Everything You Ever Wanted to Know About WebRTC

A primer around WebRTC with everything you need to get started

Live messaging, streaming, Torrents, and other similar real-time data transfer techniques have greatly affected and improved our virtual experiences. First, they were available natively, now through the web, bringing an impressive amount of possibilities to this universal platform. That’s all thanks to APIs known as WebSocket and WebRTC.

WebSocket revision

Now, I bet that you’ve heard of WebSocket. In fact, you’ve likely used it in some of your previous projects. Still, if not, here’s a brief overview.

WebSocket is a simple API that allows you to connect your clients indirectly, i.e., through a server. Basically, if one user wants to send some data to the other one, they both need to be connected with the server first and then pass the data through it. This approach naturally comes with some pros and cons. While it allows you to optionally process data on the server-side and assures the completeness of sent information (through TCP protocol), it isn’t really fast. Because of the proxy-like server, a generic transfer can take up to 1 second, which isn’t much, but can make a huge difference in use-cases like audio and video calls.