High Performance Browser Networking

Chapter 18. WebRTC

Web Real-Time Communication (WebRTC) is a collection of standards, protocols, and JavaScript APIs, the combination of which enables peer-to-peer audio, video, and data sharing between browsers (peers). Instead of relying on third-party plug-ins or proprietary software, WebRTC turns real-time communication into a standard feature that any web application can leverage via a simple JavaScript API.

Delivering rich, high-quality, RTC applications such as audio and video teleconferencing and peer-to-peer data exchange requires a lot of new functionality in the browser: audio and video processing capabilities, new application APIs, and support for half a dozen new network protocols. Thankfully, the browser abstracts most of this complexity behind three primary APIs:

• MediaStream: acquisition of audio and video streams

• RTCPeerConnection: communication of audio and video data

• RTCDataChannel: communication of arbitrary application data

All it takes is a dozen lines of JavaScript code, and any web application can enable a rich teleconferencing experience with peer-to-peer data transfers. That’s the promise and the power of WebRTC! However, the listed APIs are also just the tip of the iceberg: signaling, peer discovery, connection negotiation, security, and entire layers of new protocols are just a few components required to bring it all together.

Not surprisingly, the architecture and the protocols powering WebRTC also determine its performance characteristics: ...