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Design and implementation of a novel secured and wide WebRTC signalling mechanism for multimedia over internet

机译:在互联网上多媒体的新型和宽WebRTC信令机制的设计与实现

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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.
机译:现代和免费的技术,称为Web的实时通信(的WebRTC)得到了改进,允许在没有插件的浏览器到浏览器的多媒体通信。在合同,已的WebRTC未分类的一个特定的信令机制来设置,建立和浏览器之间的端到端通信。本申请的主要目标是产生和执行而不插件不同用户之间通过Internet进行多媒体通信的新颖的WebRTC信令机制。此外,它已被应用在不同的浏览器,如浏览器,Safari浏览器,谷歌Chrome,Firefox和Opera没有任何下载或费用。此应用程序使用下ASP.net和C#语言JavaScript语言设计的。此外,为防止无关用户访问或攻击会话,用户ID发起和加入过程中使用的加密技术已完成。该系统已经在多个用户之间真正的实现了就业;因此,带宽消耗,CPU,和体验质量(QoE)的评价完成的。结果表明,其适用于不同的浏览器,多个用户,和不同的网络,诸如以太网和无线的原始信令机制。此外,它设置会话发起方,节省通信即使发起者叶效率,并与现有的参与者沟通的新参与者,等等。这研究的重点是建立一个新的信号机制,资源的支援WebRTC视频会议的局限。

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