Multi-user media streaming service for e-learning based web real-time communication technology

Abstract

Web real-time communication (WebRTC) standards do not define precisely how two browsers establish and control their communication. Therefore, a signalling mechanism/protocol has not specified in WebRTC. The essential goal of this research is to create and apply a WebRTC bi-directional video conferencing based on mesh topology (many-to-many) using Google Chrome, Firefox, Opera, and Explorer. This experiment involved through Ethernet and Wireless of the Internet and 4G networks in e-learning. The signalling mechanism of this experiment has been created and implemented using JavaScript language along with MultiConnection libraries. In addition, an evaluation of quality of experience (QoE), resources, such as bandwidth consumption, and CPU performance was done. In this paper, a novel implementation was accomplished over e-learning using different networks, different browsers, many peers, opening one or many rooms concurrently, defining room initiator, sharing the information of the new user with participants, using user identification (user-id), and so on. Moreover, the paper also highlights the advantages and disadvantages of using WebRTC video conferencing.

Evaluating the Forest Ecosystem through a Semi-Autonomous Quadruped Robot and a Hexacopter UAV

“…For example, the Real-Time Streaming Protocol (RTSP) and HTTP Live Streaming (HLS) [ 93 , 94 ], which are widely used for streaming, offer latency of up to 3 s. This was more than what was required for time-critical use cases. We use Web Real-Time Communication (WebRTC) [ 95 ], which provided a latency that was less than 250 ms. However, it also presented a new challenge, i.e., WebRTC was not supported in Python.…”