Community Space using WebRTC

Real-time communication (RTC) is a new standard and industry-wide effort that expand the web browsing model, allowing access to information in areas like social media, chat, video conferencing, and television over the internet, and unified communication. These systems users can view, record, remark, or edit video and audio content flows using time-critical cloud infrastructures that enforce the quality of services. However, there are many proprietary protocols and codecs available that are not easily interoperable and scalable to implement multipoint video-conference systems. WebRTC (Web Real-Time Communication) is a State-of-the-Art open technology that makes real-time communication capabilities in audio, video, and data transmission possible in real-time communication through web browsers using JavaScript APIs (Application Programming Interfaces) without plug-ins. This paper aims to introduce the P2P video conferencing system based on Web Real-Time Communication (WebRTC). In this paper, we have proposed a web-based peer-to-peer real-time communication system using the Mozilla Firefox together with the ScaleDrone service that enables users to communicate with high-speed data transmission over the communication channel using WebRTC technology, HTML5 and use Node.js server address. Our experiments show that WebRTC is a capable building block for scalable live video conferencing within a web browser.We also study a thermal camera for the videoconference system to identify the temperature body results regarding the COVID 19 crisis.

Web Real-Time Communication (WebRTC) introduces real-time multimedia communication as native capabilities of Web browsers. With the adoption of WebRTC the Web browsers will be able to use WebRTC to communicate with one another (peer-to-peer), and with WebSocket servers such as Mobicents SIP Servlets and other server technologies that support WebSocket communication to enable SIP-to-WebRTC communication. This paper outlines the potential of WebRTC and discusses the two common methods of doing real-time communication in Web browsers through WebRTC. The methods are JavaScript Object Notation (JSON) via XMLHttpRequest (XHR) and Session Initiation Protocol (SIP) via WebSocket. A three-user WebRTC video chat prototype application was developed and used to evaluate both methods. Additional signalling overhead introduced into a browser by each method was determined. The results showed WebRTC-SIP/WS has more overhead than WebRTC-JSON/XHR. These signalling overhead findings are useful in that they could help application developers make decision on their choice of technologies and protocols when developing WebRTC-supported applications.

WebRTC (Web Real-Time Communication) is one of the most popular technologies that provide the infrastructure for real-time data communications. Many applications used in military and civilian life depend on WebRTC technology directly or indirectly in the background. Due to its wide area of applications, the security of WebRTC communications is of great importance. In this paper, we improve the existing security infrastructure of WebRTC. The proposed solution provides a real-time communication infrastructure between peers on the network by making use of quite new and popular blockchain and smart contract technologies. Within the scope of the paper, a web-based application with video calling feature demonstrating blockchain interactions and suggested security enhancements have been developed. In the application, operations such as adding, updating, and authorizing users can be performed on the smart contract deployed on the live Ethereum network. Based on the principle of immutability of blockchains, users on the smart contract are mapped to the peers (nodes) in the WebRTC communication to be established and used in the identity verification process. Consequently, this paper proposes a secure identity verification method that could be used by real-time WebRTC communication peers to verify each other on the blockchain. The proposed solution is shown to resilient against man-in-the-middle attacks.

Network traffic recognition serves as a basic condition for network operators to differentiate and prioritize traffic for a number of purposes, from guaranteeing the Quality of Service (QoS), to monitoring safety, as well as monitoring and detecting anomalies. Web Real-Time Communication (WebRTC) is an open-source project that enables real-time audio, video, and text communication among browsers. Since WebRTC does not include any characteristic pattern for semantically based traffic recognition, this paper proposes models for recognizing traffic generated during WebRTC audio and video communication based on statistical characteristics and usage of machine learning in Weka tool. Five classification algorithms have been used for model development, such as Naive Bayes, J48, Random Forest, REP tree, and Bayes Net. The results show that J48 and BayesNet have the best performances in this experimental case of WebRTC traffic recognition. Future work will be focused on comparison of a wide range of machine learning algorithms using a large enough dataset to improve the significance of the results.

Existing video conferencing systems that are often used in telehealth services have been criticized for a number of reasons: (a) they are often too expensive to purchase and maintain, (b) they use proprietary technologies that are incompatible to each other, and (c) they require fairly skilled IT personnel to maintain the system. There is a need for less expensive, compatible, and easy-to-use video conferencing system. The web real-time communication (WebRTC) promises to deliver such a solution by enabling web browsers with real-time communications capabilities via simple JavaScript APIs. Utilizing WebRTC, users can conduct video/audio calls and data sharing through web browsers without having to purchase or download extra software. Though the promise and prospective of WebRTC have been agreed on, there have not been many cases of real life applications (in particular in telehealth) that utilizes the WebRTC. In this paper, we present our practical experience in the design and implementation of a video conferencing system for telehealth based on WebRTC. Our video conferencing system is a part of a large tele-home monitoring project that is being carried out at six locations in five different states in Australia. One of the aims of the project is to evaluate whether high-bandwidth enabled telehealth services, delivered through tele-home monitoring, can be cost effective, and improve healthcare outcomes and access to care. This paper however focuses on WebRTC-based video conferencing system which allows online meetings between remotely located care coordinators and patients at their home. We discuss the underlying issues, detailed design and implementation, and current limitations of using WebRTC in a real life application.

Web Real-Time Communications (WebRTC) technology is an enabler of native transmission of multimedia information between two and more Web browsers. WebRTC is based on HTML version 5 (HTML5) and JavaScript languages. Although both real-time media streams (audio, video) and non-real-time non-media flows (also present in WebRTC architecture) are transmitted directly between browsers, to enable transmission of information necessary for session management (and more precisely: session establishment) purposes, a signalling server is needed. Such server may be regarded as a rendezvous point of end systems that use WebRTC technology. In this paper, node.js run-time environment and principle of building of Webrtc signalling server with the use of scripts written in JavaScript and run in node.js are presented. Example of script of a simple signalling server also is shown. The exemplary server connects users of one and only session, although this session may serve not only one-to-one, but also many-to-many connections. Despite its simplicity, this is a fully functional signalling server, able to serve signalling messages for purposes of message ex-change, or for purposes of initialization of WebRTC-based Internet telephony or simple conferencing systems.

During an emergency situation disabled people face many obstacles to access the emergency systems since they fail to support IP-based services and do not take into account their accessibility requirements. This paper describes the key elements of the EMYNOS platform which is a Next Generation emergency platform that aims to provide IP based emergency services and integrate information provided by the social media, user's profile and contextual information in order to provide personalized emergency services and real time communication. To achieve this, tactile and pressure feedback from haptic devices as well as real time communication audio-video, text-video and voice-video calls will be initiated through the browser-to-browser web real time communication (WebRTC) technology. In addition, WebRTC technology will be used as a backup platform in case the default infrastructure collapses.

Web identity and resolution identifies, authenticates, and locates users on the Web. It's a critical aspect of Web Real-Time Communications (WebRTC), a joint W3C and IETF effort to develop standards for secure real-time communication over the Web. Although WebRTC proposes a security architecture to integrate with federated Web identity systems, the architecture isn't compatible with identity protocols that don't satisfy WebRTC requirements. To fill the gap, the authors describe two alternative architectures to adapt incompatible identity protocols to the WebRTC security architecture. They also describe a Web-of-trust model to address hierarchical identity systems' limitations, and a mirror-presence system to locate users on the Web in real time while they are moving between websites.

Web Real-Time Communication (WebRTC) is an open-source challenge that allows real-time communique talents immediately within internet browsers and cellular applications. It permits peer- to-peer communique, which includes audio, video, and information sharing, without the want for any extra plugins or 1/3-party software program. WebRTC abstracts the complexities of organising direct communique channels among devices, allowing developers to integrate seamless real-time conversation functionalities into web applications. It leverages a aggregate of standardized protocols, such as Session Initiation Protocol (SIP) and Interactive Connectivity Establishment (ICE), and JavaScript APIs to allow steady and efficient conversation among customers, enhancing the general person enjoy for diverse online collaboration, conferencing, and conversation programs.

One of the most widely used technologies for peer-to-peer real-time data transmission is WebRTC (Web Real-Time Communication).Both in the military and in civilian life, WebRTC technology is employed in many background applications, either directly or indirectly.For this reason, WebRTC communication security is crucial.The main goal of this research is to enhance WebRTC's current security framework by utilizing relatively new and smart contract technologies, the suggested solution offers a real-time system for communication amongst peers on the network.Additionally, a web application featuring a video calling function that illustrates blockchain interactions as well as recommended security improvements has been built.Furthermore, the application may be used on the smart contract that is active on the Ethereum network to carry out tasks like declaring, modifying, and authorizing users.As a result, this article suggests a safe identity verification technique that peers for real-time WebRTC communication could utilize to validate one another on the blockchain.

The emerging paradigm of network function virtualization (NFV) technology promises an efficient solution for optimized service deployment in the cloud computing environment thanks to its ability to dynamically add or remove virtual resources when there is a change in workload. Nevertheless, telecom providers are still facing a challenging issue in efficiently adopting NFV to deploy Web real-time communication (WebRTC) service on top of IP multimedia subsystem (IMS). Providing WebRTC service increases the inherent complexity of the IMS system in terms of the number of service nodes as virtual network functions (VNFs) and the way they interact, both of which play significant roles in the problem of optimally allocating resources. This paper proposes a virtualized interworking system between IMS and WebRTC called NFV-based interworking architecture, and describes the mechanism for VNFs to exchange messages with each other. We present an analytic system model considering the constraints of resources, quality of service (QoS), and service costs. A real-time Markov approximation-based resource allocation algorithm (RIDRA) is then designed allowing a provisioned resource at service nodes to be reconfigured in time to meet performance requirements. The proposed solution is evaluated on the large scale by simulation and on the small scale by our developed testbed. Experimental results reveal that our algorithm effectively responds to fluctuating service demands with a service cost reduced by 19% via efficiently allocating virtual resources while maintaining QoS requirement.

Multimedia services are one of the internet needs with high data traffic network count since 2019. Two of the multimedia services, Web Real-Time Communications (WebRTC) and Session Initiation Protocol (SIP), have been widely used in applications for conducting video conferencing. The main objective of this research is to analyze network performance by an application with a client-server for audio and video communications developed with WebRTC and SIP protocols. The SIP system uses the FreePBX server, and the softphone application uses Bria. Whereas WebRTC uses JavaScript with servers on Ubuntu using Node.js. The analysis application uses a star topology and runs on a local network using Wi-Fi. After testing, the results show that the throughput, jitter, and packet loss of WebRTC are better than SIP. This result is caused by several factors including the type of codec, the type of platform, and the way of signaling and routing of each protocol.

Videoconference System Based on WebRTC With Access to the PSTN

We present lessons learned in developing cross platform multi-party team applications. Our apps include a range of communication and collaboration scenarios: document and content sharing in a team space, an agent-based meeting helper, phone number dialer via a voice-over-IP (VoIP) gateway, and multi-party call in peer-to-peer or client-server mode. We use web real-time communication (WebRTC) to enable the audio and video media paths in the apps. We use frameworks such as Chrome Apps and Apache Cordova to create apps that can be accessed from a browser, or installed on a desktop, mobile device, or wearable. The challenges and techniques described in our paper related to audio, video, network, power conservation and security are important to other developers building cross-platform apps involving WebRTC, VoIP and cloud services.

In this article, we propose a quality-based video bitrate control method for web real-time communication (WebRTC)-based teleconferences. Video bitrate is controlled on the basis of quality of service (QoS) parameters such as delay and packet-loss rate in WebRTC. Therefore, the amount of transferred data may increase because media streams are transmitted at excessive quality levels depending on QoS conditions (e.g., the jitter and packet-loss rate are low). An increase in transferred data leads to higher operational cost (i.e., data transferred cost) and affects profitable growth. In the proposed method, quality desired by a service provider is set as TargetQuality, and the video bitrate of each stream is controlled aiming at TargetQuality, thereby suppressing the amount of transferred data while maintaining sufficient quality. The proposed method is implemented to an actual teleconference system and is evaluated in terms of its effect at reducing the amount of transferred data. The results show that the amount of transferred data can be reduced by more than 40% by setting the value of TargetQuality appropriately.