WebNSM a novel scalable WebRTC signalling mechanism for many-to-many video conferencing

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

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

Performance evaluation of resources management in WebRTC for a scalable communication

Web Real-Time Communication (WebRTC) offers peer-to-peer communications without any plug-ins. However, WebRTC cannot provide scalability because of its method that depends on a single server or due to the resource limitations and network topology in the architectural of the WebRTC. This paper aims to design a real environment using MATLAB simulation tools to specify the limitations of resources in WebRTC for bi-directional video conferencing, such as CPU performance, bandwidth consumption and Quality of Experience (QoE) using different topologies such as mesh, star and hybrid (a combination of unidirectional/star & bi-directional/mesh). Moreover, several CPU cores like i3, i5, i7, Xeon, i9 and Xeon Phi, as well as bandwidths: 0.5, 1, 5, 10, 30, 50, 100, 500 and 1000 (Mb/s) were considered to achieve and expand the scalability. In this implementation, the factors of real-time implementation were used. Thus, the utilized measurements were already validated while MATLAB presents coefficient with 95% confidence bound. Additionally, this paper highlights the obstructions are preventing scalability in WebRTC using a centralized server. This illustration is beneficial for interested developers who intend to use WebRTC duplex video conferencing among undefined users and different topologies. Furthermore, our simulation-based’ performance evaluation shows the efficiency of the hybrid topology in decreasing the bandwidth overhead and CPU load in WebRTC

Performance Evaluation of Resources Management in WebRTC for a Scalable Communication

Web Real-Time Communication (WebRTC) offers peer-to-peer communications without any plugins. However, WebRTC cannot provide scalability because of its method that depends on a single server or due to the resource limitations and network topology in the architectural of the WebRTC. This paper aims to design a real environment using MATLAB simulation tools to specify the limitations of resources in WebRTC for bi-directional video conferencing, such as CPU performance, bandwidth consumption and Quality of Experience (QoE) using different topologies such as mesh, star and hybrid (a combination of unidirectional/star & bi-directional/mesh). Moreover, several CPU cores like i3, i5, i7, Xeon, i9 and Xeon Phi, as well as bandwidths: 0.5, 1, 5, 10, 30, 50, 100, 500 and 1000 (Mb/s) were considered to achieve and expand the scalability. In this implementation, the factors of real-time implementation were used. Thus, the utilized measurements were already validated while MATLAB presents coefficient with 95% confidence bound. Additionally, this paper highlights the obstructions are preventing scalability in WebRTC using a centralized server. This illustration is beneficial for interested developers who intend to use WebRTC duplex video conferencing among undefined users and different topologies. Furthermore, our simulation-based performance evaluation shows the efficiency of the hybrid topology in decreasing the bandwidth overhead and CPU load in WebRTC

WebNSM a novel scalable WebRTC signalling mechanism for many-to-many video conferencing

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

WebNSM: a novel WebRTC signalling mechanism for one-to-many bi-directional video conferencing

Many developers are interested in using Web Real-Time Communication (WebRTC) for video conferencing. But, selecting the appropriate topology in the architectural design of the WebRTC application is considered as one of the most potential issues. This paper aims to create a novel WebRTC signalling mechanism called WebNSM for one-to-many (bi-directional) video conferencing. WebNSM was created for unlimited users over the star topology based on Socket.io (API) mechanism and Multi-Connection methods. This implementation was applied via Wired of LAN and WAN networks and evaluated a signalling performance (WebNSM), bandwidth consumption, CPU performance, memory usage, Quality of Experience (QoE), maximum links and RTP calculation. In addition, it supplies video conferencing on different browsers without any user registration or additional software. The results demonstrate a signalling mechanism among various users, devices and networks to open one or multi rooms at the same time using the same server, specify room initiator to keep the session active even if another peer leaves, involving new user with current participants, etc. Moreover, this paper highlights the limitations of CPU performance, bandwidth consumption and using star topology for WebRTC video conferencing

Design and evaluation of browser-to-browser video conferencing in WebRTC

This paper describes the Web Real-Time Communication (WebRTC) technology and the implementation of its clients and server. The main aim is to design and implement WebRTC video conferencing between browsers in real implementation using Chrome and (Wired & WiFi) of LAN & WAN networks. Also, an evaluation of CPU performance, bandwidth consumption and Quality of Experience (QoE) was achieved. Moreover, a signalling channel between browsers using the WebSocket protocol via Node.js platform has been created and executed. This paper will give web developer an opportunity to comprehend the WebRTC technology, as well as to understand how to design WebRTC video conferencing

Performance evaluation of resources management in WebRTC for a scalable communication

Web Real-Time Communication (WebRTC) offers peer-to-peer communications without any plug-ins. However, WebRTC cannot provide scalability because of its method that depends on a single server or due to the resource limitations and network topology in the architectural of the WebRTC. This paper aims to design a real environment using MATLAB simulation tools to specify the limitations of resources in WebRTC for bi-directional video conferencing, such as CPU performance, bandwidth consumption and Quality of Experience (QoE) using different topologies such as mesh, star and hybrid (a combination of unidirectional/star & bi-directional/mesh). Moreover, several CPU cores like i3, i5, i7, Xeon, i9 and Xeon Phi, as well as bandwidths: 0.5, 1, 5, 10, 30, 50, 100, 500 and 1000 (Mb/s) were considered to achieve and expand the scalability. In this implementation, the factors of real-time implementation were used. Thus, the utilized measurements were already validated while MATLAB presents coefficient with 95% confidence bound. Additionally, this paper highlights the obstructions are preventing scalability in WebRTC using a centralized server. This illustration is beneficial for interested developers who intend to use WebRTC duplex video conferencing among undefined users and different topologies. Furthermore, our simulation-based’ performance evaluation shows the efficiency of the hybrid topology in decreasing the bandwidth overhead and CPU load in WebRTC