A design of multimedia conferencing system based on WebRTC Technology

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.

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 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.

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.

Videoconference System Based on WebRTC With Access to the PSTN

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.

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.

The wide bandwidths of the communication and advanced bilateral robot control technologies have led us to realize multi-sensational teleoperation systems that provide auditory, visual, and haptic information of remote site to its operator simultaneously. Such teleoperation system is expected to be applied to medicine, education, work in hazardous environment, online game or other use. Especially, haptic sensation is extremely required to improve its operability and safety in medical operations. In this chapter, a teleoperation system is assumed to consist of three components, i.e., audio, video, and haptics transmission systems. In addition, each transmission system can be distinguished into local site and remote site. When the distance between the remote site and the local site of a teleoperation system is far enough, the performance of its communication line becomes crucial factor to decide the operability of the teleoperation. In general, since the data rate of video information is higher than the bandwidth of communication line, data compression is indispensable. Moreover, data compression and decompression are considered as time consuming processes. As a result, video transmission system must be delayed in principle. In multi-media context, audio transmission system should be delayed artificially so as to keep its playout delay to be the same as that of video transmission system. For example, lip sync is necessary for its operator to feel the audio and video contents in naturally. From this point of view, the bandwidth of its communication line is the only requirement for realize video and audio transmission systems. Even if these transmission systems are used to make a conversation for its operator with a remote person, it is said that 200ms delay is allowable. This requirement is rather negligible in the Internet in nowadays. On the other hand, a haptics transmission system requires short and stable performance of delay for communication line. This is because the achievable bandwidth of haptic sensation is limited by the round trip time of communication line since haptics transmission controller includes the delay inside of its closed control loop. In fact, a human being can feel the sense of touch at the tip of a finger up to about 400Hz. To recognize this bandwidth of haptic sensation, its sampling period must be higher than 800Hz according to Shannon’s sampling theory. Thus, the round trip time of communication line must be shorter than 1ms. This value is much shorter than the allowable delay of audio and video transmission systems. In short, haptics transmission system requires short delay while video transmission system requires wide bandwidth for communication line. 2

Abstract: Human interaction is important in today's internet-based world. To effortlessly approach each other to communicate people can use the new technology i.e. web real-time communication (WebRTC). Our real-time communication system will provide virtual voice rooms to users to audio chat in groups or peer to peer using technologies like WebRTC, Node.js, Web Sockets and React.js. On the basis of a range of topics, users can create public/private rooms and join them as well. The implications of this study are better than the previous projects

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.

RF technologies has facilitated communication including audio, images, video and data transmission. However, the RF spectrum is getting increasingly crowded, and alternative methods to transfer data need to be explored. In this paper, software-hardware co-design is implemented to design the audio data synchronization and transmission for an optical-through-air transceiver system. Besides audio transmission, the performance of the VLC system is verified through testing serial data transmission and text transmission. The experimental results show that the VLC system can achieve up to 4 Mb/s data rate using OOK modulation with red light. This system provides a practical step-by-step guidance for setting up a VLC system for data transmission.

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.