Does WebRTC Use WebSockets? (And When You Should)
WebRTC has become the go-to technology for real-time audio, video, and data sharing across browsers and devices. But many developers wonder — does WebRTC actually use WebSockets under the hood? The short answer: not for media transport, but it can for signalling.
Let’s break this down.
📡 How WebRTC Transports Media
At its core, WebRTC sends audio and video streams using RTP (Real-time Transport Protocol) over secure UDP. This is designed for low-latency delivery, which is crucial for live communication.
To establish this media path, WebRTC needs a signalling process — a way for peers to exchange information about codecs, network addresses, and encryption keys.
💬 Where WebSockets Come In
WebSockets are not built into the WebRTC spec for media transport. Instead, they can be used as one of many signalling transport options.
For example, you might:
- Use WebSockets to exchange SDP (Session Description Protocol) offers and answers.
- Send ICE candidates over WebSockets to help peers discover optimal connection routes.
This is especially popular in applications where persistent, bidirectional communication is needed between the browser and a server — such as chat systems or multi-user conferencing platforms.
🔍 WebRTC vs WebSockets — Key Differences
| Feature | WebRTC | WebSockets |
|---|---|---|
| Purpose | Real-time peer-to-peer audio, video, data | Persistent browser–server communication |
| Transport | SRTP/SRTCP over UDP/TCP | TCP-based |
| Built-in NAT Traversal | Yes (STUN/TURN) | No |
| Media Support | Yes | No (binary/text data only) |
🚀 When to Use Them Together
Many developers combine WebRTC and WebSockets to get the best of both worlds:
- WebRTC for peer-to-peer streaming of audio/video/data.
- WebSockets for signalling, room management, or chat.
A common architecture might use Node.js with Socket.IO to handle signalling logic, and then pass the connection to WebRTC for actual media exchange.
✅ Best Practices
- Keep signalling lightweight — once the WebRTC connection is established, media should flow directly peer-to-peer.
- Secure WebSockets with WSS to protect against eavesdropping.
- Use TURN servers as a fallback to guarantee connectivity when peer-to-peer isn’t possible.
Final Thoughts
WebRTC doesn’t require WebSockets for media transport, but pairing the two can make your real-time application more robust and feature-rich. Understanding their roles helps you design scalable, efficient communication systems.
For more articles like this one, visit SoftpageCMS for expert VoIP and WebRTC insights.
