Everything you need to know about the WebRTC Encoder

WebRTC (Web Real-Time Communication) is an open source code item that supports real-time communication between browsers. By using the WebRTC encoder, users can realize the functions of audio and video calls on web browsers. The following is an article about the webRTC encoder application:           

Application of WebRTC encoder

With the continuous development of Internet technology, real-time communication has become an indispensable part of people's daily life. Through the application of the WebRTC encoder, users can achieve high -quality audio and video calls on the web browser without installing any plug -in or third -party applications.

First of all, the WebRTC encoder uses the built -in audio and video codecs built by modern browsers to realize the ability to transmit real -time data transmission. This means that users can simply access a web link to make audio and video calls directly with other users without need to care about the complexity of the underlying technology.

Secondly, the WebRTC encoder provides a point -to -point (P2P) communication method, which means that communication data will be transmitted directly between the communication between the communication without passing the intermediate server. This not only improves the security of communication, but also reduces delay, making the quality of calls more stable and smooth.

In addition, the WebRTC encoder also supports interconnection with the traditional telephone network (PSTN). Users can dial and answer ordinary telephones through the WebRTC encoder to achieve a wider range of communication coverage.

Example of WebRTC on the encoder ui           

Easy to watch on the browser