Orchestrating Service Migration for Low Power MEC-Enabled IoT Devices

Multi-Access Edge Computing (MEC) is a key enabling technology for Fifth Generation (5G) mobile networks. MEC facilitates distributed cloud computing capabilities and information technology service environment for applications and services at the edges of mobile networks. This architectural modification serves to reduce congestion, latency, and improve the performance of such edge colocated applications and devices. In this paper, we demonstrate how reactive service migration can be orchestrated for low-power MEC-enabled Internet of Things (IoT) devices. Here, we use open-source Kubernetes as container orchestration system. Our demo is based on traditional client-server system from user equipment (UE) over Long Term Evolution (LTE) to the MEC server. As the use case scenario, we post-process live video received over web real-time communication (WebRTC). Next, we integrate orchestration by Kubernetes with S1 handovers, demonstrating MEC-based software defined network (SDN). Now, edge applications may reactively follow the UE within the radio access network (RAN), expediting low-latency. The collected data is used to analyze the benefits of the low-power MEC-enabled IoT device scheme, in which end-to-end (E2E) latency and power requirements of the UE are improved. We further discuss the challenges of implementing such schemes and future research directions therein

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

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

Large-Scale Measurement of Real-Time Communication on the Web

Web Real-Time Communication (WebRTC) is getting wide adoptions across the browsers (Chrome, Firefox, Opera, etc.) and platforms (PC, Android, iOS). It enables application developers to add real-time communications features (text chat, audio/video calls) to web applications using W3C standard JavaScript APIs, and the end users can enjoy real-time multimedia communication experience from the browser without the complication of installing special applications or browser plug-ins. As WebRTC based applications are getting deployed on the Internet by thousands of companies across the globe, it is very important to understand the quality of the real-time communication services provided by these applications. Important performance metrics to be considered include: whether the communication session was properly setup, what are the network delays, packet loss rate, throughput, etc. At Callstats.io, we provide a solution to address the above concerns. By integrating an JavaScript API into WebRTC applications, Callstats.io helps application providers to measure the Quality of Experience (QoE) related metrics on the end user side. This thesis illustrates how this WebRTC performance measurement system is designed and built and we show some statistics derived from the collected data to give some insight into the performance of today’s WebRTC based real-time communication services. According to our measurement, real-time communication over the Internet are generally performing well in terms of latency and loss. The throughput are good for about 30% of the communication sessions

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

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