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A new approach to the implementation of VoIP for SOHO network

机译:SOHO网络实现VoIP的新方法

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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.
机译:VoIP已经从基于分组的网络的进步和革命中涌现,并且在过去的几年中已经取得了巨大的进步。通常,有两种实现VoIP的方法,一种是软电话形式,另一种是专用IP电话形式。在本文中,我们描述了一种实现VoIP的新方案,该方案与这两种方法都不同,而不是像在专用IP电话中那样实现Internet的所有层,或者像在软电话中那样开发用于处理语音通信的软件应用程序在实现VoIP的过程中,我们依靠普通模拟电话(即生成语音和信令信息)和计算机应用程序(即将基于分组的数据传输到目的地)的功能。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,并且用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。我们已经使用SIP作为我们的信令协议来处理呼叫建立和拆除。我们提出的方案的原型已成功完成并在我们大学的LAN设置上进行了测试,并且给出了可接受的语音质量,明显的延迟和延迟,这是在实施VoIP时要考虑的重要指标。

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