What is WebRTC?
Web Real-time Communications (WebRTC) is both an open source project and an industry standard and specification. It supports real-time transfer of local media, such as voice, video, and data, between the browser and the device. This enables users to communicate from their main Web browser without complex plug-ins or other hardware.
Google first announced the WebRTC project in May 2011 to develop a common protocol for enabling high-quality RTC applications in browsers, mobile platforms, and IoT devices. At the time, Flash and plug-ins were the only way to provide real-time communication. Two years later, after a lot of work, the first cross-browser video call was established between Chrome and Firefox. Since then, support for WebRTC has exploded in the developer community as more and more enterprises add support for the specification. Today, WebRTC is available natively in Chrome, Firefox, Safari, Edge, Android, and iOS to varying degrees, and is a widely popular video calling tool.
WebRTC API,
The WebRTC API has three main components, each of which plays a unique role in the WebRTC specification:
MediaStream (GetUserMedia) :
The MediaStream API provides a way to access the device’s camera and microphone using JavaScript. It controls where multimedia streaming data is consumed and provides some control over the device that produces the media. It also exposes information about devices that can capture and present media.
RTCPeerConnection:
Peer-to-peer connections are at the heart of the WebRTC standard. It provides a way for participants to establish a direct connection to their peers without the need for an intermediate server (except for signaling). Each participant inserts the media retrieved from the media streaming API into the peer connection to create an audio or video feed. There’s a lot going on behind the scenes with the PeerConnection API. It handles SDP negotiation, codec implementation, NAT traversal, packet loss, bandwidth management, and media transfer.
RTCDataChannel:
The RTCDataChannel API is set up to allow two-way data transfer of any type of data (media or otherwise) directly between peers. It is designed to mimic the WebSocket API, rather than rely on TCP connections, which use UDP-based streams and have the configurability of the Flow Control Transport Protocol (SCTP) protocol, despite high reliability, latency, and bottlenecks. This design has the best of both worlds: reliable delivery like TCP, but less network congestion like UDP.
The open source framework WebRTC has been developed for 10 years and has now become an official Web standard
Read more: EasyRTC Video conferencing cloud service
EasyRTC is the global real-time audio development platform developed by TSINGSEE Qingxi Video team based on Webrtc, supporting one-to-one and one-to-many video calls.
EasyRTC has MCU and SFU architecture, no need to install client and plug-in, pure H5 online video conference system, support wechat small program, H5 page, APP, PC client and other access methods, greatly meet the needs of voice and video social, online education and training, video conference and telemedicine and other scenarios.
With the rapid development of mobile Internet, AI, 5G and other emerging technologies, combined with WebRTC technology, more application scenarios will be derived in the future, changing the way of life such as clothing, food, housing and transportation of human beings.