signature")
-
-    # Find meeting by room name
-    meeting = await meetings_controller.get_by_room_name(event.room)
-    if not meeting:
-        raise HTTPException(status_code=404, detail="Meeting not found")
-
-    # Handle participant events
-    if event.event == "muc-occupant-joined":
-        # Store event and update participant count
-        await meetings_controller.participant_joined(
-            meeting.id, {"timestamp": event.timestamp, "data": event.data}
-        )
-        current_count = getattr(meeting, "num_clients", 0)
-        await meetings_controller.update_meeting(
-            meeting.id, num_clients=current_count + 1
-        )
-    elif event.event == "muc-occupant-left":
-        # Store event and update participant count
-        await meetings_controller.participant_left(
-            meeting.id, {"timestamp": event.timestamp, "data": event.data}
-        )
-        current_count = getattr(meeting, "num_clients", 0)
-        await meetings_controller.update_meeting(
-            meeting.id, num_clients=max(0, current_count - 1)
-        )
-    elif event.event == "jibri-recording-on":
-        # Store recording started event
-        await meetings_controller.recording_started(
-            meeting.id, {"timestamp": event.timestamp, "data": event.data}
-        )
-    elif event.event == "jibri-recording-off":
-        # Store recording stopped event
-        await meetings_controller.recording_stopped(
-            meeting.id, {"timestamp": event.timestamp, "data": event.data}
-        )
-
-    return {"status": "ok", "event": event.event, "room": event.room}
-
-
-@router.post("/jibri/recording-complete")
-async def jibri_recording_complete(event: JibriRecordingEvent, request: Request):
-    """
-    Handle Jibri recording completion webhook.
-
-    This endpoint is called by the Jibri finalize script when a recording
-    is completed and uploaded to storage.
-    """
-    # Verify webhook signature
-    body = await request.body()
-    signature = request.headers.get("x-jitsi-signature", "")
-
-    if not verify_jitsi_webhook_signature(body, signature):
-        raise HTTPException(status_code=401, detail="Invalid webhook signature")
-
-    # Find meeting by room name
-    meeting = await meetings_controller.get_by_room_name(event.room_name)
-    if not meeting:
-        raise HTTPException(status_code=404, detail="Meeting not found")
-
-    # Store recording completion event
-    await meetings_controller.add_event(
-        meeting.id,
-        "recording_completed",
-        {
-            "recording_file": event.recording_file,
-            "recording_status": event.recording_status,
-            "timestamp": event.timestamp,
-        },
-    )
-
-    # TODO: Trigger recording processing pipeline
-    # This is where we would:
-    # 1. Download the recording file from Jibri storage
-    # 2. Create a transcript record in the database
-    # 3. Queue the audio processing tasks (chunking, transcription, etc.)
-    # 4. Update meeting status to indicate recording is being processed
-
-    return {
-        "status": "ok",
-        "room_name": event.room_name,
-        "recording_file": event.recording_file,
-        "message": "Recording processing queued",
-    }
-
-
-@router.get("/jitsi/health")
-async def jitsi_health_check():
-    """Simple health check endpoint for Jitsi webhook configuration."""
-    return {
-        "status": "ok",
-        "service": "jitsi-webhooks",
-        "timestamp": datetime.now(tz=timezone.utc).isoformat(),
-        "webhook_secret_configured": bool(settings.JITSI_WEBHOOK_SECRET),
-    }
--- a/server/reflector/video_platforms/jitsi/tasks.py
+++ b/server/reflector/video_platforms/jitsi/tasks.py
@@ -1,3 +0,0 @@
-"""Jitsi-specific worker tasks."""
-
-# Placeholder for Jitsi recording tasks
--- a/server/reflector/video_platforms/registry.py
+++ b/server/reflector/video_platforms/registry.py
@@ -1,56 +0,0 @@
-from typing import TYPE_CHECKING, Dict, Literal, Type, overload
-
-from .base import VideoPlatformClient, VideoPlatformConfig
-
-if TYPE_CHECKING:
-    from .jitsi import JitsiClient
-    from .whereby import WherebyClient
-
-# Registry of available video platforms
-_PLATFORMS: Dict[str, Type[VideoPlatformClient]] = {}
-
-
-def register_platform(name: str, client_class: Type[VideoPlatformClient]):
-    """Register a video platform implementation."""
-    _PLATFORMS[name.lower()] = client_class
-
-
-@overload
-def get_platform_client(
-    platform: Literal["jitsi"], config: VideoPlatformConfig
-) -> "JitsiClient": ...
-
-
-@overload
-def get_platform_client(
-    platform: Literal["whereby"], config: VideoPlatformConfig
-) -> "WherebyClient": ...
-
-
-def get_platform_client(
-    platform: str, config: VideoPlatformConfig
-) -> VideoPlatformClient:
-    """Get a video platform client instance."""
-    platform_lower = platform.lower()
-    if platform_lower not in _PLATFORMS:
-        raise ValueError(f"Unknown video platform: {platform}")
-
-    client_class = _PLATFORMS[platform_lower]
-    return client_class(config)
-
-
-def get_available_platforms() -> list[str]:
-    """Get list of available platform names."""
-    return list(_PLATFORMS.keys())
-
-
-# Auto-register built-in platforms
-def _register_builtin_platforms():
-    from .jitsi import JitsiClient
-    from .whereby import WherebyClient
-
-    register_platform("jitsi", JitsiClient)
-    register_platform("whereby", WherebyClient)
-
-
-_register_builtin_platforms()
--- a/server/reflector/video_platforms/whereby/init.py
+++ b/server/reflector/video_platforms/whereby/init.py
@@ -1,6 +0,0 @@
-"""Whereby video platform integration."""
-
-from .client import WherebyClient
-from .router import router
-
-__all__ = ["WherebyClient", "router"]
--- a/server/reflector/video_platforms/whereby/client.py
+++ b/server/reflector/video_platforms/whereby/client.py
@@ -1,113 +0,0 @@
-import hmac
-from datetime import datetime
-from hashlib import sha256
-from typing import Any, Dict, Optional
-
-import httpx
-
-from reflector.db.rooms import Room, VideoPlatform
-from reflector.settings import settings
-
-from ..base import MeetingData, VideoPlatformClient
-
-
-class WherebyClient(VideoPlatformClient):
-    PLATFORM_NAME = VideoPlatform.WHEREBY
-
-    def __init__(self, config):
-        super().__init__(config)
-        self.headers = {
-            "Content-Type": "application/json; charset=utf-8",
-            "Authorization": f"Bearer {self.config.api_key}",
-        }
-        self.timeout = 10
-
-    async def create_meeting(
-        self, room_name_prefix: str, end_date: datetime, room: Room
-    ) -> MeetingData:
-        data = {
-            "isLocked": room.is_locked,
-            "roomNamePrefix": room_name_prefix,
-            "roomNamePattern": "uuid",
-            "roomMode": room.room_mode,
-            "endDate": end_date.isoformat(),
-            "recording": {
-                "type": room.recording_type,
-                "destination": {
-                    "provider": "s3",
-                    "bucket": settings.RECORDING_STORAGE_AWS_BUCKET_NAME,
-                    "accessKeyId": self.config.aws_access_key_id,
-                    "accessKeySecret": self.config.aws_access_key_secret,
-                    "fileFormat": "mp4",
-                },
-                "startTrigger": room.recording_trigger,
-            },
-            "fields": ["hostRoomUrl"],
-        }
-
-        async with httpx.AsyncClient() as client:
-            response = await client.post(
-                f"{self.config.api_url}/meetings",
-                headers=self.headers,
-                json=data,
-                timeout=self.timeout,
-            )
-            response.raise_for_status()
-            meeting_data = response.json()
-
-        return MeetingData(
-            meeting_id=meeting_data["meetingId"],
-            room_name=meeting_data["roomName"],
-            room_url=meeting_data["roomUrl"],
-            host_room_url=meeting_data["hostRoomUrl"],
-            platform=self.PLATFORM_NAME,
-            extra_data={
-                "startDate": meeting_data["startDate"],
-                "endDate": meeting_data["endDate"],
-                "recording": meeting_data.get("recording", {}),
-            },
-        )
-
-    async def get_room_sessions(self, room_name: str) -> Dict[str, Any]:
-        async with httpx.AsyncClient() as client:
-            response = await client.get(
-                f"{self.config.api_url}/insights/room-sessions?roomName={room_name}",
-                headers=self.headers,
-                timeout=self.timeout,
-            )
-            response.raise_for_status()
-            return response.json()
-
-    async def delete_room(self, room_name: str) -> bool:
-        return True
-
-    async def upload_logo(self, room_name: str, logo_path: str) -> bool:
-        try:
-            async with httpx.AsyncClient() as client:
-                with open(logo_path, "rb") as f:
-                    response = await client.put(
-                        f"{self.config.api_url}/rooms{room_name}/theme/logo",
-                        headers={
-                            "Authorization": f"Bearer {self.config.api_key}",
-                        },
-                        timeout=self.timeout,
-                        files={"image": f},
-                    )
-                    response.raise_for_status()
-            return True
-        except Exception:
-            return False
-
-    def verify_webhook_signature(
-        self, body: bytes, signature: str, timestamp: Optional[str] = None
-    ) -> bool:
-        if not signature or not self.config.webhook_secret:
-            return False
-
-        try:
-            expected = hmac.new(
-                self.config.webhook_secret.encode(), body, sha256
-            ).hexdigest()
-            return hmac.compare_digest(expected, signature)
-        except Exception:
-            return False
--- a/server/reflector/video_platforms/whereby/tasks.py
+++ b/server/reflector/video_platforms/whereby/tasks.py
@@ -1,4 +0,0 @@
-"""Whereby-specific worker tasks."""
-
-# Placeholder for Whereby-specific background tasks
-# This can be extended with Whereby-specific processing tasks in the future
--- a/server/reflector/views/jibri_webhook.py
+++ b/server/reflector/views/jibri_webhook.py
@@ -1,126 +0,0 @@
-from pathlib import Path
-from typing import Annotated, Any, Dict, Optional
-
-import structlog
-from fastapi import APIRouter, Depends, HTTPException
-from pydantic import BaseModel
-
-import reflector.auth as auth
-from reflector.db.transcripts import SourceKind, transcripts_controller
-from reflector.jibri_events import JitsiEventParser
-from reflector.pipelines.main_file_pipeline import task_pipeline_file_process
-from reflector.settings import settings
-
-logger = structlog.get_logger(__name__)
-
-router = APIRouter(prefix="/api/v1/jibri", tags=["jibri"])
-
-
-class RecordingReadyRequest(BaseModel):
-    session_id: str
-    path: str  # Relative path from recordings directory
-    meeting_url: str
-
-
-@router.post("/recording-ready")
-async def handle_recording_ready(
-    request: RecordingReadyRequest,
-    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
-) -> Dict[str, Any]:
-    user_id = user["sub"] if user else None
-
-    recordings_base = Path(settings.JIBRI_RECORDINGS_PATH or "/recordings")
-    recording_path = recordings_base / request.path
-
-    if not recording_path.exists():
-        raise HTTPException(
-            status_code=404, detail=f"Recording path not found: {request.path}"
-        )
-
-    recording_file = recording_path / "recording.mp4"
-    events_file = recording_path / "events.jsonl"
-
-    if not recording_file.exists():
-        raise HTTPException(status_code=404, detail="Recording file not found")
-
-    # Parse events if available
-    metadata = {}
-    participant_count = 0
-
-    if events_file.exists():
-        parser = JitsiEventParser()
-        metadata = parser.parse_events_file(str(recording_path))
-        participant_count = len(metadata.get("participants", []))
-        logger.info(
-            "Parsed Jibri events",
-            session_id=request.session_id,
-            event_count=metadata.get("event_count", 0),
-            participant_count=participant_count,
-        )
-    else:
-        logger.warning("No events file found", session_id=request.session_id)
-        metadata = {
-            "room": {"meeting_url": request.meeting_url, "name": request.session_id},
-            "participants": [],
-            "speaker_stats": {},
-            "event_count": 0,
-        }
-
-    # Create transcript using controller
-    title = f"Meeting: {metadata.get('room', {}).get('name', request.session_id)}"
-    transcript = await transcripts_controller.add(
-        name=title,
-        source_kind=SourceKind.FILE,
-        source_language="en",
-        target_language="en",
-        user_id=user_id,
-    )
-
-    # Store Jitsi data in appropriate fields
-    update_data = {}
-
-    # Store participants if available
-    if metadata.get("participants"):
-        update_data["participants"] = metadata["participants"]
-
-    # Store events data (room info, speaker stats, etc.)
-    update_data["events"] = {
-        "jitsi_metadata": metadata,
-        "session_id": request.session_id,
-        "recording_path": str(recording_path),
-        "meeting_url": request.meeting_url,
-    }
-
-    if update_data:
-        await transcripts_controller.update(transcript, update_data)
-
-    # Copy recording file to transcript data path
-    # The pipeline expects the file to be in the transcript's data path
-    upload_file = transcript.data_path / "upload.mp4"
-    upload_file.parent.mkdir(parents=True, exist_ok=True)
-
-    # Create symlink or copy the file
-    import shutil
-
-    shutil.copy2(recording_file, upload_file)
-
-    # Update status to uploaded
-    await transcripts_controller.update(transcript, {"status": "uploaded"})
-
-    # Trigger processing pipeline
-    task_pipeline_file_process.delay(transcript_id=transcript.id)
-
-    logger.info(
-        "Jibri recording ready for processing",
-        transcript_id=transcript.id,
-        session_id=request.session_id,
-        participant_count=participant_count,
-    )
-
-    return {
-        "status": "accepted",
-        "transcript_id": transcript.id,
-        "session_id": request.session_id,
-        "events_found": events_file.exists(),
-        "participant_count": participant_count,
-    }
--- a/server/reflector/views/rooms.py
+++ b/server/reflector/views/rooms.py
@@ -12,12 +12,9 @@ from pydantic import BaseModel
 import reflector.auth as auth
 from reflector.db import get_database
 from reflector.db.meetings import meetings_controller
-from reflector.db.rooms import VideoPlatform, rooms_controller
+from reflector.db.rooms import rooms_controller
 from reflector.settings import settings
-from reflector.video_platforms.factory import (
-    create_platform_client,
-)
-from reflector.worker.webhook import test_webhook
+from reflector.whereby import create_meeting, upload_logo

 logger = logging.getLogger(__name__)

@@ -25,6 +22,7 @@ router = APIRouter()


 def parse_datetime_with_timezone(iso_string: str) -> datetime:
+    """Parse ISO datetime string and ensure timezone awareness (defaults to UTC if naive)."""
    dt = datetime.fromisoformat(iso_string)
    if dt.tzinfo is None:
        dt = dt.replace(tzinfo=timezone.utc)
@@ -44,12 +42,11 @@ class Room(BaseModel):
    recording_type: str
    recording_trigger: str
    is_shared: bool
-    platform: VideoPlatform = VideoPlatform.WHEREBY
-
-
-class RoomDetails(Room):
-    webhook_url: str | None
-    webhook_secret: str | None
+    ics_url: Optional[str] = None
+    ics_fetch_interval: int = 300
+    ics_enabled: bool = False
+    ics_last_sync: Optional[datetime] = None
+    ics_last_etag: Optional[str] = None


 class Meeting(BaseModel):
@@ -72,42 +69,34 @@ class CreateRoom(BaseModel):
    recording_type: str
    recording_trigger: str
    is_shared: bool
-    webhook_url: str
-    webhook_secret: str
-    platform: VideoPlatform
+    ics_url: Optional[str] = None
+    ics_fetch_interval: int = 300
+    ics_enabled: bool = False


 class UpdateRoom(BaseModel):
-    name: str
-    zulip_auto_post: bool
-    zulip_stream: str
-    zulip_topic: str
-    is_locked: bool
-    room_mode: str
-    recording_type: str
-    recording_trigger: str
-    is_shared: bool
-    webhook_url: str
-    webhook_secret: str
-    platform: VideoPlatform
+    name: Optional[str] = None
+    zulip_auto_post: Optional[bool] = None
+    zulip_stream: Optional[str] = None
+    zulip_topic: Optional[str] = None
+    is_locked: Optional[bool] = None
+    room_mode: Optional[str] = None
+    recording_type: Optional[str] = None
+    recording_trigger: Optional[str] = None
+    is_shared: Optional[bool] = None
+    ics_url: Optional[str] = None
+    ics_fetch_interval: Optional[int] = None
+    ics_enabled: Optional[bool] = None


 class DeletionStatus(BaseModel):
    status: str


-class WebhookTestResult(BaseModel):
-    success: bool
-    message: str = ""
-    error: str = ""
-    status_code: int | None = None
-    response_preview: str | None = None
-
-
-@router.get("/rooms", response_model=Page[RoomDetails])
+@router.get("/rooms", response_model=Page[Room])
 async def rooms_list(
    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
-) -> list[RoomDetails]:
+) -> list[Room]:
    if not user and not settings.PUBLIC_MODE:
        raise HTTPException(status_code=401, detail="Not authenticated")

@@ -121,18 +110,6 @@ async def rooms_list(
    )


-@router.get("/rooms/{room_id}", response_model=RoomDetails)
-async def rooms_get(
-    room_id: str,
-    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
-):
-    user_id = user["sub"] if user else None
-    room = await rooms_controller.get_by_id_for_http(room_id, user_id=user_id)
-    if not room:
-        raise HTTPException(status_code=404, detail="Room not found")
-    return room
-
-
@router.post("/rooms", response_model=Room)
 async def rooms_create(
    room: CreateRoom,
@@ -151,13 +128,13 @@ async def rooms_create(
        recording_type=room.recording_type,
        recording_trigger=room.recording_trigger,
        is_shared=room.is_shared,
-        webhook_url=room.webhook_url,
-        webhook_secret=room.webhook_secret,
-        platform=room.platform,
+        ics_url=room.ics_url,
+        ics_fetch_interval=room.ics_fetch_interval,
+        ics_enabled=room.ics_enabled,
    )


-@router.patch("/rooms/{room_id}", response_model=RoomDetails)
+@router.patch("/rooms/{room_id}", response_model=Room)
 async def rooms_update(
    room_id: str,
    info: UpdateRoom,
@@ -201,49 +178,42 @@ async def rooms_create_meeting(
    if meeting is None:
        end_date = current_time + timedelta(hours=8)

-        platform = room.platform
-        client = create_platform_client(platform)
+        whereby_meeting = await create_meeting("", end_date=end_date, room=room)
+        await upload_logo(whereby_meeting["roomName"], "./images/logo.png")

-        platform_meeting = await client.create_meeting("", end_date=end_date, room=room)
-        await client.upload_logo(platform_meeting.room_name, "./images/logo.png")
-
-        meeting_data = {
-            "meeting_id": platform_meeting.meeting_id,
-            "room_name": platform_meeting.room_name,
-            "room_url": platform_meeting.room_url,
-            "host_room_url": platform_meeting.host_room_url,
-            "start_date": current_time,
-            "end_date": end_date,
-        }
+        # Now try to save to database
        try:
            meeting = await meetings_controller.create(
-                id=meeting_data["meeting_id"],
-                room_name=meeting_data["room_name"],
-                room_url=meeting_data["room_url"],
-                host_room_url=meeting_data["host_room_url"],
-                start_date=meeting_data["start_date"],
-                end_date=meeting_data["end_date"],
+                id=whereby_meeting["meetingId"],
+                room_name=whereby_meeting["roomName"],
+                room_url=whereby_meeting["roomUrl"],
+                host_room_url=whereby_meeting["hostRoomUrl"],
+                start_date=parse_datetime_with_timezone(whereby_meeting["startDate"]),
+                end_date=parse_datetime_with_timezone(whereby_meeting["endDate"]),
                user_id=user_id,
                room=room,
            )
        except (asyncpg.exceptions.UniqueViolationError, sqlite3.IntegrityError):
+            # Another request already created a meeting for this room
+            # Log this race condition occurrence
            logger.info(
                "Race condition detected for room %s - fetching existing meeting",
                room.name,
            )
            logger.warning(
-                "Platform meeting %s was created but not used (resource leak) for room %s",
-                meeting_data["meeting_id"],
+                "Whereby meeting %s was created but not used (resource leak) for room %s",
+                whereby_meeting["meetingId"],
                room.name,
            )
+
+            # Fetch the meeting that was created by the other request
            meeting = await meetings_controller.get_active(
                room=room, current_time=current_time
            )
            if meeting is None:
+                # Edge case: meeting was created but expired/deleted between checks
                logger.error(
-                    "Meeting disappeared after race condition for room %s",
-                    room.name,
-                    exc_info=True,
+                    "Meeting disappeared after race condition for room %s", room.name
                )
                raise HTTPException(
                    status_code=503, detail="Unable to join meeting - please try again"
@@ -255,21 +225,215 @@ async def rooms_create_meeting(
    return meeting


-@router.post("/rooms/{room_id}/webhook/test", response_model=WebhookTestResult)
-async def rooms_test_webhook(
-    room_id: str,
+class ICSStatus(BaseModel):
+    status: str
+    last_sync: Optional[datetime] = None
+    next_sync: Optional[datetime] = None
+    last_etag: Optional[str] = None
+    events_count: int = 0
+
+
+class ICSSyncResult(BaseModel):
+    status: str
+    hash: Optional[str] = None
+    events_found: int = 0
+    events_created: int = 0
+    events_updated: int = 0
+    events_deleted: int = 0
+    error: Optional[str] = None
+
+
+@router.post("/rooms/{room_name}/ics/sync", response_model=ICSSyncResult)
+async def rooms_sync_ics(
+    room_name: str,
    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
 ):
    user_id = user["sub"] if user else None
+    room = await rooms_controller.get_by_name(room_name)

-    room = await rooms_controller.get_by_id(room_id)
    if not room:
        raise HTTPException(status_code=404, detail="Room not found")

-    if user_id and room.user_id != user_id:
+    if user_id != room.user_id:
        raise HTTPException(
-            status_code=403, detail="Not authorized to test this room's webhook"
+            status_code=403, detail="Only room owner can trigger ICS sync"
        )

-    result = await test_webhook(room_id)
-    return WebhookTestResult(**result)
+    if not room.ics_enabled or not room.ics_url:
+        raise HTTPException(status_code=400, detail="ICS not configured for this room")
+
+    from reflector.services.ics_sync import ics_sync_service
+
+    result = await ics_sync_service.sync_room_calendar(room)
+
+    if result["status"] == "error":
+        raise HTTPException(
+            status_code=500, detail=result.get("error", "Unknown error")
+        )
+
+    return ICSSyncResult(**result)
+
+
+@router.get("/rooms/{room_name}/ics/status", response_model=ICSStatus)
+async def rooms_ics_status(
+    room_name: str,
+    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
+):
+    user_id = user["sub"] if user else None
+    room = await rooms_controller.get_by_name(room_name)
+
+    if not room:
+        raise HTTPException(status_code=404, detail="Room not found")
+
+    if user_id != room.user_id:
+        raise HTTPException(
+            status_code=403, detail="Only room owner can view ICS status"
+        )
+
+    next_sync = None
+    if room.ics_enabled and room.ics_last_sync:
+        next_sync = room.ics_last_sync + timedelta(seconds=room.ics_fetch_interval)
+
+    from reflector.db.calendar_events import calendar_events_controller
+
+    events = await calendar_events_controller.get_by_room(
+        room.id, include_deleted=False
+    )
+
+    return ICSStatus(
+        status="enabled" if room.ics_enabled else "disabled",
+        last_sync=room.ics_last_sync,
+        next_sync=next_sync,
+        last_etag=room.ics_last_etag,
+        events_count=len(events),
+    )
+
+
+class CalendarEventResponse(BaseModel):
+    id: str
+    room_id: str
+    ics_uid: str
+    title: Optional[str] = None
+    description: Optional[str] = None
+    start_time: datetime
+    end_time: datetime
+    attendees: Optional[list[dict]] = None
+    location: Optional[str] = None
+    last_synced: datetime
+    created_at: datetime
+    updated_at: datetime
+
+
+@router.get("/rooms/{room_name}/meetings", response_model=list[CalendarEventResponse])
+async def rooms_list_meetings(
+    room_name: str,
+    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
+):
+    user_id = user["sub"] if user else None
+    room = await rooms_controller.get_by_name(room_name)
+
+    if not room:
+        raise HTTPException(status_code=404, detail="Room not found")
+
+    from reflector.db.calendar_events import calendar_events_controller
+
+    events = await calendar_events_controller.get_by_room(
+        room.id, include_deleted=False
+    )
+
+    if user_id != room.user_id:
+        for event in events:
+            event.description = None
+            event.attendees = None
+
+    return events
+
+
+@router.get(
+    "/rooms/{room_name}/meetings/upcoming", response_model=list[CalendarEventResponse]
+)
+async def rooms_list_upcoming_meetings(
+    room_name: str,
+    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
+    minutes_ahead: int = 30,
+):
+    user_id = user["sub"] if user else None
+    room = await rooms_controller.get_by_name(room_name)
+
+    if not room:
+        raise HTTPException(status_code=404, detail="Room not found")
+
+    from reflector.db.calendar_events import calendar_events_controller
+
+    events = await calendar_events_controller.get_upcoming(
+        room.id, minutes_ahead=minutes_ahead
+    )
+
+    if user_id != room.user_id:
+        for event in events:
+            event.description = None
+            event.attendees = None
+
+    return events
+
+
+@router.get("/rooms/{room_name}/meetings/active", response_model=list[Meeting])
+async def rooms_list_active_meetings(
+    room_name: str,
+    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
+):
+    """List all active meetings for a room (supports multiple active meetings)"""
+    user_id = user["sub"] if user else None
+    room = await rooms_controller.get_by_name(room_name)
+
+    if not room:
+        raise HTTPException(status_code=404, detail="Room not found")
+
+    current_time = datetime.now(timezone.utc)
+    meetings = await meetings_controller.get_all_active_for_room(
+        room=room, current_time=current_time
+    )
+
+    # Hide host URLs from non-owners
+    if user_id != room.user_id:
+        for meeting in meetings:
+            meeting.host_room_url = ""
+
+    return meetings
+
+
+@router.post("/rooms/{room_name}/meetings/{meeting_id}/join", response_model=Meeting)
+async def rooms_join_meeting(
+    room_name: str,
+    meeting_id: str,
+    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
+):
+    """Join a specific meeting by ID"""
+    user_id = user["sub"] if user else None
+    room = await rooms_controller.get_by_name(room_name)
+
+    if not room:
+        raise HTTPException(status_code=404, detail="Room not found")
+
+    meeting = await meetings_controller.get_by_id(meeting_id)
+
+    if not meeting:
+        raise HTTPException(status_code=404, detail="Meeting not found")
+
+    if meeting.room_id != room.id:
+        raise HTTPException(
+            status_code=403, detail="Meeting does not belong to this room"
+        )
+
+    if not meeting.is_active:
+        raise HTTPException(status_code=400, detail="Meeting is not active")
+
+    current_time = datetime.now(timezone.utc)
+    if meeting.end_date <= current_time:
+        raise HTTPException(status_code=400, detail="Meeting has ended")
+
+    # Hide host URL from non-owners
+    if user_id != room.user_id:
+        meeting.host_room_url = ""
+
+    return meeting
--- a/server/reflector/views/transcripts.py
+++ b/server/reflector/views/transcripts.py
@@ -5,7 +5,7 @@ from fastapi import APIRouter, Depends, HTTPException, Query
 from fastapi_pagination import Page
 from fastapi_pagination.ext.databases import apaginate
 from jose import jwt
-from pydantic import BaseModel, Field, constr, field_serializer
+from pydantic import BaseModel, Field, field_serializer

 import reflector.auth as auth
 from reflector.db import get_database
@@ -19,15 +19,14 @@ from reflector.db.search import (
    SearchOffsetBase,
    SearchParameters,
    SearchQuery,
+    SearchQueryBase,
    SearchResult,
    SearchTotal,
    search_controller,
-    search_query_adapter,
 )
 from reflector.db.transcripts import (
    SourceKind,
    TranscriptParticipant,
-    TranscriptStatus,
    TranscriptTopic,
    transcripts_controller,
 )
@@ -64,7 +63,7 @@ class GetTranscriptMinimal(BaseModel):
    id: str
    user_id: str | None
    name: str
-    status: TranscriptStatus
+    status: str
    locked: bool
    duration: float
    title: str | None
@@ -97,7 +96,6 @@ class CreateTranscript(BaseModel):
    name: str
    source_language: str = Field("en")
    target_language: str = Field("en")
-    source_kind: SourceKind | None = None


 class UpdateTranscript(BaseModel):
@@ -116,19 +114,7 @@ class DeletionStatus(BaseModel):
    status: str


-SearchQueryParamBase = constr(min_length=0, strip_whitespace=True)
-SearchQueryParam = Annotated[
-    SearchQueryParamBase, Query(description="Search query text")
-]
-
-
-# http and api standards accept "q="; we would like to handle it as the absence of query, not as "empty string query"
-def parse_search_query_param(q: SearchQueryParam) -> SearchQuery | None:
-    if q == "":
-        return None
-    return search_query_adapter.validate_python(q)
-
-
+SearchQueryParam = Annotated[SearchQueryBase, Query(description="Search query text")]
 SearchLimitParam = Annotated[SearchLimitBase, Query(description="Results per page")]
 SearchOffsetParam = Annotated[
    SearchOffsetBase, Query(description="Number of results to skip")
@@ -138,7 +124,7 @@ SearchOffsetParam = Annotated[
 class SearchResponse(BaseModel):
    results: list[SearchResult]
    total: SearchTotal
-    query: SearchQuery | None = None
+    query: SearchQuery
    limit: SearchLimit
    offset: SearchOffset

@@ -174,7 +160,6 @@ async def transcripts_search(
    limit: SearchLimitParam = DEFAULT_SEARCH_LIMIT,
    offset: SearchOffsetParam = 0,
    room_id: Optional[str] = None,
-    source_kind: Optional[SourceKind] = None,
    user: Annotated[
        Optional[auth.UserInfo], Depends(auth.current_user_optional)
    ] = None,
@@ -188,12 +173,7 @@ async def transcripts_search(
    user_id = user["sub"] if user else None

    search_params = SearchParameters(
-        query_text=parse_search_query_param(q),
-        limit=limit,
-        offset=offset,
-        user_id=user_id,
-        room_id=room_id,
-        source_kind=source_kind,
+        query_text=q, limit=limit, offset=offset, user_id=user_id, room_id=room_id
    )

    results, total = await search_controller.search_transcripts(search_params)
@@ -215,7 +195,7 @@ async def transcripts_create(
    user_id = user["sub"] if user else None
    return await transcripts_controller.add(
        info.name,
-        source_kind=info.source_kind or SourceKind.LIVE,
+        source_kind=SourceKind.LIVE,
        source_language=info.source_language,
        target_language=info.target_language,
        user_id=user_id,
@@ -350,6 +330,8 @@ async def transcript_update(
    transcript = await transcripts_controller.get_by_id_for_http(
        transcript_id, user_id=user_id
    )
+    if not transcript:
+        raise HTTPException(status_code=404, detail="Transcript not found")
    values = info.dict(exclude_unset=True)
    updated_transcript = await transcripts_controller.update(transcript, values)
    return updated_transcript
--- a/server/reflector/views/transcripts_process.py
+++ b/server/reflector/views/transcripts_process.py
@@ -6,7 +6,7 @@ from pydantic import BaseModel

 import reflector.auth as auth
 from reflector.db.transcripts import transcripts_controller
-from reflector.pipelines.main_file_pipeline import task_pipeline_file_process
+from reflector.pipelines.main_live_pipeline import task_pipeline_process

 router = APIRouter()

@@ -34,13 +34,13 @@ async def transcript_process(
        )

    if task_is_scheduled_or_active(
-        "reflector.pipelines.main_file_pipeline.task_pipeline_file_process",
+        "reflector.pipelines.main_live_pipeline.task_pipeline_process",
        transcript_id=transcript_id,
    ):
        return ProcessStatus(status="already running")

    # schedule a background task process the file
-    task_pipeline_file_process.delay(transcript_id=transcript_id)
+    task_pipeline_process.delay(transcript_id=transcript_id)

    return ProcessStatus(status="ok")

--- a/server/reflector/views/transcripts_upload.py
+++ b/server/reflector/views/transcripts_upload.py
@@ -6,7 +6,7 @@ from pydantic import BaseModel

 import reflector.auth as auth
 from reflector.db.transcripts import transcripts_controller
-from reflector.pipelines.main_file_pipeline import task_pipeline_file_process
+from reflector.pipelines.main_live_pipeline import task_pipeline_process

 router = APIRouter()

@@ -92,6 +92,6 @@ async def transcript_record_upload(
    await transcripts_controller.update(transcript, {"status": "uploaded"})

    # launch a background task to process the file
-    task_pipeline_file_process.delay(transcript_id=transcript_id)
+    task_pipeline_process.delay(transcript_id=transcript_id)

    return UploadStatus(status="ok")
--- a/Show More
+++ b/Show More
Author	SHA1	Message	Date
Mathieu Virbel	311d453e41	feat: implement frontend for calendar integration (Phase 3 & 4) ## Frontend Implementation ### Meeting Selection & Management - Created MeetingSelection component for choosing between multiple active meetings - Shows both active meetings and upcoming calendar events (30 min ahead) - Displays meeting metadata with privacy controls (owner-only details) - Supports creation of unscheduled meetings alongside calendar meetings ### Waiting Room - Added waiting page for users joining before scheduled start time - Shows countdown timer until meeting begins - Auto-transitions to meeting when calendar event becomes active - Handles early joining with proper routing ### Meeting Info Panel - Created collapsible info panel showing meeting details - Displays calendar metadata (title, description, attendees) - Shows participant count and duration - Privacy-aware: sensitive info only visible to room owners ### ICS Configuration UI - Integrated ICS settings into room configuration dialog - Test connection functionality with immediate feedback - Manual sync trigger with detailed results - Shows last sync time and ETag for monitoring - Configurable sync intervals (1 min to 1 hour) ### Routing & Navigation - New /room/{roomName} route for meeting selection - Waiting room at /room/{roomName}/wait?eventId={id} - Classic room page at /{roomName} with meeting info - Uses sessionStorage to pass selected meeting between pages ### API Integration - Added new endpoints for active/upcoming meetings - Regenerated TypeScript client with latest OpenAPI spec - Proper error handling and loading states - Auto-refresh every 30 seconds for live updates ### UI/UX Improvements - Color-coded badges for meeting status - Attendee status indicators (accepted/declined/tentative) - Responsive design with Chakra UI components - Clear visual hierarchy between active and upcoming meetings - Smart truncation for long attendee lists This completes the frontend implementation for calendar integration, enabling users to seamlessly join scheduled meetings from their calendar applications.	2025-08-18 19:29:56 -06:00
Mathieu Virbel	f286f0882c	feat: implement Phase 2 - Multiple active meetings per room with grace period This commit adds support for multiple concurrent meetings per room, implementing grace period logic and improved meeting lifecycle management for calendar integration. ## Database Changes - Remove unique constraint preventing multiple active meetings per room - Add last_participant_left_at field to track when meeting becomes empty - Add grace_period_minutes field (default: 15) for configurable grace period ## Meeting Controller Enhancements - Add get_all_active_for_room() to retrieve all active meetings for a room - Add get_active_by_calendar_event() to find meetings by calendar event ID - Maintain backward compatibility with existing get_active() method ## New API Endpoints - GET /rooms/{room_name}/meetings/active - List all active meetings - POST /rooms/{room_name}/meetings/{meeting_id}/join - Join specific meeting ## Meeting Lifecycle Improvements - 15-minute grace period after last participant leaves - Automatic reactivation when participant rejoins during grace period - Force close calendar meetings 30 minutes after scheduled end time - Update process_meetings task to handle multiple active meetings ## Whereby Integration - Clear grace period when participants join via webhook events - Track participant count for grace period management ## Testing - Add comprehensive tests for multiple active meetings - Test grace period behavior and participant rejoin scenarios - Test calendar meeting force closure logic - All 5 new tests passing This enables proper calendar integration with overlapping meetings while preventing accidental meeting closures through the grace period mechanism.	2025-08-18 19:03:41 -06:00
Mathieu Virbel	ffcafb3bf2	feat: add Celery background tasks for ICS sync	2025-08-18 17:22:41 -06:00
Mathieu Virbel	27075d840c	feat: add ICS calendar API endpoints for room configuration and sync	2025-08-18 17:03:23 -06:00
Mathieu Virbel	30b5cd45e3	feat: calendar integration	2025-08-18 16:51:30 -06:00
				`@@ -1 +0,0 @@`
				`b9d891d3424f371642cb032ecfd0e2564470a72c:server/tests/test_transcripts_recording_deletion.py:generic-api-key:15`