WebRTC

Most real-time web features keep a server in the middle. Long polling, SSE, and WebSockets are good fits for notifications, chat, dashboards, and collaborative state, all of which route data through your application servers.

Video calls are different. Sending every audio and video packet through your servers adds latency and burns bandwidth.

WebRTC is a set of browser APIs and network protocols for low-latency audio, video, and data transfer directly between peers when the network allows it.

WebRTC is not "magic peer-to-peer." Most users are behind NATs, firewalls, mobile networks, or corporate proxies, so a production WebRTC system still needs servers for signaling, NAT traversal, relay fallback, and group-call scaling.

This chapter covers what WebRTC does and what it deliberately leaves to your application, how signaling, SDP, ICE, STUN, and TURN fit together, how media and data move once a connection is established, why group calls usually need media servers, and what to monitor when operating WebRTC in production.