A new approach to the implementation of VoIP for SOHO network

Ali Mohsin, Muhammad Awais Bin Altaf, N. M. Sheikh, M. Javed, Muhammad Usman
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Abstract

VoIP has emerged from the advancements and revolutions in packet-based networks and has seen tremendous improvements over the years. Normally there are two approaches to the implementation of VoIP, one in the form of soft phone and other in the form of dedicated IP phone. In this paper we describe a new scheme to the implementation of VoIP which is different from both of these two approaches in that rather than implementing all the layers of internet as in dedicated IP phones or developing a software application to handle voice communication as in soft phone, We have relied on the functionalities of ordinary analog telephones(i.e. to generate voice and signaling information) and computer application(i.e. to transfer packet based data to the destination) in implementing VoIP. In our proposed scheme, actual voice data and signaling data(DTMF tones)will be transferred to the computer through our proposed hardware and the computer application written in JAVA takes the responsibility of transferring this voice data to the destination in the form of IP packets. We have used SIP as our signaling protocol to handle call setup and tear down. The prototype of our proposed scheme has been successfully completed and tested on our university LAN setup and I t is giving acceptable voice quality, un noticeable delays and latency, which are important metrics to consider while implementing VoIP.
SOHO网络VoIP实现的一种新方法
VoIP是从基于分组的网络的进步和革命中出现的,并且多年来取得了巨大的进步。VoIP的实现通常有两种方式,一种是软电话,另一种是专用IP电话。在本文中,我们描述了一种实现VoIP的新方案,它不同于这两种方法,因为它不是像专用IP电话那样实现互联网的所有层,也不是像软电话那样开发一个软件应用程序来处理语音通信,我们依赖于普通模拟电话的功能(即。生成声音和信号信息)和计算机应用程序(即将基于分组的数据传输到目的地),以实现VoIP。在我们提出的方案中,实际的语音数据和信令数据(DTMF音调)将通过我们提出的硬件传输到计算机,用JAVA编写的计算机应用程序负责将这些语音数据以IP数据包的形式传输到目的地。我们使用SIP作为信令协议来处理呼叫建立和中断。我们提出的方案的原型已经成功完成,并在我们的大学局域网设置上进行了测试,它提供了可接受的语音质量,不明显的延迟和延迟,这是实现VoIP时要考虑的重要指标。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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