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abstract: Interactive remote e-learning is one of the youngest and most popular methods that is used in today's teaching method. WebRTC, on the other hand, has become the popular concept and method in real time communication. Unlike the old fashioned Adobe Flash, user will communicate directly to each other rather than calling server as the middle man. The world is changing from plug-in to web-browser. However, the WebRTC have not been widely used for school education. By taking into consideration of the WebRTC solution for data transferring, we propose a new Cloud based interactive multimedia which enables virtual lab learning environment. Three modules were proposed along with an efficient solution for achieving optimized network bandwidth. The One-to-Many communication was introduced in the video conferencing and scalability was tested for the application. The key technical contribution is to establish a sufficient system that designed to utilize the WebRTC in its best way in educational world in the Vlab platform and reduces the tool cost and improves online learning experience. / Dissertation/Thesis / Masters Thesis Computer Science 2014

El texto completo de este trabajo no está disponible en el Repositorio Académico UPC por restricciones de la casa editorial donde ha sido publicado. / This research paper proposes a low-cost telepresence technological model focused on primary education. Its aim is to give students a new resource/communication channel for classes, which would be used when they cannot attend school due to health problems that do not affect their learning process. This solution seeks students to not be passive listeners during a session, but that they interact with their classmates and teachers during class. To validate the model, a telepresence platform based on WebRTC was developed. It was tested in three schools in different geographical areas belonging to socioeconomic sector C, collecting data from the students who tested the tool, as well as from classmates, teachers, and parents. / Revisión por pares

In this thesis an audio and video streaming system is implemented for its use in rescue robots. WebRTC technology is used in order to stream in real time. Implemented in an architecture based on a Web server, two pages running WebRTC and a TURN1-STUN2 server, the system has been tested in terms of CPU and bandwidth utilization. Measurements show that when WebRTC is run in an Intel Core i3, less than 10% of CPU is used, whereas on smaller tablets the performance is not enough for running the application with the desired quality of service.

The topic of this work is verification of support WebRTC communication through selected Open Source PBX. This work examine demands for WebRTC communications and describes configuration of branch centers for this type of communication. In the theoretical part is reader acquainted with the term WebRTC and with protocols related to this kind of communications. The purpose of this part of the work is to bring the reader closer look to the principles of functioning to ensuring support for this kind of communications. This is also connected with Description of basic interfaces of WebRTC applications. Further the reader finds the configuration of the selected Open Source PBX so that they can make audio-video call between WebRTC clients. This section is divided into three subchapters, each of it deals with the same problems for one of the aforementioned PBX. At the end of each chapter where the PBX PBX is configured step-by-step, test calls are made. These calls are captured by the Wireshark packet analyzer and serve as a demonstration of the WebRTC configuration functionality. At the end of this section, PBXs are compared against each other about WebRTC support. Practical part is dealing with laboratory task for students which are studying subject telecommunication and information systems. In the task students will be configuring WebRTC for PBX Asterisk. The task contains brief description of WebRTC and comments for all steps for configuration. All steps and facts are demonstrated by exemplary configuration files.

Les réseaux mobiles 4G basés sur la norme LTE (Long Term Evolution), sont des réseaux tout IP. Les différents problèmes de transport IP comme le retard, la gigue et la perte despaquets peuvent fortement dégrader la qualité des communications temps réel telles que la téléphonie. Les opérateurs ont mis en oeuvre des mécanismes d’optimisation du transport de la voix dans le réseau afin d'améliorer la qualité perçue. Cependant, les algorithmes propriétaires de gestion de la qualité dans les terminaux ne sont pas spécifiés dans les standards. Dans ce contexte, nous nous intéressons aux mécanismes d'adaptation de média, intégrés dans les terminaux afin d'améliorer la qualité d’expérience (QoE). En particulier, nous évaluons de manière expérimentale des métriques QoE de la voix sur LTE (VoLTE) en utilisant une méthode de test standardisée. Nous proposons d’améliorer la méthode de test et discutons la manière dont cette méthode peut être étendue pour évaluer les performances du buffer de gigue. Nous évaluons également de manière expérimentale la qualité de WebRTC dans différentes conditions radios en utilisant un réseau réel. Nous évaluons l'impact du buffer de gigue et de la variation du débit sur la qualité mesurée. Pour améliorer la robustesse des codecs contre la perte de paquets, nous proposons d’utiliser une redondance simple au niveau applicatif. Nous implémentons cette redondance pour le codec EVS (Enhanced Voice Service) et nous évaluons ses performances. Enfin, nous proposons un protocole de signalisation qui permet d’envoyer des requêtes de redondance au cours d’une communication afin d’activer ou désactiver celle-ci dynamiquement. / Fourth-generation mobile networks, based on the Long Term Evolution (LTE) standard, are all- IP networks. Thus, mobile telephony providers are facing new types of quality degradations related to the voice packet transport over IP network such as delay, jitter and packet loss. These factors can heavily degrade voice communications quality. The real-time constraint of such services makes them highly sensitive to delay and loss. Network providers have implemented several network optimizations for voice transport to enhance perceived quality. However, the proprietary quality management algorithms implemented in terminals are left unspecified in the standards. In this context, we are interested in media adaptation mechanisms integrated in terminals to enhance the overall Quality of Experience (QoE). In particular, we experimentally evaluate Voice over LTE (VoLTE) QoE metrics such as delay and Mean Opinion Score (MOS) sing a standardized test method. We propose some enhancements to the actual test method and discuss how this method can be extended to evaluate de-jitter buffer performance. We also experimentally evaluate WebRTC voice quality in different radio conditions using a realLTE test network. We evaluate the impact of jitter buffer and bit rate variations on the measured quality. To enhance voice codec robustness against packet loss, we propose a simple application layer redundancy. We implemented it for the Enhanced Voice Service (EVS) codec and evaluate it. Finally, we propose a signaling protocol that allows sending redundancy requests during a call to dynamically activate or deactivate the redundancy mechanism.

This thesis project explores two well-known real-time web technologies: WebSocket and WebRTC. It explores the use of a mobile phone as a gateway to connect wireless devices with short range of radio links to the Internet in order to foster an Internet of Things (IoT). This thesis project aims to solve the problem of how to collect real-time data from an IoT device, using the Earl toolkit. With this thesis project an Earl device is able to send real-time data to Internet connected devices and to other Earl devices via a mobile phone acting as a gateway. This thesis project facilitates the use of Earl in design projects for IoT devices. IoT enables communication with many different kinds of “things” such as cars, fridges, refrigerators, light bulbs, etc. The benefits of IoT range from financial savings due to saving energy to monitoring the heart activity of a patient with heart problems. There are many approaches to connect devices in order to create an IoT. One of these approaches is to use a mobile phone as a gateway, i.e., to act as a router, between IoT and the Internet. The WebSocket protocol provides efficient communication sessions between web servers and clients by reducing communication overhead. The WebRTC project aims to provide standards for real-time communications technology. WebRTC is important because it is the first real-time communications standard which is being built into browsers. This thesis evaluates the benefits which these two protocols offer when using a mobile phone as a gateway between an IoT and Internet. This thesis project implemented several test beds, collected data concerning the scalability of the protocols and the latency of traffic passing through the gateway, and presents a numerical analysis of the measurement results. Moreover, an LED module was built as a peripheral for an Earl device. The conclusion of the thesis is that WebSocket and WebRTC can be utilized to connect IoT devices to Internet. / I detta examensarbete utforskas två välkända realtidsteknologier på internet: WebSocket och WebRTC. Det utforskar användandet av en mobiltelefon som gateway för att ansluta trådlösa enheter - med kort räckvidd - till Internet för att skapa ett Internet of Things (IoT). Det här examensarbetet försöker med hjälp av verktyget Earl lösa problemet med hur insamlandet av realtidsdata från en IoT-enhet skall genomföras. I det här examensprojektet kan en Earl-enhet skicka data i realtid till enheter med Internetanslutning, samt till andra Earl-enheter, med hjälp av en mobiltelefon som gateway. Detta projektarbete förenklar användandet av Earl i design-projekt ör IoT-enheter. IoT tillåter kommunikation mellan olika sorters enheter, så som bilar, kyl- och frysskåp, glödlampor etc. Fördelarna med IoT kan vara allt från ekonomiska - tack vare minskad energiförbrukning - till medicinska i form av övervakning av puls hos patienter med hjärtproblem. Det finns många olika tillvägagångssätt för att sammankoppla enheter till ett IoT. Ett av dessa är att använda en mobiltelefon som en gateway, dvs en router mellan IoT och internet. WebSocket-protokollet erbjuder effektiv kommunikation mellan web-servrar och klienter tack vare minskad överflödig dataöverföring. WebRTC-projektet vill erbjuda standarder för realtidskommunikation. WebRTC är viktigt då det är den första sådana standarden som inkluderas i webläsare. Det här examensarbetet utvärderar fördelarna dessa två protokoll erbjuder i det fallet då en mobiltelefon används som gateway mellan ett IoT och Internet. I det här examensprojektet implementerades ett flertal testmiljöer, protokollens skalbarhet och fördröjningen av trafiken genom mobiltelefonen (gateway) undersöktes. Detta presenteras i en numerisk analys av mätresultaten. Dessutom byggdes en LED-modul som tillbehör till en Earl-enhet. Slutsatsen av examensarbetet är att WebSocket och WebRTC kan användas till att ansluta IoT-enheter till Internet.

Context. Introduction of efficient multimedia technologies combined with the spreading of high-speed internet connection all over the world has led to the continuous increase in demand of multimedia services, particularly video and audio. One of the major demands are flexible, interoperable and cost-effective lightweight remote collaboration systems in companies. Web Real Time Communication (WebRTC) is an emerging peer to peer technology that is promising to be the solution to many digital real-time communication challenges. With its fantastic one-to-one communication capabilities, WebRTC supports fast and smooth audio calls, video calls, conferencing, data (media file, document and screen) sharing, gaming and all sorts of messages exchange, all being done straight out of the browser. However, as shown by investigations and interviews supported by Ericsson AB and Semcon AB as party of the MERCO (Mediated Effective Remote Collaboration) international project, many corporate use cases of remote collaboration involve applications beyond the conventional one to one communication. Present videoconferencing systems (telepresence) limits the collaboration flexibility due to their lack of the ability to adapt to system resource usage, hence tend to be too heavy for less powerful devices (laptops, tablets, phones). Moreover, their installation and maintenance costs are too expensive for small companies.  Therefore, new flexible, lightweight and less expensive solutions for remote collaboration need to be developed. Objectives. The main objective of this thesis is to identify technical solutions to address the challenges of resource usage flexibility in WebRTC multi-party remote collaboration systems. Despite concurrent developments of both commercial and free solutions that provide multi-party videoconferencing services using WebRTC, present solutions such as the conventional Multipoint Control Unit (MCU), Selective Forwarding Unit (SFU) and Fully Meshed architectures suffers from issues of excessive resource usage and cannot deliver the acceptable quality of experience in different use cases, particularly the mobile environment. The aim of this thesis is to investigate lightweight technical solutions that can be used to improve the system resource usage in WebRTC multiparty conferencing systems. Through understanding the architectural designs, benchmarking the performance of various technologies used in WebRTC and selecting the most suitable techniques a prototype is developed as a proof of concept. Methods. The first part of the thesis is dedicated to comprehensive study of fundamentals, background information and related works on WebRTC. This gives knowledge of technologies, techniques and performance evaluation metrics which help in making appropriate technical decisions during the experimental development of WebRTC solutions. The second part of the thesis is dedicated to experimental investigation in which two WebRTC signaling technologies (XSockets and NodeJs) are evaluated based on call setup time in WebRTC group call. Two lightweight technical solutions for improving resource usage flexibility (Switching video quality based on speech and using emotions and gestures instead of video) are evaluated based on system resources (CPU, memory, disk and network) and user experience. Results. Based on call setup time of WebRTC multi-party calls, the experimental results indicates that XSockets is a better signaling technology than NodeJs. The two proposed lightweight solutions have shown a remarkable improvement based on systems resource usage. A 15% reduction of CPU usage is observed when using speech controlled video quality switching and further 10% reduction is observed when video is replaced by emotions and gestures. Conclusions. Despite the minimal resource usage achieved by using emotions technique, this solution has usability issues as it cannot detect emotions in poor lighting environment. Consequently, the solution of switching video quality based on speech is chosen for further implementation. Though, this technique can be further improved through using machine learning techniques, the current implementation can significantly reduce the amount CPU, memory, disk and network usage to allow up to 6 participants to join a single conference call while maintain acceptable quality of experience.

In recent years there has been a predominant increase in multimedia services such as live streaming, Video on Demand (VoD), video conferencing, videos for the learning. Streaming of high quality videos has become a challenge for service providers to enhance the user’s watching experience. The service providers cannot guarantee the perceived quality. In order to enhance the user’s expectations, it is also important to estimate the quality of video perceived by the user. There are different video streaming protocols that are used to stream from server to client. In this research, we aren’t focused on the user’s experience. We are mainly focused on the performance behavior of the protocols. In this study, we investigate the performance of the HTTP, RTSP and WebRTC protocols when streaming is carried out for H.265 encoder. The study addresses for the objective assessment of different protocols over VoD streaming at the network and application layers. Packet loss and delay variations are altered at the network layer using network emulator NetEm when streaming from server to client. The metrics at the network layer and application layer are collected and analyzed. The video is streamed from server to a client, the quality of the video is checked by some of the users. The research method has been carried out using an experimental testbed. The metrics such as packet counts at network layer and stream bitrate at application layer are collected for HTTP, RTSP and WebRTC protocols. Variable delays and packet losses are injected into the network to emulate real world. Based on the results obtained, it was found at the application layer that, out of the three protocols, HTTP, RTSP and WebRTC, the stream bitrate of the video transmitted using HTTP was less when compared to the other. Hence, HTTP performs better in the application layer. At the network layer, the packet counts of the video transmitted were collected using TCP port for HTTP and UDP port for RTSP and WebRTC protocols. The performance of HTTP was found to be stable in most of the scenarios. On comparing RTSP and WebRTC, the number of packet counts collected were more in number for RTSP when compared to WebRTC. This is because, the protocol and also the streamer are using more resources to transmit the video. Hence, both the protocols RTSP and WebRTC are performing better relatively.

Vid utveckling av applikationer som har nätverkskrav så är det inte alltid lätt att veta vilken teknik som bör väljas. Realtidskommunikation öppnar upp flera möjligheter till applikationer där höga krav ställs på svarstider. I denna studie så jämförs WebSocket och WebRTC för att ge ett bättre underlag vid val av bakomliggande teknik för att uppnå realtidskommunikation i spel. Två olika spelartefakter utvecklas där den ena stödjer WebSocket och den andra stödjer WebRTC. En användarstudie används sedan för att utvärdera dessa. WebSocket och WebRTC ställs även mot varandra i ett experiment som resulterar i att WebSocket presterar bättre.