WebRTC
Last Updated: January 9, 2026
Every real-time technology we have covered so far routes traffic through servers. Long polling, WebSockets, and SSE all require a server in the middle. For chat messages and notifications, that is fine.
But what about video calls?
When you are on a video call with someone across the world, routing every video frame through a server adds latency and costs a fortune in bandwidth.
WebRTC (Web Real-Time Communication) solves this by enabling direct peer-to-peer connections between browsers. Once the connection is established, audio and video flow directly from one browser to another without touching a server.
This dramatically reduces latency and server costs for real-time media applications.
But peer-to-peer is not simple. Browsers sit behind firewalls and NATs. They do not have public IP addresses. Establishing a connection requires a dance of signaling, ICE candidates, and STUN/TURN servers.
Understanding this complexity is essential for building video calling, screen sharing, and other peer-to-peer applications.
In this chapter, you will learn:
- How WebRTC establishes peer-to-peer connections
- The role of signaling servers, STUN, and TURN
- The WebRTC APIs for media and data channels
- When to use WebRTC versus server-mediated communication
- Architecture patterns for production WebRTC applications