Why to Use WebRTC in Mobile Application
Web Real-Time Communication (WebRTC) is an open-source project, which enables mobile applications and web browsers to real-time communicate via simple APIs. It allows you to make audio and video communication inside one platform and thus eliminates the need to install separate applications and install plugins.
WebRTC is supported by different mobile companies and web browsers, such as Opera, Mozilla, Microsoft, Google, and Apple and is standardized through the Internet Engineering Task Force and World Wide Web Consortium. To acquire the above-mentioned features, WebRTC implements the following APIs.
It represents synchronized streams of media, such as audio tracking and video streaming at the same time when making a video call. This API is also known as GetUserMedia API. Its input data from the users and keep the Media Streams, audio and video, synchronized, which is then transferred to another peer in a synchronized way.
The WebRTC Uses RTCPeerConnection API when communicating and streaming data between browsers. It handles efficient and stable communication and streaming between peers. In this process, the application facilitates users for encryption and bandwidth management. Along with this feature, the application requires a mechanism to send control messages and coordinate communication, which is known as signaling. For this purpose, the developers use different protocols, such as XMRP and SIP.
This API enables the peer-to-peer exchange of arbitrary data with high throughput and low latency. This API has several potential uses, such as decentralized networks, file transfer, real-time text chat, remote desktop applications, and gaming. Several features of RTCDataChannel API enable flexible and powerful peer-to-peer communication.
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