ADVANCED VIDEOCONFERENCING SERVICES BASED ON WEBRTC

2018

Remote collaboration is common nowadays in conferencing, telehealth and remote teaching applications. To support these interactive use cases, Real-Time Communication (RTC) solutions, as the open-source WebRTC framework, are generally used. WebRTC is peer-to-peer by design, which entails that each sending peer needs to encode a separate, independent stream for each receiving peer in the remote session. This approach is therefore expensive in terms of number of encoders and not able to scale well for a large number of users. To overcome this issue, a WebRTC-compliant framework is proposed in this paper, where only a limited number of encoders are used at sender-side. Consequently, each encoder can transmit to a multitude of receivers at the same time. The conference controller, a centralized Selective Forwarding Unit (SFU), dynamically forwards the most suitable stream to each of the receivers, based on their bandwidth conditions. Moreover, the controller dynamically recomputes the encoding bitrates of the sender, to follow the long-term bandwidth variations of the receivers and increase the delivered video quality. The benefits of this framework are showcased using a demo implemented using the Jitsi-Videobridge software, a Web-RTC SFU, for the controller and the Chrome browser for the peers. Particularly, we demonstrate how our framework can improve the received video quality up to 15% compared to an approach where the encoding bitrates are static and do not change over time.

International Journal of Electrical and Computer Engineering (IJECE), 2020

Many years ago, Flash was essential in browsers to interact with the user media devices, such as a microphone and camera. Today, Web Real-Time Communication (WebRTC) technology has come to substitute the flash, so browsers do not need the flash to access media devices or establish their communication. However, WebRTC standards do not express precisely how browsers can record audios, videos or screen instead of describing getUserMedia API that enables a browser to access microphone and camera. The prime objective of this research is to create a new WebRTC recording mechanism to record audios, videos, and screen using Google Chrome, Firefox, and Opera. This experiment applied through Ethernet and Wireless of the Internet and 4G networks. Also, the recording mechanism of this research was obtained based on JavaScript Library for audio, video, screen (2D and 3D animation) recording. Besides, different audio and video codecs in Chrome, Firefox and Opera were utilised, such as VP8, VP9, and H264 for video, and Opus codec for audio. Not only but also, various bitrates (100 bytes bps, 1 Kbps, 100 Kbps, 1 MB bps, and 1 GB bps), different resolutions (1080p, 720p, 480p, and HD (3840* 2160)), and various frame-rates (fps) 5, 15, 24, 30 and 60 were considered and tested. Besides, an evaluation of recording mechanism, Quality of Experience (QoE) through actual users, resources, such as CPU performance was also done. In this paper, a novel implementation was accomplished over different networks, different browsers, various audio and video codecs, many peers, opening one or multi browsers at the same time, keep the streaming active as much as the user needs, save the record, using only audio and/or video recording as conferencing with full screen, etc.

International Journal for Research in Applied Science & Engineering Technology (IJRASET), 2022

With the modern and rapid development of the internet, people's connections are more important than ever, and they're looking for new ways to communicate with one another in real time. WebRTC is a futuristic technology that enables realtime communication in audio, video, and data transmission through web browsers without the need for a plugin by using JavaScript APIs (Application Programming Interface). In this paper, we present a web peer-to-peer real-time communication system that allows users to communicate across a communication channel with high-speed data transmission using WebRTC technology, HTML5, and a Node.js server address. The outcome demonstrates that the system is stable, fully functioning, and secure.

Electronic Notes in Theoretical Computer Science, 2016

The research presented in this paper introduces RendezVous , which is a system oriented to provide its users with a greater communication availability through the convergence of the conventional telephony and the real-time multimedia on the web browsers. The system consists on a web application that provides a videoconference room for multiple users without having to download any additional software. For the cases in which a user with no internet connection might need to participate in a conference, it is possible to dial and answer phone calls to/from the PSTN directly on the web browser, allowing the telephonic user to interact with the others using his/her voice. The aim of the present project is to propose a unified communications system that differs from others mainly in the interaction with the telephone network directly from a web browser while having an active videoconference, allowing real-time exchange of media streams between these two technologies. The system RendezVous was implemented with the use of WebRTC (Web Real-Time Communications) for the transmission of audio and video on real-time, Node.js as a web and signaling server, as well as the software Asterisk for providing telephonic access, along with jsSIP, which is a JavaScript library for implementing a SIP User Agent.

International Journal of Electrical and Computer Engineering (IJECE), 2018

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. Keyword: Local area network (LAN) Mesh topology Quality of experience (QoE) Socket.IO The real-time web communication (WebRTC) Web new signalling mechanism (WebNSM) Wide area network (WAN)

International Journal of Scientific Research in Computer Science, Engineering and Information Technology, 2021

The covid-19 pandemic has led to things happening virtually. Students are attending their classes in online mode. More than 50 percent of the working population is working from home. Online meetings have become necessary part of everyone's life. With the existing platforms, users need to setup or install packages on their systems to run the application which sometimes becomes confusing for first timers or non-technical people. This paper proposes to build a full-fledged feature rich web-based video conferencing application using WebRTC technology. WebRTC is used to enable real time audio and video communication from a web browser without the need of installing software or plugins so that users can focus on their work rather than worrying about how to use a video conferencing platform.

Intelligent Computing, 2018

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.

2016

The Web Real-Time Communication (WebRTC) framework provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web-browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context, and gives requirements for which RTP features, profiles, and extensions need to be supported.

Multimedia Tools and Applications, 2018

Remote video collaboration is common nowadays in conferencing, telehealth and remote teaching applications. To support these low-latency and interactive use cases, Real-Time Communication (RTC) solutions are generally used. WebRTC is an open-source project for real-time browser-based conferencing, developed with a peer-to-peer architecture in mind. In this peer-to-peer architecture, each sending peer needs to encode a separate, independent stream for each receiving peer participating in the remote session, which makes this approach expensive in terms of encoders and not able to scale well for a large number of users. This paper proposes a WebRTC-compliant framework to solve this scalability issue, without impacting the quality delivered to the remote peers. In the proposed framework, each sending peer is only equipped with a limited number of encoders, much smaller than and independent of the number of receiving peers. Consequently, each encoder transmits to a multitude of receivers at the same time, to improve scalability. A centralized node based on the Selective Forwarding Unit (SFU) principle, called conference controller, forwards the best stream to the receiving peers, based on their bandwidth conditions. Moreover, the conference controller dynamically recomputes the encoding bitrates of the sending peers, to maximize the quality delivered to the receiving peers. This approach allows to closely follow the long-term bandwidth variations of the receivers, even with a limited number of encoders at sender-side, and increase the delivered video quality. An integer linear programming formulation for the bitrate recomputation problem is presented, which can be optimally solved when the number of receivers is small. An approximate, scalable method is also proposed using the K-means clustering algorithm. The gains brought by the proposed framework have been confirmed in both simulation and emulation, through a testbed implementation using the Google Chrome browser and the open-source Jitsi-Videobridge software. Particularly, we focus on a remote collaboration scenario where the interaction among the remote participants is dominated by a single peer, as in a remote teaching scenario. When a single sending peer equipped with three encoders transmits to 28 receiving peers, the proposed framework improves the average received video bitrate up to 15%, compared to a static solution where the encoding bitrates do not change over time. Moreover, the dynamic bitrate recomputation is more efficient than a static association in terms of encoders used at sender-side. For the same configuration mentioned above, the same received bitrate is obtained in the static case using four encoders as in the dynamic case using three encoders.

2014

There is increasing interest in videoconferencing for social networks, but web developers are facing two basic challenges: Easy integration and scalability. The latter can be solved by a scalable architecture based on central media routers, which makes pragmatic use of the available resources while allowing low complexity server components. Easy integration, on the other hand, requires a simple API with inherent support for multipoint and intelligent audio and video composition. In this paper we describe how the EU-funded research project Vconect has addressed these two challenges and compare the results with Google+Hangout and WebRTC. As a prove of concept, the Vconect platform has been integrated with SAPO Campus, a social network operated by Portugal Telecom Group, with positive results from initial trials.