What is WebRTC?
WebRTC (Web Real-Time Communication) is an API developed by the W3C and standardized by the RTCWEB IETF group, designed to enable peer-to-peer communication directly between web browsers. This communication can handle multiple types of data, including video, audio, and files.
Why WebRTC is Important
WebRTC allows browsers to establish direct channels for video and data communication without requiring external plugins or software. It is built into major browsers like Chrome, Firefox, and Opera, providing a seamless real-time experience for users.
How WebRTC Works
WebRTC relies on three core components:
- getUserMedia – Grants access to a device’s camera and microphone for capturing video and audio.
- RTCPeerConnection – Facilitates audio-video calls between devices.
- RTCDataChannel – Enables peer-to-peer data exchange between devices.
By leveraging these components, WebRTC establishes direct connections between devices, allowing efficient real-time communication.
WebRTC and iOS Support
WebRTC has long been supported by many browsers, but Apple’s support was delayed. Safari 11 added WebRTC support through WebKit, but on iOS, WebRTC is not fully native. While Android users can build fully native WebRTC apps, iOS limitations (with its 43% US market share) present challenges for browser-based communication apps on mobile platforms.
How WebRTC Works
WebRTC has a wide variety of use cases:
- Hangouts, Snapchat, WhatsApp – Real-time video and voice calling.
- Bank of America – Video chat support for ATMs.
- PeerCDN – File sharing across large networks.
- Robotics – Remote control of devices using WebRTC’s data channels.
The main reason for its adoption is that browser-to-browser communication reduces server costs significantly, sometimes by up to 90%, making it ideal for real-time applications.
Splendornet and WebRTC
At Splendornet, we have been leveraging WebRTC for years to build efficient real-time applications. Our implementations include:
- Web-based multi-user video conferencing
- Mobile audio-video chat systems
- Advanced features like bandwidth control, multiple peer connections, and canvas/video element streaming
Examples of WebRTC Applications We Implemented:
- Using getUserMedia with canvas and CSS filters
- Streaming from a video element to another video element
- Streaming from a video element to a peer connection
- Recording streams from a canvas element
- Dynamically changing bandwidth
- Managing multiple peer connections simultaneously
- Displaying peer connection states
- Using Web Audio output as input to peer connection
- Using peer connection as input to Web Audio
- RTCDataChannel for data exchange
WebRTC enables highly efficient, real-time communication and has been successfully integrated into many of our projects, delivering smooth and reliable user experiences.
