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基于SIP协议的VoIP网络电话的研究与实现

Research and Implemention of VoIP Client Based on SIP Signaling Protocol

【作者】 周桦

【导师】 胡晨;

【作者基本信息】 东南大学 , 微电子与固体电子学, 2005, 硕士

【摘要】 越来越庞大的互联网,给人们带来了更加快捷的数据传输方式。而且信息化进程的推进,电信业的发展,促进了互联网各种增值业务的发展,使得在网络上传输语音成为可能。这样的技术逐步成为了一项热点,它就是网络电话。它以资费低廉,维护成本低等诸多优点,越来越受到各方的关注。目前,使用网络电话的人数越来越多,人们对于网络电话终端的要求也越来越高。人们不单需要接入网络的电脑作为终端,也需要低成本的终端解决方案。所以,对于网络电话的终端进行研究,有较为实用的现实意义。网络电话技术,一般称为VoIP技术,即Voice over Internet Protocol。它不是单一的一种技术,而是综合了通讯网络的多种技术在内的一个系统,包括了信令协议、语音编解码、媒体传输和网关控制。SIP,Session Initiation Protocol,是信令协议的一种。它是应用层的协议。相对于另一种信令协议H.323来说,SIP具有建立呼叫过程简单,文本消息更加直观等优点,正被越来越多的网络电话设备供应商多支持。本课题的主要目标,就是在对SIP信令协议进行研究的基础上,对SIP的实现进行尝试。然后,设计一个基于SIP信令协议的网络电话的客户端的原型,以简单实现通过SIP信令协议进行呼叫,在局域网中实现VoIP功能的网络电话客户端。本文的主要内容是,首先对VoIP技术的原理进行了概述,分析了VoIP技术与传统的电话交换技术之间的差别,并对组成VoIP系统的四个部分做了简单描述。接着,通过对SIP协议进行了研究,以及对oSIP进行了研究与分析,在oSIP基础之上形成了一组API函数以实现SIP UA的各种功能。然后,对网络电话终端进行了设计,使用Motorola ColdFire5272搭建了目标系统,使用uClinux作为目标系统的软件平台,使用oSIP以及形成的一组API函数作为信令协议的实现,使用OKI ML7074实现语音的编解码,使用oRTP实现了媒体的传输。最后,通过搭建具体的实验环境,对信令协议的实现及终端进行功能性测试。并在文章的最后对本课题的结果进行了总结与展望,提出了工作中的不足之处。

【Abstract】 Internet offers people an easy way to transfer data. And with the development of information evolution and telecom industry, value added services are stimulated. It is possible now that transfer voice data through Internet. It is Internet Phone. It becomes a hot spot. Internet Phone attracts more and more attentions for its virtues like low fees and low maintain cost. At present, more and more people adopt Internet Phone to communicate with others. Internet Terminal is demanding. Not only computer based terminals, but also low cost terminals are needed by people. So it has practical sense investigating Internet Phone Terminal.Internet Phone technology is called VoIP technology, viz. Voice over Internet Protocol. It is not a single but a system integrated with several technologies constructing communication network, including Signaling Protocol, Voice Codec, Media Transport and Gateway Control. SIP, Session Initiation Protocol, is one of the Signaling Protocols. It is in application layer. Compared with another Signaling Protocol H.323, it has virtues such as more simple in call flow and text messages. And it is taken into use by more and more equipment provider.The main object is trying to implement SIP stack. Then, design a prototype of voip client based on the stack, which can exchange sip messages with other clients.It this article, firstly, on the base of research of SIP protocol and oSIP, a set of API functions was designed to implement SIP UA functions. Then, a Internet Phone Terminal was designed. Motorola ColdFire 5272 was used to construct target system. uClinux was used as OS. oSIP and the API sets was used to implement SIP. OKI ML7074 was used to implement Voice Codec. At last, an experiment was constructed to validate the terminal functions. At the bottom, summary was bought up.

【关键词】 VoIP网络电话SIPoSIPuClinuxoRTP
【Key words】 VoIPInternet PhoneSIPoSIPuClinuxoRTP
  • 【网络出版投稿人】 东南大学
  • 【网络出版年期】2007年 04期
  • 【分类号】TN916.2;TP393.4
  • 【被引频次】11
  • 【下载频次】809
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