WebRTC & WebSocket Test
Test your browser's support for modern real-time communication technologies. Verify WebRTC and WebSocket capabilities instantly.
Verify your browser's support for WebRTC and WebSocket protocols. Essential for video conferencing, real-time chat, collaborative editing, and live streaming applications.
WebRTC & WebSocket Test
Test your browser's support for modern real-time communication technologies including WebRTC and WebSocket protocols.
What is WebRTC?
WebRTC (Web Real-Time Communication) is an open-source technology that enables peer-to-peer audio, video, and data sharing directly between browsers without requiring plugins or third-party software. It's the technology powering modern video conferencing tools, screen sharing, and real-time collaboration applications.
WebRTC Key Features
- Peer-to-Peer: Direct browser-to-browser communication without server intermediation
- Audio & Video: Built-in support for real-time audio/video streaming
- Data Channels: Send arbitrary data with low latency
- Screen Sharing: Share entire screens or specific applications
- No Plugins: Works natively in modern browsers
- Encryption: All data is encrypted by default (DTLS/SRTP)
What is WebSocket?
WebSocket is a protocol providing full-duplex communication channels over a single TCP connection. Unlike traditional HTTP requests, WebSocket maintains a persistent connection between client and server, enabling real-time bidirectional data flow. It's ideal for chat applications, live notifications, collaborative tools, and real-time dashboards.
WebSocket Key Features
- Persistent Connection: Single long-lived connection instead of multiple HTTP requests
- Bidirectional: Both client and server can send messages at any time
- Low Latency: No connection overhead after initial handshake
- Efficient: Minimal protocol overhead compared to HTTP polling
- Real-Time: Instant message delivery without polling delays
- Scalable: Can handle thousands of concurrent connections
WebRTC vs WebSocket: When to Use Which?
Use WebRTC For:
- Video conferencing (Zoom, Google Meet)
- Voice calls (VoIP applications)
- Screen sharing
- Peer-to-peer file transfer
- Live streaming
- Gaming (P2P multiplayer)
- Remote desktop control
Use WebSocket For:
- Chat applications (Slack, Discord)
- Live notifications
- Real-time dashboards
- Collaborative editing (Google Docs)
- Live sports scores/stock tickers
- Multiplayer games (signaling)
- IoT device communication
Real-World Applications
| Application | Technology | Why |
|---|---|---|
| Zoom, Google Meet | WebRTC | Peer-to-peer video/audio streaming |
| Slack, Discord | WebSocket | Real-time text messaging |
| WhatsApp Web | Both | WebSocket for messages, WebRTC for calls |
| Google Docs | WebSocket | Collaborative real-time editing |
| Twitch | Both | WebRTC for streaming, WebSocket for chat |
| Trading Platforms | WebSocket | Live price updates |
Browser Support
Both WebRTC and WebSocket are well-supported in modern browsers:
✅ Fully Supported
- Chrome 23+ / Edge 79+
- Firefox 22+
- Safari 11+
- Opera 18+
- Mobile: iOS Safari 11+, Chrome Mobile
❌ Not Supported
- Internet Explorer (all versions)
- Old browser versions (pre-2015)
- Some corporate/restricted browsers
Troubleshooting Common Issues
WebRTC Not Working
- Update your browser: Ensure you're using a recent version
- Check permissions: Grant camera/microphone access when prompted
- Firewall/VPN: Corporate firewalls may block WebRTC traffic
- Browser extensions: Some privacy extensions block WebRTC
- Network restrictions: NAT and firewall configurations can interfere
WebSocket Connection Fails
- Firewall blocking: Corporate firewalls often block WebSocket ports
- Proxy issues: Some proxies don't support WebSocket protocol
- SSL/TLS: Use secure WebSocket (wss://) on HTTPS sites
- Browser security: Check if browser security settings block connections
- Server issues: The WebSocket server might be down or misconfigured
Frequently Asked Questions
What's the difference between WebRTC and WebSocket?
WebRTC is designed for peer-to-peer audio/video/data transfer directly between browsers, ideal for video calls and screen sharing. WebSocket provides client-server real-time communication, perfect for chat, notifications, and live updates. WebRTC is P2P (peer-to-peer), while WebSocket is client-server. Many applications use both: WebSocket for signaling and control, WebRTC for actual media streaming.
Do I need both WebRTC and WebSocket?
It depends on your application. For video calls, you typically need WebRTC for media streaming and WebSocket (or similar) for signaling to establish the connection. For text chat or live updates, WebSocket alone is sufficient. For collaborative tools, you might use WebSocket for text synchronization and WebRTC for optional voice chat.
Is WebRTC secure?
Yes, WebRTC is secure by design. All media streams are encrypted using DTLS (Datagram Transport Layer Security) and SRTP (Secure Real-time Transport Protocol). However, WebRTC can reveal your real IP address even when using a VPN, which is a privacy consideration. Browsers implement security measures, but developers must also follow best practices for signaling security.
Can WebSocket work through firewalls?
WebSocket typically works through standard firewalls since it uses HTTP ports (80/443) and starts as an HTTP request before upgrading to WebSocket protocol. However, some corporate firewalls or proxies specifically block WebSocket connections. Using secure WebSocket (wss:// over port 443) has better success rates as it looks like regular HTTPS traffic.
Why does my browser fail WebRTC tests?
Common reasons include: outdated browser version, disabled JavaScript, browser privacy extensions blocking WebRTC, corporate firewall restrictions, or browser security settings preventing media access. Try testing in a standard browser without extensions, ensure you're on HTTPS (not HTTP), and check that your browser is up to date.
What are STUN and TURN servers in WebRTC?
STUN (Session Traversal Utilities for NAT) servers help browsers discover their public IP address and port for P2P connections. TURN (Traversal Using Relays around NAT) servers relay traffic when direct P2P connection isn't possible due to restrictive firewalls. Most WebRTC applications need both for reliable connectivity across different network configurations.
How much bandwidth does WebRTC use?
WebRTC bandwidth usage varies greatly: audio-only calls use 24-128 kbps, standard video calls use 300-800 kbps, HD video calls use 1-2 Mbps, and 4K video can use 5-10 Mbps or more. WebRTC adapts to network conditions, reducing quality when bandwidth is limited. Screen sharing bandwidth depends on content complexity and frame rate.
Can WebSocket replace HTTP?
No, WebSocket doesn't replace HTTP but complements it. Use HTTP for traditional request-response patterns, caching, and RESTful APIs. Use WebSocket for real-time bidirectional communication. WebSocket connections even start as HTTP requests before upgrading. Most applications use both: HTTP for initial page loads and API calls, WebSocket for live updates.
What happens if WebSocket connection drops?
Good WebSocket implementations include reconnection logic. When a connection drops, the client should: detect the disconnection, wait a brief period, attempt to reconnect with exponential backoff, and restore application state after reconnecting. Users might experience temporary interruption of live updates but shouldn't lose data if the application handles reconnection properly.
Is WebRTC suitable for large group calls?
WebRTC P2P works well for 2-4 participants. Beyond that, you need either mesh architecture (each peer connects to all others, very bandwidth-intensive), SFU (Selective Forwarding Unit, server routes streams), or MCU (Multipoint Control Unit, server mixes streams). Most commercial video conferencing uses SFU for better scalability and quality.
Best Practices
🎯 Development Recommendations
- Always test both technologies: Verify browser support before using features
- Implement fallbacks: Provide alternative communication methods if WebRTC/WebSocket unavailable
- Use secure protocols: Always use HTTPS, wss://, and secure WebRTC connections
- Handle reconnection: Implement automatic reconnection for WebSocket drops
- Monitor performance: Track connection quality and adapt bitrates in WebRTC
- Test across browsers: Different browsers have varying WebRTC implementations
- Consider mobile: Mobile browsers may have different capabilities and bandwidth constraints
- Provide user feedback: Show connection status and quality indicators
- Optimize for mobile: Use lower resolutions and frame rates on mobile devices
- Respect privacy: Request permissions only when needed, explain why