scholarly journals Performance Analysis of WebRTC and SIP for Video Conferencing

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.

Author(s):  
Isac Gnanaraj J ◽  
Sriram .

One of emerging trends in the mobile network era is Network Mobility (NEMO). It was standardized by the Internet Engineering Task Force (IETF) and gained attention of the researchers because of research opportunities that it provides. Though it was developed based on MIPv6, there are few spots that must be analyzed and rectified, especially in the security aspects. According to the literatures, NEMO lacks in providing a robust Authentication, Authorization and Accounting (AAA) services to its users. AAA operations must be performed for all the players of the mobile network, because a hacker may reside at any place and try to access the mobile network by hiding behind valid or genuine nodes’ addresses. This research work aims to provide an AAA framework for NEMO by comprising three different mechanisms which are developed for Local Mobile Node (LMN), Visiting Mobile Node (VMN) and Mobile Router (MR). Simulation and performance analysis are done.


Author(s):  
German Dario Beltran Constain ◽  
Dario Andres Benavides Moreno ◽  
Victor Manuel Quintero Florez ◽  
Jenny Cuatindioy Imbachi

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.


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


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