scholarly journals Video Conferencing WebApp

Author(s):  
Sanjay Majhi

Abstract: During the last few years, video conferencing has become very popular and very reliable as a tool to bridge the gap where travel is not an option. And the COVID-19 epidemic has also led to lockdown orders that have led to dramatic changes in the way people work. The number of people working in the home (WFH) had increased significantly during the pandemic. The need for distance learning has also increased and has become a compulsory education system in the midst of this current situation. The Companies are also adopting an innovative recruitment process at such time. So to address this issue, our project aims to build a conference app that helps to provide communication between people through audio conferencing, video conferencing, screen sharing and messaging in real time. In this, we have created group video chat with the help of WebRTC technology and socket programming. Also we have added real-time chat feature and screen share feature. We had created the web app using Jquery for front end and node.js express.js for signaling server and real time database of Firebase for storing chats and user information. WebRTC helped us to create peer to peer connection and with the help of sockets we have done transfer of sdp packets and ice candidates. We have discussed extensively about them in our paper. Keywords: Video conferencing app, sockets, webrtc, peer-to-peer, realtime data transfer

Author(s):  
Rahul Kumar Mohata ◽  
Amita Goel ◽  
Vasudha Bahl ◽  
Nidhi Sengar

The covid-19 pandemic has led to things happening virtually. Students are attending their classes in online mode. More than 50 percent of the working population is working from home. Online meetings have become necessary part of everyone's life. With the existing platforms, users need to setup or install packages on their systems to run the application which sometimes becomes confusing for first timers or non-technical people. This paper proposes to build a full-fledged feature rich web-based video conferencing application using WebRTC technology. WebRTC is used to enable real time audio and video communication from a web browser without the need of installing software or plugins so that users can focus on their work rather than worrying about how to use a video conferencing platform.


With the advancement in communication and development of technologies like VoIP and Video Conferencing, Web Real-Time Communication (WebRTC) is developed to communicate without plugins and stream the videos on a real time. It was initially developed by Web Consortium(W3C) and Internet Engineering Task Force (IETF). It allows to transfer videos and audios between different browsers. This research paper, analyse the parameters during the call in different browsers and conditions (number of end points). The concept of WebRTC is inspired from Session Initiation Protocol(SIP). It helps in the establishment of sessions and maintain it. It also supports data and message transmissions. It also works on remote location and different network transmission protocols. It also allows peer to peer communication. In this research work, we examine the behaviour of WebRTC and SIP during the call from different browsers. We examine the different parameters like packets sent, jitter, VO-Width and bandwidth during the call and call supported on cloud during our experimental work.


2021 ◽  
Vol 77 (2) ◽  
pp. 98-108
Author(s):  
R. M. Churchill ◽  
C. S. Chang ◽  
J. Choi ◽  
J. Wong ◽  
S. Klasky ◽  
...  

Author(s):  
Chun-ying Huang ◽  
Yun-chen Cheng ◽  
Guan-zhang Huang ◽  
Ching-ling Fan ◽  
Cheng-hsin Hsu

Real-time screen-sharing provides users with ubiquitous access to remote applications, such as computer games, movie players, and desktop applications (apps), anywhere and anytime. In this article, we study the performance of different screen-sharing technologies, which can be classified into native and clientless ones. The native ones dictate that users install special-purpose software, while the clientless ones directly run in web browsers. In particular, we conduct extensive experiments in three steps. First, we identify a suite of the most representative native and clientless screen-sharing technologies. Second, we propose a systematic measurement methodology for comparing screen-sharing technologies under diverse and dynamic network conditions using different performance metrics. Last, we conduct extensive experiments and perform in-depth analysis to quantify the performance gap between clientless and native screen-sharing technologies. We found that our WebRTC-based implementation achieves the best overall performance. More precisely, it consumes a maximum of 3 Mbps bandwidth while reaching a high decoding ratio and delivering good video quality. Moreover, it leads to a steadily high decoding ratio and video quality under dynamic network conditions. By presenting the very first rigorous comparisons of the native and clientless screen-sharing technologies, this article will stimulate more exciting studies on the emerging clientless screen-sharing technologies.


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