Les services de conversation Web en temps réel permettent aux utilisateurs d'avoir des appels audio et vidéo et de transférer directement des données sur Internet. Les opérateurs OTT (OTT) tels que Google, Skype et WhatsApp proposent des services de communication économiques avec des fonctionnalités de conversation évoluées. Avec l'introduction de la norme de Web Real Time Communication (WebRTC), n'importe quelle page Web peut désormais offrir des services d'appel. WebRTC est utilisé comme technologie sous-jacente pour déployer de nouvelles plateformes de communication centrées sur le Web. Ces plates-formes visent à offrir de nouvelles méthodes modernes de contact et de communication sur le web. Contrairement aux réseaux de télécommunication traditionnels, les identités sur le Web sont basées sur des profils d'utilisateur et des informations d'identification auto-affirmés. Par conséquent, les opérateurs Web sont incapables d'assurer la fiabilité de leurs abonnés. Les services de communication Web restent exposés à des menaces dans lesquelles le contexte social entre les parties communicantes est manipulé. Un attaquant se définit comme une entité de confiance pour transmettre de fausses informations à l'utilisateur ciblé. Les menaces typiques contre le contexte social comprennent la fausse représentation d'identité, l’hameçonnage, le spam et la distribution illégale de contenu. Afin d'assurer la sécurité sur les services de communication Web, la confiance entre les parties communicantes doit être établie. La première étape consiste à permettre aux utilisateurs d'identifier leurs participants communicants afin de savoir avec qui ils parlent. Cependant, l'authentification seule ne peut garantir la fiabilité d'un appelant. De nouvelles méthodes d'estimation de la réputation de l'appelant devraient également être intégrées dans les services d'appel Web. Par conséquent, dans cette thèse, nous présentons un nouveau cadre de confiance qui fournit des informations sur la fiabilité des appelants dans les réseaux de communication Web. Notre approche est organisée en quatre parties. Premièrement, nous décrivons la notion de confiance dans la communication web en temps réel. Un modèle de confiance est présenté pour identifier les relations de confiance nécessaires entre les entités d'un système de communication. Les paramètres requis pour calculer la confiance dans les services de communication Web sont officiellement introduits. Deuxièmement, nous montrons comment les protocoles Single-Sign-On (SSO) peuvent être utilisés pour authentifier les utilisateurs d'une manière Peer-to-Peer (P2P) sans dépendre de leur fournisseur de service. Nous présentons une comparaison entre trois protocoles d'authentification appropriés (OAuth, BrowserID, OpenID Connect). La comparaison montre que OpenID Connect est le meilleur candidat en termes de confidentialité des utilisateurs. Troisièmement, un modèle de calcul de confiance est proposé pour mesurer la fiabilité des appelants dans un réseau de communication. La légitimité et l'authenticité d'un appelant sont calculées à l'aide de recommandations, tandis que la popularité d'un appelant est estimée en utilisant son comportement de communication. Un abonné d'un service de communication sera capable de visualiser la confiance calculée d'autres membres avant d'initier ou d'accepter une demande d'appel. Enfin, la réputation d'un appelant est utilisée pour lutter contre les appels nuisibles générés sur les réseaux de communication. Les appels de nuisance sont décrits comme des appels de spam non sollicités en masse générés sur un réseau de communication à des fins de marketing et de tromperie. Les enregistrements de données d'appel et les commentaires reçus par les parties communicantes sont utilisés pour déterminer la réputation de l'appelant. La réputation évaluée est utilisée pour différencier les spammeurs et les appelants légitimes du réseau / Real-time web conversational services allow users to have audio and video calls over the Internet. Over-The-Top operators such as Google and Facebook offer cost-effective communication services with advanced conversational features. With the introduction of WebRTC standard, any website or web application can now have built-in communication capabilities. WebRTC technology is expected to boost Voice-Over-IP by making it more robust, flexible and accessible. Telco operators also intend to use the underlying technology to offer communication services to their subscribers over the web. Emerging web-centric communication platforms aims to offer modern methods of contacting and communicating over the web. However, web operators are unable to ensure the trustworthiness of their subscribers, since identities are based on self-asserted user profiles and credentials. Thus, they remain exposed to many social threats in which the context between communicating parties is manipulated. An attacker usually misrepresents himself to convey false information to the targeted victim. Typical social threats include phishing, spam, fraudulent telemarketing and unlawful content distribution. To ensure user security over communication networks, trust between communicating parties needs to be established. Communicating participants should be able to verify each other’s identity to be sure of whom they are talking to. However, authentication alone cannot guarantee the trustworthiness of a caller. New methods of estimating caller’s reputation should also be built in web calling services. In this thesis, we present a novel trust framework that provides information about the trustworthiness of callers in web communication networks. Our approach is organized in four parts. Firstly, we describe the notion of trust in real-time web communication services. A trust model approach is presented to formally introduce the trust computation parameters and relationships in a communication system. Secondly, we detail the mechanism of identity provisioning that allows communicating participants to verify each other’s identity in a Peer-to-Peer fashion. The choice of authentication protocol highly impacts user privacy. We showed how OpenID Connect used for Single-Sign-On authentication purposes can be effectively used for provisioning identities while preserving user privacy. Thirdly, a trust computational model is proposed to measure the trustworthiness of callers in a communication network. The legitimacy and genuineness of a caller’s identity is computed using recommendations from members of the network. On the other hand, the popularity of a caller is estimated by analyzing its behavior in the network. Each subscriber will be able to visualize the computed trust of other members before initiating or accepting a call request. Lastly, the reputation of a caller is used to combat nuisance calls generated over communication networks. Nuisance calls are described as unsolicited bulk spam phone calls generated for marketing and deceptive purposes. Caller’s reputation is computed using the diversity of outgoing calls, call duration, recommendations from called participants, reciprocity and repetitive nature of calls. The reputation is used to differentiate between legitimate and nuisance calls generated over the network

CareLigo is a medical technology company that supplies heart failure patients with a home-based care solution called OPTILOGG. OPTILOGG helps patients to keep track of their symptoms, provides medication instructions and educates them about their illness. CareLigo requested an expansion of OPTILOGG which would allow patients to talk to care providers via video communication. This thesis describes the creation of this video communication solution and how this can be done in the best way for both patients and caregivers. Based on literature studies on human-computer interaction, user interface (UI) and user experience (UX) design, a standalone Android application was developed for care providers as well as an extension in OPTILOGG taking into account that the users of OPTILOGG are often elderly with multimorbidity. Three accessibility aids were added to the extension of the OPTILOGG Android application in addition to the video solution. The first helping addition was a touch area expansion for buttons. The second aid was a screen reader feature that vocally describes objects the user clicks on. The third tool was a speech recognition feature that allowed patients to navigate in OPTILOGG with voice. The video communication between the standalone care provider application and OPTILOGG was based on WebRTC and was developed using a software development kit from a cloud communications provider called Sinch. / CareLigo är ett medicintekniskt företag som tillhandahåller hjärtsviktspatienter med en hembaserad vårdlösning kallad OPTILOGG. OPTILOGG hjälper patienter att hålla koll på deras symptom, ger medicineringsanvisningar och utbildar dem om sin sjukdom. CareLigo eftersökte en utökning av OPTILOGG vilket skulle ge patienterna möjlighet att prata med vårdgivare via videokommunikation. Detta examensarbete beskriver skapandet av denna videokommunikationslösning och hur detta kan göras på bästa sätt för både patienter och vårdgivare. Utifrån litteraturstudier gällande människa-dator-interaktion samt användargränssnitts (UI)-och interaktionsdesign (UX) utvecklades en fristående Androidapplikation för vårdgivare och en utökning i OPTILOGG med hänsyn tagen till att användarna av OPTILOGG ofta är multisjuka och äldre. Tre tillgänglighetshjälpmedel tillades i utökningen av OPTILOGG. Det första hjälpmedlet var en förstoring av klickareor runt knappar. Den andra tillgänglighetsåtgärden var en skärmläsarfunktion som beskriver objekt som användaren klickar på. Det tredje verktyget var en funktion för taligenkänning som gjorde det möjligt för patienter att navigera i OPTILOGG med röst. Videokommunikationen mellan den fristående vårdgivarapplikationen och OPTILOGG baserades på WebRTC och utvecklades med hjälp av ett utvecklingsverktyg från en molnkommunikationsle-verantör som heter Sinch.

Web browsers are being used for network diagnostics. Users commonly verify their Internet speed by using a website, Bredbandskollen.se or speedtest.net for example. These test often need third party software, Flash or Java applets. This thesis aims at prototyping an application that pushes the boundaries of what the modern web browser is capable of producing regarding network measurements, without any third party software. The contributions of this thesis are a set of suggested tests that the modern browser should be capable of performing without third party software. These tests can potentially replace some of network technicians dedicated test equipment with web browser capable deceives such as mobile phones or laptops. There exist both TCP and UDP tests that can be combined for verifying some Quality of Service (QoS) metrics. The TCP tests can saturate a gigabit connection and is partially compliant with RFC 6349, which means the traditional Internet speed test sites can obtain more metrics from a gigabit throughput test then they do today.

Real-time streaming is becoming popular nowadays on the internet. WebRTC is a promising web technology that enables media stream transmission between two browsers without other third-party plugins. However, traditional WebRTC applications can only establish peer-to-peer (P2P) communications, which cannot be directly used in one-to-more streaming scenarios such as a multi-party video conference. This thesis project presents a development of a multi-cloud simulation framework to implement software multicast of WebRTC streams to enable oneto-more real-time streaming. The framework can generate a cloud network topology with a simple script, and provides flexible ways to model the network links with parameters such as bandwidth, packet loss, and link latency. By using selective forwarding units (SFUs), a stream publisher only needs to send a single copy of the data to the cloud, the data packets are duplicated and forwarded to multiple subscribers. In addition, three resource allocation algorithms are included in the framework to determine the data center for a task. To evaluate the performance, this framework enables people to monitor the throughputs and running streams on the data centers during the experiments. We develop benchmark applications as substitutes for real WebRTC traffic. These applications can generate UDP stream packets with the same dimension of WebRTC packets and provide the customization of stream parameters. In this report, we present a comparison of the stream performances under different allocation algorithms. Finally, as an outcome of this project, we design an integrated functional test to simulate a real-world scenario. The result shows that the framework is able to work well on complex real cases and simulate most of the multi-cloud networks.

The behaviour of WebRTC when the real-time communication with audio is done over a non-optimal network is investigated in this thesis. Different methods for collecting and analyzing data from an online survey are considered. A test environment was developed from which two online surveys would be conducted, where the outgoing packets had various interferences applied to them by the server. This was made in order to be able to simulate a non-optimal network e g WiFi. The participants are told to listen to various audio sequences and are asked to rate the quality as they perceive it. Although considered, video was not used in the surveys, as it would have increased the complexity of the surveys and increasing the risk of having the participants rejecting the surveys. Two independent surveys were conducted. The first survey utilized WebRTC for sending the audio, this was compared to the second survey which instead used Icecast. The result showed that WebRTC behaves well when there was only one type of interference added. Compared to Icecast it had lower performance. However, this could be contributed to the fact that two independent groups were used and the surveys had low participation rates. The surveys proved the feasibility of conducting online surveys for measuring perceived quality, although the participation rate was extremely low (2.8%), something that has to be considered when performing online surveys.

Möjligheterna på webben blir hela tiden större. Teknologier för webbaserade spel börjar dyka upp som tillåter animationer och realtidskommunikationskrav till den standard som spel gjorda för desktops idag har. I detta arbete testas WebRTC och Websockets mot varandra för att ta fram underlag för vilken lösning som presterar bäst i olika scenarion. Ett experiment utfördes och resultatet pekade på att WebRTC klarade av fler scenarion bättre än Websockets till en nivå som relaterad forskningslitteratur pekar på är acceptabla nivåer av nätverksfördröjning för att uppnå kraven för realtidskommunikation. Experimentet gick ut på att simulera nätverksfördröjning vid olika antal samtida användare i ett webbaserat spel. För framtida arbete spekuleras det kring användning av andra webbläsare, enheter och kopplingar till Internet of all things.

Background:  Cloud computing technology has come a long way with various technological advancements in the past few years. It has been accelerated with the evolution of various virtualization technologies. Currently almost every social platform and small-scale applications look towards cloud to deploy their services successfully and provide maximum satisfaction to their end-user. Thus, virtualizing their services becomes utmost important to deploy and develop their applications. This alone emphasizes the importance of Docker containers in the development world. Docker containers right now are playing a very important role in the field of cloud computing. Since Multimedia plays a huge role in our day to day lives and most people crave for faster and efficient responses, it is essential to develop our applications with better Real time communication capabilities. Thus, we are determining which container orchestration tool serves best for Real time communication applications.  A multimedia application is developed and deployed using WebRTC based Kurento media server and the performance of the server is measured when the application is deployed. We have chosen Kubernetes and Docker Swarm as container platforms for this thesis. The Servers and Clients are virtualized and metrics such as CPU Utilization, Network Traffic, Container overhead, Memory Utilization are measured. These metrics provide the performance overhead in different scenarios for each orchestration technology. This will be helpful to analyze and understand the effect of Kurento server on these technologies. Thus, the results are expected to determine which orchestration technology serves best for RTC applications. Objectives: The objectives of this project are:  • To implement WebRTC based Kurento server in a container orchestrated environment.  • To extract performance metrics such as Network Traffic, CPU and Memory Utilization while server is running.  • To compare WebRTC based Kurento server in Kubernetes and Docker Swarm.   Method: Kubernetes and Docker Swarm environments are setup and then docker images with video conferencing application(One-to-One call and One-to-Many call) using Kurento media server is deployed in them. Once either of the applications is running, experiments are performed for analyzing performance metrics like CPU Utilization, Memory Utilization, Network Traffic and overhead using monitoring tool, Prometheus. Along with Kubernetes and Docker Swarm, Kurento server is also deployed on a stand-alone container to estimate the performance overhead. Later, statistical analysis(ANOVA and differences of Standard error) is done over these metrics and conclusions are drawn.  Results: Based on the performed experiments and the extracted metrics, for One-to-One call application, Kubernetes showed better resource utilization for CPU and Network Traffic while it consumed more memory over Docker Swarm. Similar behaviour is observed for One-to-Many application. When application is scaled, the percent of resource utilization increase in Kubernetes is higher when compared to Docker Swarm, but overall resource utilization of Kubernetes is much lower than that of Docker Swarm.  Conclusions: WebRTC based Kurento media server is investigated in  Kubernetes and Docker Swarm. From the detailed analysis there is significant overhead in Docker Swarm than in Kubernetes for CPU Utilization and Network Traffic. For Memory Utilization, this is opposite. Packet Loss resulted in 0 percent as network transfer is within the same network . By considering all the metrics and providing evidence that numbers obtained in this thesis are statistically significant and not by fluctuations(ANOVA and post-hoc analysis), we can better recommend Kubernetes over Docker Swarm for Web based Real Time Communication.   However, not all applications need the complex deployment, scheduling, and scaling services (or the overhead) that Kubernetes offers. But to meet the increasing demand for seamless Real time communications, and to suffice user requirements, the overheard offered by it is acceptable.

The goal of this thesis is to design a system which would allow video data streaming from a mobile device and real time playback using a standard web browser. The technological background and the implementation platform are both part of this thesis. Web Real Time Communications (WebRTC) technology was used for acquiring multimedia data on mobile device. This technology is natively supported in the latest major web browsers and in WebView component (Android version 5.0 and above). Sending push notifications from a server to a mobile device to start the streaming is done with Google Cloud Messaging technology. The resultant system allows a user to start the application on mobile device with easy web browser access. This starts the multimedia stream from device, which can be parametrized and secured by password. The benefit of this thesis is the overview of WebRTC technology and its demonstration. The IP camera implementation shows how easy it is to use the WebRTC in real applications.

The aim of this thesis is to investigate existing design patterns for WebRTC applications in order to achieve a scalable, performant and efficient WebRTC application that keeps streams unique. Further, this thesis shows how these can be implemented using JavaScript technologies. Through a literature study, we conclude that the design patterns full mesh using a signaling server and star topology with a media server that relays streams, called Selective Forwarding Unit (SFU). Both these design patterns have quality attributes that are desirable. We propose an approach of combining these patterns in the same application in order to achieve a scalable application that can fit a broad spectrum of use cases while being efficient. As full mesh is performant and cost-effective in comparison to an SFU but does not scale well with increasing number of participants, we investigate ways to optimize a full mesh session to use it as long as possible before converting a session to using an SFU. We came up with a way to optimize a full mesh session by limiting the bandwidth used for the media streams which reduces the CPU usage for the clients. The proposed approach of combining full mesh and an SFU is implemented based on a previous WebRTC application and a high-level description of how that was achieved is included in this thesis. We perform an experiment where we measure the client’s CPU usage using the above-mentioned approaches in order to reinforce our findings. The result show that limiting the bandwidth of media streams can increase the possible number of participants in a full mesh session and that it is possible to transfer an ongoing session from full mesh to an SFU and back again. We conclude that combining these patterns in the same application is a viable strategy when creating a WebRTC application.

This thesis presents how a solution to fetch and stream a video feed from the astrovisualization software OpenSpace to a web page can be designed. The streaming protocol that was used was WebRTC. Three different methods for fetching data and creating a video feed were investigated: WebRTC, Spout, and GStreamer. Through user tests, the GStreamer method was determined to be the best option for the streaming solution. / <p>Examensarbetet är utfört vid Institutionen för teknik och naturvetenskap (ITN) vid Tekniska fakulteten, Linköpings universitet</p>