Over the past several years, part of the live streaming content dramatically increased. But the main problem of the live content is latency. Waiting for the answer to a simple question for about 5-10 seconds gets annoying and kills all communication dynamics, doesn’t it? These are typical latency values for standard streaming Live protocols like HLS or DASH. However, here is one of the game changers in Live streaming - WebRTC. This technology makes streaming with a millisecond’s long latency possible from all common browsers.

While cooking WebRTC on the frontend side is very simple, on the backend side, it’s a real pain in the neck: most of the open-source solutions (and to be honest, not only open-source) aren't suitable for actual high load. At the same time, there isn’t enough information on how to develop your WebRTC backend.

In this talk, we will:

- revise the theory, i.e. the definition of objects in video streaming, an overview of the most popular live streaming protocols nowadays, and what latency they have;
- explain what WebRTC is from the backend point of view. Why did we fail with the open source implementations? And how to make your WebRTC implementation on the backend side?
- discuss the top problems that you will face developing your own WebRTC and how to overcome them (picture freezes and artifacts, black screen, etc.)
- And last but not least, show the result of our development in numbers.