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首页> 外文期刊>International Journal of Wireless Communications and Mobile Computing >Study SIP Protocol on Asterisk Phone System and Offer Solutions to Its Security
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Study SIP Protocol on Asterisk Phone System and Offer Solutions to Its Security

机译:研究Asterisk电话系统上的SIP协议并为其安全性提供解决方案

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Undoubted every organization's heart is it's phone system. The old phone systems couldn't perform any method to make phone center and voice transmission intelligent on network and they had determinate abilities. Meantime Voice over Internet Protocol (VoIP) introduced itself to the world and performed a lot of abilities for clients like voice transmission on network. Many companies Investment on voip systems and implemented their methods on software and hardware packages. But between them a different production had designed and Performanced by Mark Spenser from Digium Company in 1992 which named Asterisk. Asterisk's increasing popularity's reason was its open code programs and its flexibility. VoIP systems such as Asterisk use voice transmission protocols to transfer voice over network. One of the voice transmission protocols is Session Initiation Protocol (SIP), which is one of the Asterisk's voice transmission protocols. The first and the most important point in voice transmission over network is security. Security can be divided to two parts as inscrutability of invaders to network and coding transmitted voices over network to prevention of illegal listening. In this project at first we tried to introduce Asterisk phone system's structure and Session Initiation Protocol (SIP) and then scrutiny method of Invader's dominance to this protocol and Performance modern methods to prevent hacker's dominance and also coding voice packages to Obscure them in transmission way.
机译:毫无疑问,每个组织的内心都是电话系统。旧的电话系统无法执行任何使网络上的电话中心和语音传输智能化的方法,并且具有确定的功能。同时,互联网协议语音(VoIP)向世界介绍了自己,并为客户端执行了许多功能,例如网络上的语音传输。许多公司在voip系统上进行投资,并在软件和硬件程序包上实施其方法。但是他们之间的另一种产品是由Digium公司的Mark Spenser在1992年设计和生产的,名为Asterisk。 Asterisk越来越受欢迎的原因是其开放代码程序及其灵活性。诸如Asterisk之类的VoIP系统使用语音传输协议在网络上传输语音。语音传输协议之一是会话启动协议(SIP),它是Asterisk的语音传输协议之一。通过网络进行语音传输的第一个也是最重要的一点是安全性。安全性可以分为两部分,即入侵者难以进入网络以及对网络上传输的语音进行编码以防止非法收听。在本项目中,我们首先尝试介绍Asterisk电话系统的结构和会话发起协议(SIP),然后研究Invader对该协议的支配地位的审查方法以及性能现代方法以防止黑客的支配地位,并对语音包进行编码以使其在传输方式中变得模糊。

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