17 research outputs found

    Architecture and Protocol to Optimize Videoconference in Wireless Networks

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    [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

    DEVELOPMENT ANALYSIS OF MULTIPOINT VIDEO CONFERENCING IN EDUCATION

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    In this modern era, people can do meetings by doing video conferencing. By doing video conferencing we can reach our colleagues who are geographically far from us in real time, as if we are close to them through video and audio visual. it's just that sometimes the capacity of video conferencing tools becomes an obstacle in doing conference simultaneously for multiple locations. Video conferencing between three or more locations is possible through the Multipoint Control Unit (MCU) system. This research was conducted at one of the educational institutions which had several branches spread in various locations and each location had a video conference tool. MCU system that was used previously is the sx20 telepresence which can only accommodate 3 video conference locations simultaneously. from this problem this educational institution developed a new MCU system using Acano. By using this Acano system can support HD quality video so that video and audio can be clearly displayed.  Di era modern ini, orang dapat melakukan pertemuan dengan melakukan konferensi video. Dengan melakukan konferensi video, kita dapat menjangkau kolega kita yang secara geografis jauh dari kita secara real time, seolah-olah kita dekat dengan mereka melalui video dan audio visual. hanya saja terkadang kapasitas alat konferensi video menjadi kendala dalam melakukan konferensi secara bersamaan untuk beberapa lokasi. Konferensi video antara tiga lokasi atau lebih dimungkinkan melalui sistem Multipoint Control Unit (MCU). Penelitian ini dilakukan di salah satu lembaga pendidikan yang memiliki beberapa cabang yang tersebar di berbagai lokasi dan setiap lokasi memiliki alat konferensi video. Sistem MCU yang digunakan sebelumnya adalah telepresence sx20 yang hanya dapat menampung 3 lokasi konferensi video secara bersamaan. dari masalah ini lembaga pendidikan ini mengembangkan sistem MCU baru menggunakan Acano. Dengan menggunakan sistem Acano ini dapat mendukung video berkualitas HD sehingga video dan audio dapat ditampilkan dengan jelas

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

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    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

    Implementation and analysis of low latency video-conferencing through edge cloud computing

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    Edge cloud computing seems to be a key enabler of 5G networks which essentially brings the servers as close to the users as possible. Among all the benefits this tendency can provide, this master thesis focuses on the advantages in terms of reduction of the latency. First of all, an Edge network model that combines this paradigm with Software-defined Networks (SDN) is presented so as to provide an example of a potential production scenario. Then, a videoconference application is chosen as a particular case study of latency-sensitive and bandwidth exhaustive application and the traffic that it generates is inspected. Thanks to this analysis, a methodology to compute the latency can be proposed which is used during the test runs afterwards. Lastly, a testbed analogous to the model previously presented showcases the benefits of this approach. The results prove the improvement in the quality of the videoconference by means of a noticeable reduction of the latency when the servers are on the edge. Moreover, it is demonstrated the feasibility of providing a dynamic environment where the server can be live migrated. For the sake of providing a complete quality overview, the impact of the available bandwidth and packet loss is evaluated as well

    COMPARISON BEST VIDEO CONFERENCE FOR LEARNING AND TEACHING ACTIVITIES USING ANALYTIC HIERARHICAL PROCESS

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    During the pandemic, almost all industries have been disrupted, including the education industry. To support the sustainability of the education industry, many institutions use various video conferencing platforms. There are six aspects that need to be considered in choosing a video conference platform: Features provided, Ease of use, security level, bandwidth usage, platform stability and the ability to accommodate the number of participants in a conference room. This study shows how to prioritize these aspects in choosing a video conferencing platform carried out by educational institutions in Indonesia. The method used in this research is the Analytical Hierarchical Process (AHP). And the results of this study show the order of aspects in choosing a video conferencing platform for teaching and learning needs

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

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    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

    Manajemen Mutu Jaringan Internet di Masa Work From Home

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    This study aims to analyze the quality of the internet network during the Work From Home (WFH) period at cellular company. The methodology used in this research is pareto chart, fishbone analysis and comparative test analysis. This research was conducted in the period from December 2019 to November 2020. The variable used in this research is number of complaint, speed, jitter, packet loss and delay. The results of the analysis showed that there was an increase in the number of complaints after the WFH period. In addition, the comparative test showed that packet-loss, delay, jitter and complaints each showed a significant difference between before and after WFH. Otherwise, speed doesn’t showed a significant difference between before and after WFH. The implication of this research is cellular company must focus on packet loss, delay, jitter and complaints after WFH because it will affected on quality of internet connection

    Inter-domain interoperability framework based on WebRTC

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    Nowadays, the communications paradigm is changing with the convergence of communication services to a model based on IP networks. Applications such as messaging or voice over IP are increasing its popularity and Communication Service Providers are focusing on offering this kind of services. Moreover, Web Real Time Communication (WebRTC) has emerged as a technology that eases the creation of web applications featuring Real-Time Communications over IP networks without the need to develop and install any plug-in. It lacks of specifications in the control plane, leaving the possibility to use WebRTC over tailored web signalling solutions or legacy networks such as IP Multimedia Subsystem (IMS). This technology brings a wide range of possibilities for web developers, but Communication Service Providers are adviced to develop solutions based on the WebRTC technology as described in the Eurescom Study P2252. The lack of WebRTC specifications on the signalling platform together with the threats and opportunities that this technology represents for Communication Service Providers, makes evident the need of research on interoperability solutions for the different kind of signalling implementations and experimentation on the best way for Communication Service Providers to obtain the maximum benefit from WebRTC technology. The main goal of this thesis is precisely to develop a WebRTC interoperability framework and perform experiments on whether the Communication Service Providers should use their existing IMS solutions or develop tailored web signalling platforms for WebRTC deployments. In particular, the work developed in this thesis was completed under the framework of the Webrtc interOperability tested in coNtradictive DEployment scenaRios (WONDER) experimentation for the OpenLab project. OpenLab is a Large-scale integrating project (IP) and is part of the European Union Framework Programme 7 for Research and Development (FP7) addressing the work programme topic Future Internet Research and Experimentation.Actualmente, el paradigma de comunicaciones está cambiando gracias a la convergencia de los servicios de comunicaciones hacia un modelo basado en redes IP. Aplicaciones tales como la mensajería y la voz sobre IP están creciendo en popularidad mientras los proveedores de servicios de comunicaciones se centran en ofrecer este tipo de servicios basados en redes IP. Por otra parte, la tecnología WebRTC ha surgido para facilitar la creación de aplicaciones web que incluyan comunicaciones en tiempo real sobre redes IP sin la necesidad de desarrollar o instalar ningún complemento. Esta tecnología no especifica los protocolos o sistemas a utilizar en el plano de control, dejando a los desarrolladores la posibilidad de usar WebRTC sobre soluciones de señalizaci on web específicas o utilizar las redes de señalización existentes, tales como IMS. WebRTC abre un gran abanico de posibilidades a los desarrolladores web, aunque también se recomienda a los proveedores de servicios de comunicaciones que desarrollen soluciones basadas en WebRTC como se describe en el estudio P2252 de Eurescom. La falta de especificaciones en el plano de señalización junto a las oportunidades y amenazas que WebRTC representa para los proveedores de servicios de comunicaciones, hacen evidente la necesidad de investigar soluciones de interoperabilidad para las distintas implementaciones de las plataformas de señalización y de experimentar c omo los proveedores de servicios de comunicaciones pueden obtener el máximo provecho de la tecnología WebRTC. El objetivo principal de este Proyecto Fin de Carrera es desarrollar un marco de interoperabilidad para WebRTC y realizar experimentos que permitan determinar bajo que condiciones los proveedores de servicios de comunicaciones deben utilizar las plataformas de se~nalizaci on existentes (en este caso IMS) o desarrollar plataformas de señalización a medida basadas en tecnologías web para sus despliegues de WebRTC. En particular, el trabajo realizado en este Proyecto Fin de Carrera se llevó a cabo bajo el marco del proyecto WONDER para el programa OpenLab. OpenLab es un proyecto de integración a gran escala en el cual se desarrollan investigaciones y experimentos en el ámbito del futuro Internet y que forma parte del programa FP7 de la Unión Europea.Ingeniería de Telecomunicació

    Design and development of a software architecture for seamless vertical handover in mobile communications

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    In this work I firstly present an overview on current wireless technology and network mobility focusing on challenges and issues which arise when mobile nodes migrate among different access networks, while employing real-time communications and services. In literature many solutions propose different methods and architectures to enhance vertical handover, the process of transferring a network communication between two technologically different points of attachment. After an extensive review of such solutions this document describes my personal implementation of a fast vertical handover mechanism for Android smartphones. I also performed a reliability and performance comparison between the current Android system and my enhanced architecture which have both been tested in a scenario where vertical handover was taking place between WiFi and cellular network while the mobile node was using video streaming services. Results show the approach of my implementation to be promising, encouraging future works, some of which are suggested at the end of this dissertation together with concluding remarks
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