WebRTC is just a consensus between multiple parties about how to coordinate and send pure data, or just audio and video, from one peer to another. Then that ph

WebRTC is just a consensus between multiple parties about how to coordinate and send pure data, or just audio and video, from one peer to another. Then that phrase would extend into multiple layers of complexity. This could be the first one: - The "coordination" part is done with SDP, a type of plain-text messages that are used by both parties to reach an agreement about what video/audio codecs they are capable of understanding, and other similar details. - The "send audio and video" part is done by using ICE to find suitable ports by punching holes in router NATs, which is a common problem we have in current networks. Once ports have been agreed upon, then the audio/video is sent using good old (encrypted) RTP protocol. Anyone could implement this consensus in their applications, and so be able to send audio/video to other apps that implement the same details, but the most relevant implementations of this standard are those integrated into web browsers, because then they expose all this functionality to their JavaScript engines, thus allowing to control all this from a web page.