Web Real-Time Communication Technology Research Articles

VCVERSE – A Video Conferencing Website with Controls

In today’s digital age, remote communication and collaboration are becoming increasingly important. Video Conferencing platforms have gained popularity as they provide an immersive and interactive way to connect with others. This paper aims to develop a video chat website with a built-in notes controller, using Nodejsand Python. One of the special features of the video chat website is the integrated note taking feature. Users candownload the files during a video call to improve collaboration and information sharing. These files are synchronizedin real –time with all participants to ensure that everyone can refer to the important points discussed during the call. To enable real-time video communication, WebRTC (Web Real-Time Communication) technology is integrated into the website. WebRTC enables browser-based peer-to-peer video and audio communication without the need for additional plugins or softwareinstallations.The resulting video chat website with features will provide a user-friendlyandcollaborateplatformforremotecommunicationandteamwork.Itcanbeusedinvarious fields such as education, business meetings, and remote collaboration of distributed teams.

Peer-to-peer media streaming with HTML5

<p class="Affiliation"><span lang="EN-US">The knowledge of web real-time communication (WebRTC) and how its customers and server operations are defined in this study. However, the world wide web consortium (W3C) and the internet engineering task force (IETF) have not yet approved an absolute signaling protocol or a complete application programming interface (API) protocol to implement WebRTC and control communication planning. WebRTC requires some type of signaling mechanism. With Chrome, Firefox, and Opera, the primary objective is to create and implement a WebRTC video call across two clients (peers) in the real world while employing local area network (LAN) and wide area network (WAN). This study also demonstrated the design of the server (as an intermediary) and graphical user interface (GUI). Additionally, a signaling method for the peer-to-peer browser connection on the Node.js platform has been developed and successfully put into use. This paper will provide an understanding of web development and an ability to understand WebRTC technology. It will also discuss how to create WebRTC signaling mechanisms and build video conferencing, as well as how to increase design quality using quality of experience (QoE) techniques.</span></p>

Traversal Using Relays Around Nat by Coturn for Webrtc Systems

Real-time communication between people in the form of audio, images, video, and data is now a necessity. Web Real-Time Communication Technology, WebRTC, is implemented as an open web standard and is available to build such communication. WebRTC requires a server that functions as both signaling and media servers. Signaling server for establishing communication channels, exchanging information, and synchronizing changes. Media servers exchange media, codecs, bandwidth, and rate control. Different Real-time Protocol (RTP) topologies have different requirements on servers, including those caused by the NAT configuration used. This research focuses on the role of the TURN server using CoTurn to create web-based real-time communication services. How is the quality of WebRTC service when CoTurn is a TURN server to pass signaling and media. The method used is divided into three stages, namely: (1) installation and configuration, (2) building WebRTC, and (3) testing. This study shows that real-time communication in the form of audio and video can be carried out even through a symmetrical NAT configuration. It is just that with more than five users in the WebRTC system, communication is slowing down. The proposed further research is developing and engineering the RTP topology to increase the number of users. The CoTurn application can be installed on a Virtual Private Server (VPS) and configured as a TURN Server as a media server service in the WebRTC system.

Evaluation of WebRTC in the Cloud for Surgical Simulations: A case study on Virtual Rotator Cuff Arthroscopic Skill Trainer (ViRCAST).

Web Real-Time Communication (WebRTC) is an open-source technology which enables remote peer-to-peer video and audio connection. It has quickly become the new standard for real-time communications over the web and is commonly used as a video conferencing platform. In this study, we present a different application domain which may greatly benefit from WebRTC technology, that is virtual reality (VR) based surgical simulations. Virtual Rotator Cuff Arthroscopic Skill Trainer (ViRCAST) is our testing platform that we completed preliminary feasibility studies for WebRTC. Since the elasticity of cloud computing provides the ability to meet possible future hardware/software requirements and demand growth, ViRCAST is deployed in a cloud environment. Additionally, in order to have plausible simulations and interactions, any VR-based surgery simulator must have haptic feedback. Therefore, we implemented an interface to WebRTC for integrating haptic devices. We tested ViRCAST on Google cloud through haptic-integrated WebRTC at various client configurations. Our experiments showed that WebRTC with cloud and haptic integrations is a feasible solution for VR-based surgery simulators. From our experiments, the WebRTC integrated simulation produced an average frame rate of 33 fps, and the hardware integration produced an average lag of 0.7 milliseconds in real-time.

Using Socket.io Approach for Many-to-Many Bi-Directional Video Conferencing

Video conferencing has become a critical need in today’s world due to its importance in education and business to mention a few; also, recent years have witnessed a great revolution in communication technologies. However, there still exist limitations in these technologies in terms of the quality of communication established between two peers. Therefore, many solutions have been suggested for a variety of video conferencing applications. One of these technologies is Web Real-Time Communication technology (WebRTC). WebRTC provides the ability to efficiently perform peer-to-peer communication, which improves the quality of the communication. This work tries to propose a WebRTC bi-directional video conferencing for many-to-many (mesh topology) peers. In this work, signaling was obtained using Socket.io Library. The performance evaluation of the proposed approach was performed in terms of CPU performance, and Quality of Experience (QoE). Moreover, to validate the simulations results, a real implementation was achieved based on the following scenarios a) involving several peers, b) at the same time, opening several video rooms, c) a session will still be active even when the room initiator leaves, and d) new users can be shared with currently involved participants.

Design and implement a new mechanism for audio, video and screen recording based on WebRTC technology

Many years ago, Flash was essential in browsers to interact with the user media devices, such as a microphone and camera. Today, Web Real-Time Communication (WebRTC) technology has come to substitute the flash, so browsers do not need the flash to access media devices or establish their communication. However, WebRTC standards do not express precisely how browsers can record audios, videos or screen instead of describing getUserMedia API that enables a browser to access microphone and camera. The prime objective of this research is to create a new WebRTC recording mechanism to record audios, videos, and screen using Google Chrome, Firefox, and Opera. This experiment applied through Ethernet and Wireless of the Internet and 4G networks. Also, the recording mechanism of this research was obtained based on JavaScript Library for audio, video, screen (2D and 3D animation) recording. Besides, different audio and video codecs in Chrome, Firefox and Opera were utilised, such as VP8, VP9, and H264 for video, and Opus codec for audio. Not only but also, various bitrates (100 bytes bps, 1 Kbps, 100 Kbps, 1 MB bps, and 1 GB bps), different resolutions (1080p, 720p, 480p, and HD (3840* 2160)), and various frame-rates (fps) 5, 15, 24, 30 and 60 were considered and tested. Besides, an evaluation of recording mechanism, Quality of Experience (QoE) through actual users, resources, such as CPU performance was also done. In this paper, a novel implementation was accomplished over different networks, different browsers, various audio and video codecs, many peers, opening one or multi browsers at the same time, keep the streaming active as much as the user needs, save the record, using only audio and/or video recording as conferencing with full screen, etc.

WebRTC 기반 원격 협업 학습 플랫폼 기술 연구

Recently, as the number of smart devices (such as smart TV or Web based IPTV) increases, the way of digital broadcast contents is changed. This change leads that conventional broadcast media accepts Web platform and its services to provide more quality contents. Based on this change, in education field, education broadcasting also follows the trend. The traditional education broadcasting platforms, which just delivered the lecture in one-way, are utilized the Web technology to make interaction between teacher and student. Current education platforms, however, are insufficient to satisfy users' demands for two-way interactions. This paper proposes a new remote collaborative learning platform which able to provide high interactivity among users. Based on new functional requirements from original use case, the platform provides collaborative contents sharing and collaborative video streaming techniques by utilizing WebRTC (Web Real-Time Communication) technology. The implementation demonstrates the operability of proposed system.