One of the most exciting APIs to come out in recent years is WebRTC (which stand for Web real-time communication). The purpose of this API is to allow users to communicate in real-time streaming audio and video across platforms that support the technology.
WebRTC is made up of several individual APIs and can be broken down into three separate components, namely
getUserMedia (which we'll discuss in more depth in the next section),
Since we'll discuss
getUserMedia in the next section, we'll leave a more involved definition for it when we get there (although the name might give away what the API is intended to do).
RTCPeerConnection is what we use to connect two peers together. Once a ...