238 research outputs found

    Smart hospital emergency system via mobile-based requesting services

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    In recent years, the UK’s emergency call and response has shown elements of great strain as of today. The strain on emergency call systems estimated by a 9 million calls (including both landline and mobile) made in 2014 alone. Coupled with an increasing population and cuts in government funding, this has resulted in lower percentages of emergency response vehicles at hand and longer response times. In this paper, we highlight the main challenges of emergency services and overview of previous solutions. In addition, we propose a new system call Smart Hospital Emergency System (SHES). The main aim of SHES is to save lives through improving communications between patient and emergency services. Utilising the latest of technologies and algorithms within SHES is aiming to increase emergency communication throughput, while reducing emergency call systems issues and making the process of emergency response more efficient. Utilising health data held within a personal smartphone, and internal tracked data (GPU, Accelerometer, Gyroscope etc.), SHES aims to process the mentioned data efficiently, and securely, through automatic communications with emergency services, ultimately reducing communication bottlenecks. Live video-streaming through real-time video communication protocols is also a focus of SHES to improve initial communications between emergency services and patients. A prototype of this system has been developed. The system has been evaluated by a preliminary usability, reliability, and communication performance study

    Advanced Videoconferencing based on WebRTC

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    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

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

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    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

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

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    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

    Transmissão de video melhorada com recurso a SDN em ambientes baseados em cloud

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    The great technological development of informatics has opened the way for provisioning various services and new online-based entertainment services, which have expanded significantly after the increase in social media applications and the number of users. This significant expansion has posed an additional challenge to Internet Service Providers (ISP)s in terms of management for network, equipment and the efficiency of service delivery. New notions and techniques have been developed to offer innovative solutions such as SDN for network management, virtualization for optimal resource utilization and others like cloud computing and network function virtualization. This dissertation aims to manage live video streaming in the network automatically by adding a design architecture to the virtual network environment that helps to filter video packets from the remaining ones into a certain tunnel and this tunnel will be handled as a higher priority to be able to provide better service for customers. With the dedicated architecture, side by side, a monitoring application integrated into the system was used to detect the video packets and notify the SDN server to the existence of the video through the networkOs grandes avanços tecnológicos em informática abriram o caminho para o fornecimento de vários serviços e novos aplicações de entretenimento baseadas na web, que expandiram significativamente com a explosão no número de aplicações e utilizadores das redes sociais. Esta expansão significativa colocou desafios adicionais aos fornecedores de serviços de rede, em termos de gestão de rede, equipamento e a eficácia do fornecimento de serviços. Novas noções e técnicas foram desenvolvidas para oferecer soluções inovadoras, tais como redes definidas por software (SDN) para a gestão de rede, virtualização para a optimização da utilização dos recursos e outros, tais como a computação em nuvem e as funções de rede virtualizadas. Esta dissertação pretende gerir automaticamente a emissão de vídeo ao vivo na rede, através da adição de uma arquitetura ao ambiente de rede virtualizado, que auxilie a filtragem de pacotes de vídeo dos do restante tráfego, para um túnel específico, que será gerido com uma prioridade maior, capaz de fornecer melhor serviço aos clientes. Além do desenho da arquitectura, scripts de Python foram usados para detectar os pacotes de vídeo e injetar novas regras no controlador SDN que monitoriza o tráfego ao longo da rede.Mestrado em Engenharia de Computadores e Telemátic

    Designing and prototyping WebRTC and IMS integration using open source tools

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    WebRTC, or Web Real-time Communications, is a collection of web standards that detail the mechanisms, architectures and protocols that work together to deliver real-time multimedia services to the web browser. It represents a significant shift from the historical approach of using browser plugins, which over time, have proven cumbersome and problematic. Furthermore, it adopts various Internet standards in areas such as identity management, peer-to-peer connectivity, data exchange and media encoding, to provide a system that is truly open and interoperable. Given that WebRTC enables the delivery of multimedia content to any Internet Protocol (IP)-enabled device capable of hosting a web browser, this technology could potentially be used and deployed over millions of smartphones, tablets and personal computers worldwide. This service and device convergence remains an important goal of telecommunication network operators who seek to enable it through a converged network that is based on the IP Multimedia Subsystem (IMS). IMS is an IP-based subsystem that sits at the core of a modern telecommunication network and acts as the main routing substrate for media services and applications such as those that WebRTC realises. The combination of WebRTC and IMS represents an attractive coupling, and as such, a protracted investigation could help to answer important questions around the technical challenges that are involved in their integration, and the merits of various design alternatives that present themselves. This thesis is the result of such an investigation and culminates in the presentation of a detailed architectural model that is validated with a prototypical implementation in an open source testbed. The model is built on six requirements which emerge from an analysis of the literature, including previous interventions in IMS networks and a key technical report on design alternatives. Furthermore, this thesis argues that the client architecture requires support for web-oriented signalling, identity and call handling techniques leading to a potential for IMS networks to natively support these techniques as operator networks continue to grow and develop. The proposed model advocates the use of SIP over WebSockets for signalling and DTLS-SRTP for media to enable one-to-one communication and can be extended through additional functions resulting in a modular architecture. The model was implemented using open source tools which were assembled to create an experimental network testbed, and tests were conducted demonstrating successful cross domain communications under various conditions. The thesis has a strong focus on enabling ordinary software developers to assemble a prototypical network such as the one that was assembled and aims to enable experimentation in application use cases for integrated environments

    Enterprise WebRTC Powered by Browser Extensions

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    ABSTRACT We use browser extensions to solve two important issues in adopting WebRTC (Web Real-Time Communications) in enterprises: how to integrate WebRTC-centric communication with existing systems such as corporate directories, communication infrastructure and intranet websites, and how to traverse media paths across enterprise firewalls. Vclick is a simple and easy to use web-based video collaboration application that enables click-to-call from other webpages. SecureEdge is a network border traversal system for policy and security enforcement, and consists of a secure media relay that sits at the network border or in the cloud. A browser extension in the enterprise user's device transparently injects this media relay in every WebRTC media path needing to traverse the enterprise network edge to enable authenticated border traversal without help from the websites hosting the WebRTC pages. We attempt to generically support WebRTC in enterprises on a variety of application scenarios instead of creating another fragmented communication island. The challenges faced and techniques used in our proof-of-concepts are likely extensible to other enterprise WebRTC scenarios using the emerging HTML5 technologies
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