Can Video Conferencing Be as Easy as Telephoning?-A Home Healthcare Case Study

Copyright © 2016 by authors and Scientific Research Publishing Inc. This work is licensed under the Creative Commons Attribution International License (CC BY).In comparison with almost universal adoption of telephony and mobile technologies in modern day healthcare, video conferencing has yet to become a ubiquitous clinical tool. Currently telehealth services are faced with a bewildering range of video conferencing software and hardware choices. This paper provides a case study in the selection of video conferencing services by the Flinders University Telehealth in the Home trial (FTH Trial) to support healthcare in the home. Using pragmatic methods, video conferencing solutions available on the market were assessed for usability, reliability, cost, compatibility, interoperability, performance and privacy considerations. The process of elimination through which the eventual solution was chosen, the selection criteria used for each requirement and the corresponding results are described. The resulting product set, although functional, had restricted ability to directly connect with systems used by healthcare providers elsewhere in the system. This outcome illustrates the impact on one small telehealth provider of the broader struggles between competing video conferencing vendors. At stake is the ability to communicate between healthcare organizations and provide public access to healthcare. Comparison of the current state of the video conferencing market place with the evolution of the telephony system reveals that video conferencing still has a long way to go before it can be considered as easy to use as the telephone. Health organizations that are concerned to improve access and quality of care should seek to influence greater standardization and interoperability though cooperation with one another, the private sector, international organizations and by encouraging governments to play a more active role in this sphere

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

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

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

Design and implementation of a novel secured and wide WebRTC signalling mechanism for multimedia over internet

A modern and free technology called web real-time communication (WebRTC) was enhanced to allow browser-to-browser multimedia communication without plugins. In contract, WebRTC has not categorised a specific signalling mechanism to set, establish and end communication between browsers. The primary target of this application is to produce and implement a novel WebRTC signalling mechanism for multimedia communication between different users over the Internet without plugins. Furthermore, it has been applied over different browsers, such as Explorer, Safari, Google Chrome, Firefox and Opera without any downloading or fees. This application designed using JavaScript language under ASP.net and C# language. Moreover, to prevent irrelevant users from accessing or attacking the session, user-id for initiating and joining the course using encryption technique was done. This system has been employed in real implementation among various users; therefore, an evaluation of bandwidth consumption, CPU, and quality of experience (QoE) was accomplished. The results show an original signalling mechanism which applied to different browsers, multiple users, and diverse networks such as Ethernet and Wireless. Besides, it sets session initiator, saves the communication efficient even if the initiator leaves, and communicating new participator with existing participants, etc. This studying focuses on the creation of a new signalling mechanism, the limitations of resources for WebRTC video conferencing

WebNSM a novel scalable WebRTC signalling mechanism for many-to-many video conferencing

There is a strong focus on the use of Web Real-Time Communication (WebRTC) for many-to-many video conferencing, while the IETF working group has left the signalling issue on the application layer. The main aim of this paper is to create a novel scalable WebRTC signalling mechanism called WebNSM for many-to-many (bi-directional) video conferencing. WebNSM was designed for unlimited users over the mesh topology based on Socket.io (API) mechanism. A real implementation was achieved via LAN and WAN networks, including the evaluation of bandwidth consumption, CPU performance, memory usage, maximum links and RTPs calculation; and Quality of Experience (QoE). In addition, this application supplies video conferencing on different browsers without having to download additional software or user registration. The results present a novel signalling mechanism among various users, devices and networks to open one or multi rooms at the same time using the same server, determine room initiator to keep the session active even if the initiator or another peer leaves, sharing new user with current participants, etc. Moreover, this experiment highlights the limitations of CPU performance, bandwidth consumption and using mesh topology for WebRTC video conferencing

Architecture and Protocol to Optimize Videoconference in Wireless Networks

[EN] In the past years, videoconferencing (VC) has become an essential means of communications. VC allows people to communicate face to face regardless of their location, and it can be used for different purposes such as business meetings, medical assistance, commercial meetings, and military operations. There are a lot of factors in real-time video transmission that can affect to the quality of service (QoS) and the quality of experience (QoE). The application that is used (Adobe Connect, Cisco Webex, and Skype), the internet connection, or the network used for the communication can affect to the QoE. Users want communication to be as good as possible in terms of QoE. In this paper, we propose an architecture for videoconferencing that provides better quality of experience than other existing applications such as Adobe Connect, Cisco Webex, and Skype. We will test how these three applications work in terms of bandwidth, packets per second, and delay using WiFi and 3G/4G connections. Finally, these applications are compared to our prototype in the same scenarios as they were tested, and also in an SDN, in order to improve the advantages of the prototype.This work has been supported by the "Ministerio de Economia y Competitividad" in the "Programa Estatal de Fomento de la Investigacion Cientifica y Tecnica de Excelencia, Subprograma Estatal de Generacion de Conocimiento" within the project under Grant TIN2017-84802-C2-1-P.Jimenez, JM.; García-Navas, JL.; Lloret, J.; Romero Martínez, JO. (2020). Architecture and Protocol to Optimize Videoconference in Wireless Networks. Wireless Communications and Mobile Computing. 2020:1-22. https://doi.org/10.1155/2020/4903420S122202

Design and Implement a Hybrid WebRTC SignallingMechanism for Unidirectional & Bi-directional VideoConferencing

WebRTC (Web Real-Time Communication) is a technology that enables browser-to-browser communication. Therefore, a signalling mechanism must be negotiated to create a connection between peers. The main aim of this paper is to create and implement a WebRTC hybrid signalling mechanism named (WebNSM) for video conferencing based on the Socket.io (API) mechanism and Firefox. WebNSM was designed over a combination of different topologies, such as simplex, star and mesh. Therefore it offers several communications at the same time as one-to-one (unidirectional/bidirectional), one-to-many (unidirectional) and many-to-many (bi-directional) without any downloading or installation. In this paper, WebRTC video conferencing was accomplished via LAN and WAN networks, including the evaluation of resources in WebRTC like bandwidth consumption, CPU performance, memory usage, Quality of Experience (QoE) and maximum links and RTPs calculation. This paper presents a novel signalling mechanism among different users, devices and networks to offer multi-party video conferencing using various topologies at the same time, as well as other typical features such as using the same server, determining room initiator, keeping the communication active even if the initiator or another peer leaves, etc. This scenario highlights the limitations of resources and the use of different topologies for WebRTC video conferencing

Dynamic video bitrate adaptation for WebRTC-based remote teaching applications

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