Signalling servers for WebRTC technology with the use of node.js run-time environment

Videoconference System Based on WebRTC With Access to the PSTN

Web Real-Time Communication (WebRTC) introduces real-time multimedia communication as native capabilities of Web browsers. With the adoption of WebRTC the Web browsers will be able to use WebRTC to communicate with one another (peer-to-peer), and with WebSocket servers such as Mobicents SIP Servlets and other server technologies that support WebSocket communication to enable SIP-to-WebRTC communication. This paper outlines the potential of WebRTC and discusses the two common methods of doing real-time communication in Web browsers through WebRTC. The methods are JavaScript Object Notation (JSON) via XMLHttpRequest (XHR) and Session Initiation Protocol (SIP) via WebSocket. A three-user WebRTC video chat prototype application was developed and used to evaluate both methods. Additional signalling overhead introduced into a browser by each method was determined. The results showed WebRTC-SIP/WS has more overhead than WebRTC-JSON/XHR. These signalling overhead findings are useful in that they could help application developers make decision on their choice of technologies and protocols when developing WebRTC-supported applications.

Real-time communication (RTC) is a new standard and industry-wide effort that expand the web browsing model, allowing access to information in areas like social media, chat, video conferencing, and television over the internet, and unified communication. These systems users can view, record, remark, or edit video and audio content flows using time-critical cloud infrastructures that enforce the quality of services. However, there are many proprietary protocols and codecs available that are not easily interoperable and scalable to implement multipoint video-conference systems. WebRTC (Web Real-Time Communication) is a State-of-the-Art open technology that makes real-time communication capabilities in audio, video, and data transmission possible in real-time communication through web browsers using JavaScript APIs (Application Programming Interfaces) without plug-ins. This paper aims to introduce the P2P video conferencing system based on Web Real-Time Communication (WebRTC). In this paper, we have proposed a web-based peer-to-peer real-time communication system using the Mozilla Firefox together with the ScaleDrone service that enables users to communicate with high-speed data transmission over the communication channel using WebRTC technology, HTML5 and use Node.js server address. Our experiments show that WebRTC is a capable building block for scalable live video conferencing within a web browser.We also study a thermal camera for the videoconference system to identify the temperature body results regarding the COVID 19 crisis.

Existing video conferencing systems that are often used in telehealth services have been criticized for a number of reasons: (a) they are often too expensive to purchase and maintain, (b) they use proprietary technologies that are incompatible to each other, and (c) they require fairly skilled IT personnel to maintain the system. There is a need for less expensive, compatible, and easy-to-use video conferencing system. The web real-time communication (WebRTC) promises to deliver such a solution by enabling web browsers with real-time communications capabilities via simple JavaScript APIs. Utilizing WebRTC, users can conduct video/audio calls and data sharing through web browsers without having to purchase or download extra software. Though the promise and prospective of WebRTC have been agreed on, there have not been many cases of real life applications (in particular in telehealth) that utilizes the WebRTC. In this paper, we present our practical experience in the design and implementation of a video conferencing system for telehealth based on WebRTC. Our video conferencing system is a part of a large tele-home monitoring project that is being carried out at six locations in five different states in Australia. One of the aims of the project is to evaluate whether high-bandwidth enabled telehealth services, delivered through tele-home monitoring, can be cost effective, and improve healthcare outcomes and access to care. This paper however focuses on WebRTC-based video conferencing system which allows online meetings between remotely located care coordinators and patients at their home. We discuss the underlying issues, detailed design and implementation, and current limitations of using WebRTC in a real life application.

During an emergency situation disabled people face many obstacles to access the emergency systems since they fail to support IP-based services and do not take into account their accessibility requirements. This paper describes the key elements of the EMYNOS platform which is a Next Generation emergency platform that aims to provide IP based emergency services and integrate information provided by the social media, user's profile and contextual information in order to provide personalized emergency services and real time communication. To achieve this, tactile and pressure feedback from haptic devices as well as real time communication audio-video, text-video and voice-video calls will be initiated through the browser-to-browser web real time communication (WebRTC) technology. In addition, WebRTC technology will be used as a backup platform in case the default infrastructure collapses.

In the modern world, communication between people is being more important than ever, people are seeking for a more vivid, real-time way of communication. WebRTC (Web Real-Time Communication) is an upcoming technology that enables real-time communications capabilities in audio, video and data transmission via web browsers through Javascript APIs (Application Programming Interfaces). In this paper, we implemented a new design of a cross platform multimedia conferencing system based on WebRTC technology and HTML5. The system works well under low bandwidth 10Mbps network support with 800*600 resolution ratio. Test results indicate that the system is stable and fully functional, which offers new possibilities for individuals or small businesses to build their own multimedia conferencing system.

A simple signalling server, written in JavaScript and running in node.js run-time environment, was presented in the previous paper of the Authors [3]. In this paper, principles of building of WebRTC signalling client are presented, as well as exemplary communication between the client and the server, captured with the use of WireShark software tool during experimental WebRTC sessions. The paper includes also an example of application of Internet chat, that uses the simple signal-ling server [3]. The chat was built in two versions, which uses the same user interface and their signalling systems were build according to the same methodology, but differs in method of transmission of chat messages. The first version transmits user data (text messages of the chat) via signalling server, with the use of signalling channel. This version uses WebSocket interface for transmission of chat messages. In the case of the second version, user data are transmitted directly between web browsers, using WebRTC's channel dedicated for transmission of nonmedia data. This version transmits chat messages with the use of WebRTC's RTCDataChannel interface..

This paper describes the deployment of ATLAS offline software in containers for software development. For this we are using Docker, which is a lightweight virtualization technology that encapsulates a piece of software inside a complete file system snapshot. The deployment of offline releases via containers removes the strict requirement of compatibility between the runtime environment needed for job execution and the configuration of worker nodes at computing sites. If these two are decoupled from each other, sites can upgrade their nodes whenever and however they see fit. In this work, ATLAS software is distributed in containers either via the CernVM File System (CVMFS) or by means of a full ATLAS offline release installation. In software development, separating the build and runtime environment from the development environment allows users to take advantage of many modern code development tools that may not be available in production runtime setups like SLC6. It also frees developers from depending on resources like lxplus at CERN, allowing the use of an average laptop for ATLAS code development. We include in this document a basic comparison in performance of the two deployment options in two popular host operating systems: Ubuntu and OS X.

SummaryjsEvRBF is a time‐series forecasting method based on genetic algorithm and neural nets. Written in JavaScript language, can be executed in most web browsers. Consequently, everybody can participate in the experiments, and scientists can take advantage of nowadays available browsers and devices as computation environments. This is also a great challenge as the language support and performance varies from one browser to another. In this paper, jsEvRBF has been tested in a volunteer computing experiment, and also in a single‐browser one. Both experiments are related to forecasting currencies exchange, and the results show the viability of the proposal.

The fast mobility of HSR leads to its channel state’s more intense and frequent fluctuations, making it difficult to transmit real-time streaming media. In this paper, we propose an adaptive bitrate control algorithm for real-time streaming media transmission in HSR networks. At the receiver, we detect network congestion based on the one-way queuing delay and estimate the bandwidth according to the buffer state. To improve the sensitivity to the change of HSR network state, the bandwidth is estimated frequently. However, only the effective estimation results are fed back. The sender adjusts the bitrate according to the feedback bandwidth and the loss-based estimation bandwidth. Experiments and practical tests show that the proposed algorithm can effectively avoid unnecessary rate switching, reduce the video stalling and improve the bandwidth utilization of real-time streaming media transmission on HSR occasions.

LHC experiments are depending on a rich palette of software components to build their specific applications. These underlying software components include the ROOT analysis framework, the Geant4 simulation toolkit, monte carlo generators, grid middle-ware, graphics libraries, scripting languages, databases, tools, etc. which are provided centrally in up to date versions on multiple platforms (Linux, Mac, Windows). Until recently this set of packages has been tested and released in a tree like structure as a consistent set of versions across operating systems, architectures and compilers for LHC experiments only. Because of the tree like deployment these releases were only usable in connection with a configuration management tool which provided the proper build and run-time environments and was hindering other parties outside LHC from easily using this palette of packages.In a new approach the releases will be grouped in "flat structure" such that interested parties can start using it without configuration management, retaining all the above mentioned advantages. In addition to an increased usability the software shall also be distributed via system provided package deployment systems (rpm, apt, etc.). The approach of software deployment is following the ideas of providing a wide range of HEP specific software packages and tools in a coherent, up to date and modular way on multiple platforms. The target audience for such software deployments are individual developers or smaller development groups / experiments who don't have the resources to maintain this kind of infrastructure. This new software deployment strategy has already been successfully implemented for groups at CERN.

How do you effectively share a computational process? You could simply share a directory of source code and rely on shared conventions – shared programming language, shared tooling for build and runtime environments, a README.txt convention for communicating setup instructions, etc.

A adoção de tecnologias de informação e comunicação (TIC) nos mais diversos setores da economia, reflete o atual cenário competitivo, no qual processos e práticas podem ser otimizados, proporcionando significativas mudanças nos modelos de negócios. Neste sentido, o presente artigo descreve o desenvolvimento do Chatbot para classificação de grãos soja e cálculos dos descontos incidentes sobre a carga de grãos, possibilitando assim, confrontar os dados da classificação do Chatbot com os dados das unidades armazenadoras de grãos. Assim, considerando o aumento significativo de conversas mensais com Chatbots através de plataformas de mensagens e a relação conversacional entre organizações e usuários, bem como a carência de um dispositivo que atendesse a demanda do produtor de realizar em tempo real a classificação de grãos soja, que este projeto foi desenvolvido. Inicialmente foram realizadas entrevistas e visitas em unidades armazenadoras de grãos, como técnica para levantamento de requisitos e posterior definição da linguagem e modelagem a serem utilizados. O projeto foi desenvolvido em linguagem JavaScript, banco de dados NoSql e console Dialogflow para desenvolvimento do Chatbot. No final, foi possível constatar que o sistema criado aos sojicultores realizam cálculos de classificações e desconto evitando prejuízos a cada safra, de forma cômoda, ágil e seguindo a Instrução Normativa n° 11/2007 do Ministério da Agricultura, Pecuária e Abastecimento (MAPA), que define o padrão oficial de classificação de soja.

En la actualidad, las Tecnologías de Información y Comunicación (TIC) y asuntos relacionados, tales como el Internet de las Cosas (IoT, por sus siglas en inglés), tiene una influencia decisiva en todos los aspectos de la vida humana. IoT como fenómeno dominante es la transformación de la vida diaria mediante el uso de las funciones inteligentes como la de Identificación por Radio Frecuencia (RFID, por sus siglas en inglés) y las Redes de Tecnología de Sensores Inalámbricos (WSN, por sus siglas en inglés). Conforme el IoT avanza, se amplía en tamaño y dimensiones, mejorando muchos de los contextos de la sociedad; tales como, el sistema tradicional de biblioteca. Esta investigación propone un marco de ejecución sobre el empleo del IoT con el fin de renovar la estructura y esquema convencional de las bibliotecas a sistemas inteligentes en línea. El IoT permite la conectividad en tiempo real de un objeto físico (como un libro o cualquier otro tipo de texto) mediante el uso de las etiquetas RFID y sensores diminutos. El monitoreo continuo de los libros en tiempo real y la localización de objetos de manera geográfica son algunas de las características que se derivan del uso de las etiquetas de IoT. Estas características de IoT permiten la implementación de una línea de bibliotecas por medio de una cadena de suministros, la integración con diferentes tipos de tecnologías como la base de datos, la recopilación de datos y sistemas en la nube. El Internet de las Cosas también ofrece una panorámica de la vinculación entre el gran número de universidades y bibliotecas en el mundo en tiempo real, todo el tiempo. Se concluye que el implementar el IoT en sistemas de gestión de bibliotecas sería una estructura prometedora que puede jugar un papel vital en la organización de conocimientos del ser humano y en el acceso a informaciones para ayudar a investigadores, diseñadores y administradores en una manera más eficiente e inteligente.

The goal of this research is to present a detection algorithm for data interception between a Legitimate Access Point (LAP) and wireless devices employing passive eavesdropping techniques, analyzing the Protocol Data Units (PDUs) header of specific Open Systems Interconnection (OSI) protocol layers in real-time: segment (transport-layer), packet (network-layer), and frame (data link-layer). The designed algorithm takes advantage of a passive Man-in-the-Middle (MITM) system, avoiding to gather sensitive information of the users connected to the network. The system focuses on requests from web browsers and apps of wireless devices to analyze the online behavior of the users on the Internet utilizing cybersecurity techniques. The research approaches on a small-scale wireless network to implement educational environments based on parental control in educational institutions (primary, secondary and higher education), fomenting Information and Communication Technologies (ICT) security in virtual classrooms. The data acquisition process of the wireless devices uses the Dynamic Host Configuration Protocol (DHCP) and Domain Name System (DNS) services. It also analyzes the Transmission Control Protocol (TCP) control flags of the TCP three-way handshake protocol to identify requests from web browsers and apps. Finally, preliminary test results are presented, demonstrating that each wireless device can be characterized using information based on PDUs header, and these individual reports can be presented through an interactive dashboard to the teachers.