VoIP for Radio Networks
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چکیده
In the early eighties, a communications protocol was created that allowed the research community to send data anywhere in the world using packet switching networks. This protocol became known as TCP/IP and ultimately resulted in the creation of the revolutionary Internet. TCP/IP actually refers to a suite of different protocols but, at its core, there exists the fundamental protocol commonly referred to as IP. Up until a few years ago, the majority of applications for " IP " were targeted at non-voice type traffic. This included e-mail and the transfer of files. However, more recently, the ability to use IP for the transmission of digitized voice has become so dominant that many industry observers are now calling this the " killer " application. Voice over IP, or VoIP as it is known, has revolutionized the telephony world and, now, many in the radio communications industry are realizing that this technology can power traditional two-way radio networks and breathe new life into existing systems. The adaptation of VoIP for radio communications has been referred to as " Radio over IP " , although there is no actual standard, as such. Voice over IP or Radio over IP, whichever way you look at it, is driving the convergence of IT and communications. It can deliver several key benefits to fleet mobile operators, including public safety organizations. These include: lower costs, improved reliability and increased interoperability. All of these are derived from the ability of the technology to utilize existing IP infrastructure-the IP backbones that make up both local and wide area networks. To help organizations reap the benefits of VoIP, Radio over IP gateways have been developed to enable two-way analog or digital transceivers to be interfaced with IP networks. The purpose of this paper is to explain how Radio over IP works and to describe the typical features and benefits of these adapters. Radio and telephony use the same mechanisms to transport voice through packet switched data networks. Voice is converted into digital data using a device called a CODEC; assembled into UDP packets; and transmitted at intervals of around 25ms. At the receiving end, the CODEC also performs the function of converting the digitized samples back into an analog signal. Various digitization standards are available. The most basic standard (G.711) delivers the highest quality audio at 64Kbps. However, other standards, such as G.723.1 and G.729A, can deliver good quality audio …
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تاریخ انتشار 2013