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WebRTC (Web Real-Time Communication) is a technology that enables Web applications and sites to capture and optionally stream audio and/or video media, as well as to exchange arbitrary data between browsers without requiring an intermediary. The set of standards that comprise WebRTC makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user install plug-ins or any other third-party software.
WebRTC consists of several interrelated APIs and protocols which work together to achieve this. The documentation you'll find here will help you understand the fundamentals of WebRTC, how to set up and use both data and media connections, and more.
Here are some of the main use cases you’ll find for WebRTC out there:
- Unified communications – voice and video calling, 1:1 or group sessions
- Contact center communications – client/agent, visual assistance, remote assistance, etc
- Watch parties – watch television or a sports event together
- eCommerce and retail – from one-to-one high touch sales to live broadcast for sales events and promotions
- Telehealth, online education, legal proceedings, remote travel, fitness, dancing, tutoring, coaching, … – conduct remotely and virtually verticalized sessions you would have done in-person in the past
- Teleoperations – drive cars, forklifts, trucks, drones, boats, submarines, … – remotely
- Virtual and hybrid events – conduct webinars, large meetings and events online
- Low latency broadcasting – broadcast a sports game, auction or interactive sessions to a large audience at sub second latency
- Cloud gaming – render the visuals of a game in the cloud and send it in realtime to the player
- Machine remoting – operate a remote machine (high performance machines or highly secured/configured machines) as if it was a local one
- Virtual spaces and the metaverse – meet people in a synthetically rendered virtual environment in 2D or 3D