Design and implementation of a novel secured and wide WebRTC signalling mechanism for multimedia over internet

A modern and free technology called web real-time communication (WebRTC) was enhanced to allow browser-to-browser multimedia communication without plugins. In contract, WebRTC has not categorised a specific signalling mechanism to set, establish and end communication between browsers. The primary target of this application is to produce and implement a novel WebRTC signalling mechanism for multimedia communication between different users over the Internet without plugins. Furthermore, it has been applied over different browsers, such as Explorer, Safari, Google Chrome, Firefox and Opera without any downloading or fees. This application designed using JavaScript language under ASP.net and C# language. Moreover, to prevent irrelevant users from accessing or attacking the session, user-id for initiating and joining the course using encryption technique was done. This system has been employed in real implementation among various users; therefore, an evaluation of bandwidth consumption, CPU, and quality of experience (QoE) was accomplished. The results show an original signalling mechanism which applied to different browsers, multiple users, and diverse networks such as Ethernet and Wireless. Besides, it sets session initiator, saves the communication efficient even if the initiator leaves, and communicating new participator with existing participants, etc. This studying focuses on the creation of a new signalling mechanism, the limitations of resources for WebRTC video conferencing

Design and Implement a Hybrid WebRTC SignallingMechanism for Unidirectional & Bi-directional VideoConferencing

WebRTC (Web Real-Time Communication) is a technology that enables browser-to-browser communication. Therefore, a signalling mechanism must be negotiated to create a connection between peers. The main aim of this paper is to create and implement a WebRTC hybrid signalling mechanism named (WebNSM) for video conferencing based on the Socket.io (API) mechanism and Firefox. WebNSM was designed over a combination of different topologies, such as simplex, star and mesh. Therefore it offers several communications at the same time as one-to-one (unidirectional/bidirectional), one-to-many (unidirectional) and many-to-many (bi-directional) without any downloading or installation. In this paper, WebRTC video conferencing was accomplished via LAN and WAN networks, including the evaluation of resources in WebRTC like bandwidth consumption, CPU performance, memory usage, Quality of Experience (QoE) and maximum links and RTPs calculation. This paper presents a novel signalling mechanism among different users, devices and networks to offer multi-party video conferencing using various topologies at the same time, as well as other typical features such as using the same server, determining room initiator, keeping the communication active even if the initiator or another peer leaves, etc. This scenario highlights the limitations of resources and the use of different topologies for WebRTC video conferencing

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

Enterprise WebRTC Powered by Browser Extensions

ABSTRACT We use browser extensions to solve two important issues in adopting WebRTC (Web Real-Time Communications) in enterprises: how to integrate WebRTC-centric communication with existing systems such as corporate directories, communication infrastructure and intranet websites, and how to traverse media paths across enterprise firewalls. Vclick is a simple and easy to use web-based video collaboration application that enables click-to-call from other webpages. SecureEdge is a network border traversal system for policy and security enforcement, and consists of a secure media relay that sits at the network border or in the cloud. A browser extension in the enterprise user's device transparently injects this media relay in every WebRTC media path needing to traverse the enterprise network edge to enable authenticated border traversal without help from the websites hosting the WebRTC pages. We attempt to generically support WebRTC in enterprises on a variety of application scenarios instead of creating another fragmented communication island. The challenges faced and techniques used in our proof-of-concepts are likely extensible to other enterprise WebRTC scenarios using the emerging HTML5 technologies

Techno-Economic Feasibility of Web Real-Time Communications

WebRTC is an ongoing effort to build an open framework for real-time audio and video communication capabilities that turn Web browsers, and other clients supporting it, into a platform for person-to-person communication. Previously, real-time communication (RTC) has been achievable in the Web browser only by installing third party software. WebRTC brings native support for RTC to the Web browsers and exposes it freely to web developers via standardized JavaScript API. This brings RTC as a feature to the Web, which can foster further innovation. This thesis studies the techno-economic feasibility of WebRTC with the help of a framework for feasibility analysis of Internet protocols, developed by Levaä and Suomi (2013). To provide input for the framework, we conduct an interview study, as well as research of available Web resources. Further, we explore what market opportunities may arise, provided that WebRTC is successfully adopted. To do that, we use Value Network Configurations as a tool for studying and visualizing the possible relationships between market players and the roles they assume in the ecosystem. We find that WebRTC is a feasible technology in its basic, but highly relevant use case of one-to-one browser-to-browser communication. While we discover a number of unresolved challenges, we do not see any insurmountable obstacles that would prevent WebRTC adoption. WebRTC opens up opportunities for companies that would use it directly to deliver an RTC service, but also creates space for WebRTC PaaS providers in the market. Additionally, WebRTC interconnecting with legacy systems, such as PSTN or PLMN, opens up opportunity for telecom operators to explore creating new ways of communication for their customers

A Cloud Platform-as-a-Service for Multimedia Conferencing Service Provisioning

Multimedia Conferencing is the real-time exchange of media content (e.g. voice, video and text) between multiple participants. It is the basis of a wide range of conferencing applications such as massively multi-player online games and distance learning applications. For faster development as well as cost efficiency, developers of such conferencing applications can use conferencing services (e.g. dial-in audio conference) provided by third-parties. However, the third-party service providers face several challenges with respect to conferencing service provisioning (i.e. service development, deployment and management). One challenge is mastering complex low-level details of conferencing technologies, protocols and their interactions. Another challenge is resource elasticity. Number of conference participants varies during runtime. So resource utilization in an elastic manner is a critical factor to achieve cost efficiency. Cloud Computing can help tackle these challenges. It is a paradigm for swiftly provisioning a shared pool of configurable resources (e.g. services, applications, network and storage) on demand. It has three main service models: Infrastructure-as-a-Service (IaaS), Platform-as-a-Service (PaaS) and Software-as-a-Service (SaaS). Using a PaaS, service providers can provision conferencing services easily and offer them as SaaS. Nonetheless, cloud-based provisioning of conferencing services still remains a big challenge due to the shortcomings of existing PaaS. In this thesis, a PaaS architecture for conferencing service provisioning is proposed. It is based on a business model from the state of the art. It relies on conferencing IaaSs that, instead of VMs, offer conferencing substrates (e.g. dial-in signaling, video mixer and audio mixer). The conferencing PaaS enables composition of new conferences from substrates on the fly. Moreover, it provides conferencing service providers, who are experienced in programming, with high-level interfaces to abstract the internal complexities of conferencing. In order for PaaS to scale ongoing conferences elastically, an algorithm is also presented in this thesis. The conferencing PaaS is prototyped and performance measurements are made. The proposed algorithm’s performance is also evaluated

How far are we from WebRTC-1.0? An update on standards and a look at what's next

Real-time communication between browsers has represented an unprecedented standardization effort involving both the IETF and the W3C. These activities have involved both the real-time protocol suite and the application-level JavaScript APIs to be offered to developers in order to allow them to easily implement interoperable real-time multimedia applications in the web. This article sheds light on the current status of standardization, with special focus on the upcoming final release of the so-called WebRTC-1.0 standard ecosystem. It takes stock of the situation with respect to hot topics such as codecs, session description and stream multiplexing. It also briefly discusses how standard bodies are dealing with seamless integration of the initially competing effort known as “Object Real Time Communications.