Video conference in HTML 5

Tato bakalářská práce se věnuje návrhu a realizaci webové aplikace typu klient, která využívá protokol SIP pro signalizaci při multimediální komunikaci. Ta je přenášena přes protokol TCP, přesněji podprotokol WebSocket. V první části jsou popsány základní vlastnosti protokolu SIP, WebSocket a taky základní informace o projektu webRTC. Hlavní částí práce je popis návrhu a implementace webové aplikace pro audio-video komunikaci a pro textovou komunikaci v rámci posílání zpráv. V této části se nachází obeznámení s postupy, které byly použity během vývoje. Je popsán také výběr knihoven k tomu použitých. Na konci byla aplikace podrobena testování.This bachelor thesis follow proposal and realization of client-type web application, which uses SIP protocol for multimedia communication signalization. The communication is relayed via TCP protocol, more precisely via WebSocket. There are described basic properties of SIP and WebSocket protocols together with information about webRTC in first part of this thesis. The main part of thesis describes proposal and implementation of web application for AV and text communication. The briefing about progress reached during development is mentioned here as well as used libraries. Application was tested after finishing.

Peer-to-Peer Conferencing using Blockchain, WebRTC and SIP

  The owner of the centralized video platform has more control over uploaded content than the content producer does. But the other Blockchain-based decentralized video services are attempting to reduce ad pressure and get rid of middlemen. The article suggests a combination of a safe encryption technique and an access control mechanism created "with technology" to create a successful decentralized video streaming platform built on the Blockchain. Peer-to-peer (P2P) overlays are one of the complicated network applications and services that have been migrated to the Web as a result of the increasing support for Web Real-Time Communication (WebRTC) standard in modern browsers for real-time communications. The expansion of access networks’ bandwidth also makes it possible for end users to start their own content businesses. This paper presents a preliminary proposal of metrics and technologies to move toward a decentralized cooperative architecture for large-scale, real-time live stream content de- livery based on WebRTC, without the requirement of a Content Delivery Network (CDN) infrastructure. The paper takes into account the light of the aforementioned aspects [6]

Designing and prototyping WebRTC and IMS integration using open source tools

WebRTC, or Web Real-time Communications, is a collection of web standards that detail the mechanisms, architectures and protocols that work together to deliver real-time multimedia services to the web browser. It represents a significant shift from the historical approach of using browser plugins, which over time, have proven cumbersome and problematic. Furthermore, it adopts various Internet standards in areas such as identity management, peer-to-peer connectivity, data exchange and media encoding, to provide a system that is truly open and interoperable. Given that WebRTC enables the delivery of multimedia content to any Internet Protocol (IP)-enabled device capable of hosting a web browser, this technology could potentially be used and deployed over millions of smartphones, tablets and personal computers worldwide. This service and device convergence remains an important goal of telecommunication network operators who seek to enable it through a converged network that is based on the IP Multimedia Subsystem (IMS). IMS is an IP-based subsystem that sits at the core of a modern telecommunication network and acts as the main routing substrate for media services and applications such as those that WebRTC realises. The combination of WebRTC and IMS represents an attractive coupling, and as such, a protracted investigation could help to answer important questions around the technical challenges that are involved in their integration, and the merits of various design alternatives that present themselves. This thesis is the result of such an investigation and culminates in the presentation of a detailed architectural model that is validated with a prototypical implementation in an open source testbed. The model is built on six requirements which emerge from an analysis of the literature, including previous interventions in IMS networks and a key technical report on design alternatives. Furthermore, this thesis argues that the client architecture requires support for web-oriented signalling, identity and call handling techniques leading to a potential for IMS networks to natively support these techniques as operator networks continue to grow and develop. The proposed model advocates the use of SIP over WebSockets for signalling and DTLS-SRTP for media to enable one-to-one communication and can be extended through additional functions resulting in a modular architecture. The model was implemented using open source tools which were assembled to create an experimental network testbed, and tests were conducted demonstrating successful cross domain communications under various conditions. The thesis has a strong focus on enabling ordinary software developers to assemble a prototypical network such as the one that was assembled and aims to enable experimentation in application use cases for integrated environments

SIP client with advanced features

Hlavním cílem této bakalářské práce je návrh a realizace softwarového klienta, který bude schopen provádět textovou i video komunikaci přes síť založenou na protokolu IP. Tato aplikace by měla být spouštěna přímo z webové stránky a neměla by vyžadovat instalaci žádného dalšího software na koncový počítač. V první části jsou popsány možné technologie pro vytvoření této aplikace. Dále následuje stručný popis použitých technologií. Důraz je kladen především na protokol SIP. V práci je stručně popsána charakteristika několika knihoven, které byly uvažovány pro implementaci do aplikace. Návrh řešení obsahuje popis grafického prostředí a chování aplikace. V poslední části práce je výsledná aplikace otestována v různých modelových situacích.The main objective of this thesis is the design and implementation of client software, which will be able to conduct text and video communication over an IP based network. The client application is supposed to run from a web page without the need of any additional software. The first part describes possible technologies suitable for the development of such an application. After that follows a brief description of the selected technology with emphasis on SIP protocol. The thesis also includes a brief description of characteristics of libraries that were considered for implementation into the application. The proposed solution consists of the description of the graphic environment and behavior of the application. The developed application is tested in different situations in the last part of the thesis.

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

Signalling gateway for videoconferencing

Obecně se pod pojmem brána (Gateway) v počítačových sítích myslí uzel, který spojuje dvě sítě s odlišnými protokoly. Tato diplomová práce se zabývá překladem mezi 2 protokoly, protokolem SIP a protokolem WebSocket. Tento překlad je v práci nejprve popsán teoreticky a následně se práce zabývá způsobem řešení tohoto překladu s pomocí dostupných nástrojů na Internetu.Generally, the term gateway (gateway) in computer networks holds a node that connects two networks with different protocols. This thesis deals with the translation between two protocols, SIP protocol and WebSocket. This translation is the work first described theoretically and then work deals with the way this translation solutions using tools available on the Internet.

Performance Evaluation of WebRTC-Based Video Conferencing: A Comprehensive Analysis

In an ever-evolving technological landscape, addressing the performance challenges of real-time communication protocols is crucial. Real-time communication, facilitated by streaming media protocols, utilizes peer-to-peer or client-server models to enhance Quality of Service (QoS). WebRTC (Web Real-Time Communication) stands as a widely adopted, browser-based, open-source, peer-to-peer protocol, offering real-time media transmission through JavaScript APIs without third-party plugins. This paper presents an in-depth performance evaluation of a WebRTC-based video conferencing system using Socket.io services on a Node.js server. Our research expands on recent studies by introducing a comprehensive set of performance parameters, including Processing delay, CPU Utilization, Latency, Jitter, and Packet Loss, and packet delay. Our findings indicate that WebRTC performs exceptionally well within specific latency thresholds. However, scalability concerns emerge when a large number of clients are introduced, especially in bandwidth-constrained environments

Inter-domain interoperability framework based on WebRTC

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