Abstract
To connect with user media devices such as microphones and cameras, browsers formerly required Flash. Flash has essentially replaced Web Real-Time Communication (WebRTC) in recent years. The principles of WebRTC have not agreed on how browsers can capture audio, video, and data or screens. The major purpose of this research is to create a new WebRTC recording method that uses Google Chrome and Firefox to capture a mixture of cameras, microphones, and screens. In addition, this study used Software Engineering (SE) concepts such as the software design process and interface design, which is the description of a system's relationships with its environment, including the analysis and development of the designed systems. The MultiStreamsMixer.js libraries were used to design and implement a mixer of many cameras, microphones, and screens using the Ethernet and Wireless of the 4th generation (4G) network. The suggested technique also makes use of the JavaScript Library to capture audio, video, and screen (two-dimensional and three-dimensional animations); as well as numerous audio and video codecs, such as VP8 and VP9 for video and Opus for audio, were also utilized in Chrome and Firefox. Additionally, multiple bitrates ranging from 100 bytes per second to 1 Gigabit per second were also tested. Besides, various resolutions ranging from 480p to HD (3840* 2160) and frame rates ranging from 5 to 70, increasing by 5% each time were applied. In addition, the recording device, Quality of Experience (QoE) over real operators, and resources were evaluated.
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