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Analysis of Secure Real Time Transport Protocol on VoIP over Wireless LAN in Campus Environment

机译:校园环境中无线局域网上VoIP的安全实时传输协议分析

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In this research, we propose to implement Secure Real Time Transport Protocol (SRTP) on VoIP services in campus environment. Today, the deployment of VoIP in campus environment over wireless local area network (WLAN) is not considered on security during communication between two parties. Therefore, this study is to analyzed SRTP performance on different VoIP codec selection over wired. We have implemented a real VoIP network in University of Kuala Lumpur (UniKL), Malaysia. We use softphone as our medium communication between two parties in campus environment. The results show that implementation of SRTP is able to improve the VoIP quality between one-to-one conversation and multi conference call (many-to-many). In our experiment, it shows that iLBC, SPEEX and GSM codec are able to improve significantly the multi conference (many-to-many) VoIP quality during conversation. In additional, implementation of SRTP on G.711 and G.726 codec will decrease the multi conference (many-to-many) VoIP quality.
机译:在这项研究中,我们建议在校园环境中的VoIP服务上实现安全实时传输协议(SRTP)。如今,在两方之间的通信过程中,出于安全性考虑,不考虑在园区环境中通过无线局域网(WLAN)部署VoIP。因此,本研究旨在分析有线上不同VoIP编解码器选择时的SRTP性能。我们已经在马来西亚吉隆坡大学(UniKL)实施了一个真正的VoIP网络。我们使用软电话作为校园环境中两方之间的媒介通信。结果表明,SRTP的实施能够改善一对一对话和多方电话会议(多对多)之间的VoIP质量。在我们的实验中,它表明iLBC,SPEEX和GSM编解码器能够在通话过程中显着提高多会议(多对多)VoIP的质量。另外,在G.711和G.726编解码器上实现SRTP会降低多会议(多对多)VoIP的质量。

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