Syftet med detta examensarbete är att och utvärdera undersöka analysera och utvärdera prestandan hos WebRTC-protokollet, samt att utveckla en webbaserad klient med hjälp av JavaScript för distribuerade system och demonstrera protokollets användbarhet i ett verkligt scenario. Studien inkluderade användning av olika verktyg och bibliotek, såsom Socket.IO, Node.js, Express.js och PeerJS. De viktigaste prestandaindikatorerna som utvärderades var latens/tur- och returtid (RTT), jitter och paketförlust. Implementationen testades både lokalt och på distans. Prestandatestningen av applikationen utfördes med hjälp av webbplatserna "Chrome webrtc-internals" och "TestRTC", vilka erbjöd detaljerade insikter och statistik om WebRTC-prestanda. Resultaten indikerade att WebRTC erbjuder högpresterande och kostnadseffektiv realtidskommunikation som är kompatibel med andra applikationer som stöder protokollet. Protokollet visade sig ha robusta säkerhetsåtgärder, vara kompatibelt med distribuerade system och erbjuda stark prestanda när det gäller latens, jitter och paketförlust. Studien drog slutsatsen att WebRTC, med sin skalbarhet och förmåga att erbjuda kommunikation i realtid, är ett fördelaktigt val för distribuerade system och webbaserade videochattapplikationer. Resultaten uppmanar till ytterligare undersökningar inom områden som end-to-end-kryptering och integration av artificiell intelligens för att förbättra systemets prestanda och säkerhet. / The aim of this thesis is to analyse and evaluate the performance of the WebRTC protocol, develop a web-based client using JavaScript for distributed systems, and demonstrate the utility of the protocol in a real-world scenario. The study involved the use of various tools and libraries, including Socket.IO, Node.js, Express.js, and PeerJS. Key performance indicators evaluated were latency/round-trip time (RTT), jitter, and packet loss. The implementation was tested both locally and remotely. Performance testing of the application was conducted using the "Chrome webrtc-internals" and "TestRTC" websites, which provided detailed insights and statistics on WebRTC performance. The results indicated that WebRTC offers high-performance and cost-effective real-time communication that is compatible with other applications supporting the protocol. The protocol demonstrated robust security measures, compatibility with distributed systems, and strong performance in terms of latency, jitter, and packet loss. The study concluded that WebRTC, with its scalability and ability to provide real-time communication, is a beneficial choice for distributed systems and webbased video chat applications. The findings encourage further investigations in areas such as end-to-end encryption and the integration of artificial intelligence to enhance system performance and security.

Different communication services from delivery of written letters to telephones, voice/video over Internet Protocol(IP), email, Internet chat rooms, and video/audio conferences, immersive communications have evolved over time. A communication system of voice/video over IP is the realization of a two fundamental layered architecture, signaling layer and media layer. The signaling protocol is used to create, modify, and terminate media sessions between participants. The signaling layer is further divided into two layers, service layer and service control layer, in the IP Multimedia Subsystem (IMS) specification. Two widely used communication systems are IMS, and Peer-to-Peer Session Initiation Protocol (P2P SIP). Service providers, who behave as brokers between callers and callees, implement communication systems, heavily controlling the signaling layer. These providers do not take the diversity aspect of end users into account. This dissertation identifies three technical barriers in the current communication systems especially in the signaling layer. Those are: I. lack of openness and flexibility in the signaling layer for end users. II. difficulty of development of network-based, session-based services. III. the signaling layer becomes complex during the high call rate. These technical barriers hinder the end-user innovation with communication services. Based on the above listed technical barriers, the first part of this thesis defines a concept and architecture for a communication system in which an individual user becomes the service provider. The concept, My Own Communication Service Provider (MOCSP) and MOCSP system is proposed and followed by a call flow. Later, this thesis provides an analysis that compares the MOCSP system with existing communication systems in terms of openness and flexibility. The second part of this thesis presents solutions for network-based, session based services, leveraging the proposed MOCSP system. Two innovative services, user mobility and partial session transfer/retrieval are considered as examples for network-based, session-based services. The network-based, sessionbased services interwork with a session or are executed within a session. In both cases, a single functional entity between caller and callee consistently enables the media flow during the call initiation and/or mid-call. In addition, the cooperation of network call control and end-points is easily achieved. The last part of the thesis is devoted to extending the MOCSP for a high call rate and includes a preliminary comparative analysis. This analysis depends on four factors - scalability limit, complexity level, needed computing resources and session setup latency - that are considered to specify the scalability of the signaling layer. The preliminary analysis clearly shows that the MOCSP based solution is simple and has potential for improving the effective usage of computing resources over the traditional communication systems

Cieľom práce je navrhnúť a vytvoriť hlasom ovládaného asistenta(voicebota), ktorý bude ľahko nasaditeľný na webovú stránku. Používateľom tak bude poskytnutý moderný spôsob, ako prirodzene komunikovať cez internetový prehliadač. Hlavný dôraz je kladený na synchronizáciu medzi hlasovým asistentom a obsahom na webovej stránke. Synchronizácia je dosiahnutá obojsmerným prenosom hlasu a textových príkazov medzi klientom a serverom. Na to je použitá technológia WebRTC v kombinácií so signalizačným protokolom SIP. Práca sa zaoberá oblasťami ako VoIP telefonovanie, počítačové siete a strojové učenie(proprietárne rečové technológie od Phonexie). Benefitom nasadenia hlasového asistenta je zníženie nákladov na odchádzajúce hovory pre klientov, odľahčenie agentov na call centrách pri odpovedaní na často kladené otázky a zvýšenie záujmu zákazníkov vďaka použitiu nových technológií.