Videoconference System Based on WebRTC With Access to the PSTN
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- 作者：Alfonso Sandoval Rosas1 ; asandovalros@gmail.com ; José ; Luis Alejos Martí ; nez joseluis0791@gmail.com
- 关键词：Videoconference ; WebRTC ; WebAudio ; PSTN ; SIP ; Asterisk
- 刊名：Electronic Notes in Theoretical Computer Science
- 出版年：2016
- 出版时间：9 December 2016
- 年：2016
- 卷：329
- 期：Complete
- 页码：105-121
- 全文大小：2000 K
- 卷排序：329
文摘
The research presented in this paper introduces Rendez-Vous, 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 Rendez-Vous 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.