Files
PrivyDrop/docs/ARCHITECTURE.md
T
david_bai 10f236dc8d docs: document in-app navigation persistence (no transfer interruption)
- README.md / README.zh-CN.md:
      - Add feature bullet for in-app navigation persistence (same-tab SPA)
      - Note boundaries: refresh/tab close/new tab not covered
  - Frontend architecture (zh/en):
      - Add “State & Connection Lifecycle (In-App Navigation)” section
      - Explain Zustand store singleton and webrtcService singleton across routes
      - Guidance: avoid leaveRoom()/store reset on route-change side effects
  - System architecture (zh/en):
      - Add “Runtime Session Model (Frontend)” summary
  - AI Playbook:
      - flows.zh-CN.md: add UX optimization item for in-app navigation persistence and debugging notes
      - code-map.zh-CN.md: mark webrtcService and fileTransferStore as singletons (cross-route)
2025-11-24 15:06:10 +08:00

85 lines
5.7 KiB
Markdown

# PrivyDrop - System Architecture Overview
This document provides a high-level overview of the PrivyDrop project's overall architecture, helping developers understand how the various technical components work together.
## 1. Core Components
The PrivyDrop system is primarily composed of the following core parts:
1. **Frontend**: A Single Page Application (SPA) built with Next.js. It's the interface users interact with, responsible for handling file selection and UI presentation. It features a modern, layered architecture with a clear separation between the UI (React Components), centralized state management (Zustand), and the core WebRTC business logic, which is encapsulated in a framework-agnostic service layer.
2. **Backend**: A server built with Node.js and Express. It does not handle any file data. Its core responsibilities are:
- **Signaling Service**: Implemented with Socket.IO, it relays signaling messages (like SDP and ICE Candidates) for the "handshake" process before a WebRTC connection is established.
- **Room Management**: Handles the creation, joining, and status checking of rooms.
- **API Service**: Provides auxiliary HTTP endpoints.
3. **Redis**: An in-memory database used by the backend to store temporary data, such as room information and lists of participants, utilizing its TTL feature for automatic cleanup of expired rooms.
4. **TURN/STUN Server (Optional)**: Used to assist WebRTC with NAT traversal, ensuring a higher success rate for P2P connections in complex network environments. STUN is used to discover public IP addresses, while TURN serves as a fallback relay server.
## 2. Data Flow and Interaction Diagram
The following diagram illustrates the main flow for users establishing a connection and transferring files:
```mermaid
graph TD
subgraph "User A's Browser (Frontend)"
ClientA[UI Components] --> HooksA[Hooks Layer]
HooksA --> StoreA[Zustand Store]
HooksA --> ServiceA[WebRTC Service]
ServiceA --> StoreA
end
subgraph "User B's Browser (Frontend)"
ClientB[UI Components] --> HooksB[Hooks Layer]
HooksB --> StoreB[Zustand Store]
HooksB --> ServiceB[WebRTC Service]
ServiceB --> StoreB
end
subgraph "Server Infrastructure"
Nginx[Nginx Reverse Proxy]
Backend[Backend API / Socket.IO]
Redis[Redis Cache]
TURN[TURN/STUN Server]
end
ServiceA -- 1. Create/Join Room (HTTP/Socket) --> Nginx
Nginx --> Backend
Backend -- Read/Write Room Status --> Redis
ServiceB -- 2. Join Same Room (HTTP/Socket) --> Nginx
Backend -- 3. Broadcast user join event --> ServiceA
Backend -- 3. Broadcast user join event --> ServiceB
ServiceA -- 4. Send Signal (Offer/ICE) --> Backend
Backend -- 5. Forward Signal --> ServiceB
ServiceB -- 6. Send Signal (Answer/ICE) --> Backend
Backend -- 7. Forward Signal --> ServiceA
ServiceA <-.-> |8.&nbsp;STUN Check| TURN
ServiceB <-.-> |8.&nbsp;STUN Check| TURN
ServiceA <-..- |9.&nbsp;P2P Direct Data Transfer| ServiceB
ServiceA <-.-> |9.&nbsp;TURN Relayed Data Transfer| TURN
ServiceB <-.-> |9.&nbsp;TURN Relayed Data Transfer| TURN
```
**Flow Description:**
1. **Room Creation/Joining**: User A's frontend service layer requests the backend to create a unique room ID. The backend records this room in Redis.
2. **Sharing & Joining**: User A shares the room ID with User B. User B's frontend service layer uses this ID to request joining the room.
3. **Signaling Exchange**:
- Once there are two or more users in a room, their respective `WebRTC Service` instances begin exchanging signaling messages through the backend's Socket.IO service.
- This process includes exchanging network information (ICE candidates) and session descriptions (SDP offers/answers). The backend server acts merely as a "postman" for these messages.
4. **NAT Traversal**: The browsers use the network information obtained from the signals, along with STUN/TURN servers, to attempt and establish a direct P2P connection.
5. **P2P Connection Established**: Once the connection is successfully established—which in most cases is a direct peer-to-peer link—all file and text data are transferred directly between User A's and User B's browsers via their `WebRTC Service` instances, without passing through any server. In the rare case of highly restrictive networks where a direct connection cannot be established, data will be relayed through a TURN server. Even in this relayed mode, the data remains end-to-end encrypted by WebRTC and is not visible to the TURN server.
## 3. Design Philosophy
- **Privacy First**: Core file data is never uploaded to the server. The server only acts as an "introducer" or "matchmaker."
- **Frontend-Backend Separation**: Responsibilities are clearly separated. The frontend handles all user interaction and the complex logic of WebRTC; the backend provides lightweight, efficient signaling and room management services.
- **Horizontal Scalability**: The backend is stateless (with state managed in Redis), which theoretically allows it to be scaled horizontally by adding more Node.js instances to handle a large volume of concurrent signaling requests.
## 4. Runtime Session Model (Frontend)
- **SPA In-App Navigation Persistence**: The frontend is an SPA (App Router). Within the same browser tab, in-app navigation does not tear down the singleton app state (Zustand) nor the WebRTC connection service (webrtcService). Ongoing transfers continue, and selected/received content remains available.
- **Boundary**: Page refresh, closing the tab, or opening in a new tab are not covered. If changing layout/SSR strategy, avoid cleaning the connection during layout unmount.