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

Advanced Videoconferencing based on WebRTC

Lately, videoconference applications have experienced an evolution towards the World Wide Web. New technologies have given browsers real-time communications capabilities. In this context, WebRTC aims to provide this functionality by following and defining standards. Being a new effort, WebRTC still lacks advanced videoconferencing services such as session recording, media mixing and adjusting to varying network conditions. This paper analyzes these challenges and proposes an architecture based on a traditional communications entity, the Multipoint Control Unit or MCU as a solution

Desktop Sharing Portal

Desktop sharing technologies have existed since the late 80s. It is often used in scenarios where collaborative computing is beneficial to participants in the shared environment by the control of the more knowledgeable party. But the steps required in establishing a session is often cumbersome to many. Selection of a sharing method, obtaining sharing target’s network address, sharing tool’s desired ports, and firewall issues are major hurdles for a typical non-IT user. In this project, I have constructed a web-portal that helps collaborators to easily locate each other and initialize sharing sessions. The portal that I developed enables collaborated sessions to start as easily as browsing to a URL of the sharing service provider, with no need to download or follow installation instructions on either party’s end. In addition, I have added video conferencing and audio streaming capability to bring better collaborative and multimedia experience

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

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

Health-5G: A Mixed Reality-Based System for Remote Medical Assistance in Emergency Situations

Mixed reality is the combination of virtual and augmented reality to interactively and believably merge physical and computer-generated environments. This paper discusses the design of Health5G, a scalable mixed reality-based system that facilitates and supports emergency response by medical emergency teams. Health-5G is supported by a distributed architecture divided into four interrelated applications responsible for advanced computer-human interaction, effective real-time videoconference, medical device integration, and communication infrastructure, respectively. The mixed reality layer is provided by the headset Microsoft Hololens 2™. Health-5G is based on scenarios in which emergency personnel wear mixed reality glasses that can transmit audio, video, and data streams bidirectionally over a 5G network to medical specialists stationed in a hospital at any distance. Thanks to Health-5G, the specialist will be able to access the emergency team’s point of view at any time and provide verbal and visual instructions, including gestures and positioning of graphical markers in 3D space. In this way, emergency personnel can provide the best possible care to the patient without having to wait for them to arrive at the hospital, saving a lot of time in scenarios where every second can make a difference. Health-5G also addresses the integration of medical devices and the collection of the patient’s medical data in a scalable way through optical character recognition. A case study is discussed where Health-5G is used to attend a patient in the street suffering from syncope due to third-degree atrioventricular block. Latency and performance tests over a 5G network are also discussed. To the best of our knowledge, there is no comprehensive solution in the literature that provides all the capabilities offered by Health-5G in terms of functionality and advanced interaction mechanisms within the context of remote, immersive support in emergency situations

Adaptive cross-device videoconferencing solution for wireless networks based on QoS monitoring

The increase in CPU power and screen quality of todays smartphones as well as the availability of high bandwidth wireless networks has enabled high quality mobile videoconfer- encing never seen before. However, adapting to the variety of devices and network conditions that come as a result is still not a trivial issue. In this paper, we present a multiple participant videoconferencing service that adapts to different kind of devices and access networks while providing an stable communication. By combining network quality detection and the use of a multipoint control unit for video mixing and transcoding, desktop, tablet and mobile clients can participate seamlessly. We also describe the cost in terms of bandwidth and CPU usage of this approach in a variety of scenarios

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