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-tomany (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.
Keyword:Local area network (LAN
There is a strong focus on the use of Web Real-Time Communication (WebRTC) for many-to-many video conferencing, while the IETF working group has left the signalling issue on the application layer. The main aim of this paper is to create a novel scalable WebRTC signalling mechanism called WebNSM for many-to-many (bi-directional) video conferencing. WebNSM was designed for unlimited users over the mesh topology based on Socket.io (API) mechanism. A real implementation was achieved via LAN and WAN networks, including the evaluation of bandwidth consumption, CPU performance, memory usage, maximum links and RTPs calculation; and Quality of Experience (QoE). In addition, this application supplies video conferencing on different browsers without having to download additional software or user registration. The results present a novel signalling mechanism among various users, devices and networks to open one or multi rooms at the same time using the same server, determine room initiator to keep the session active even if the initiator or another peer leaves, sharing new user with current participants, etc. Moreover, this experiment highlights the limitations of CPU performance, bandwidth consumption and using mesh topology for WebRTC video conferencing.
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.
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