基于安全协议VoIP系统的设计与实现

发布时间：2018-08-08 11:36

【摘要】：随着网络的迅猛发展，基于IP网络的语音传输（VoIP）技术目前已经发展成了一种专门的通信技术，其应用越来越广泛。除了服务质量等问题外，客户对VoIP网络的安全性需求也越来越高，，尤其是一些涉及敏感信息的客户，例如银行、政府部门等，面临着被窃听或者攻击的威胁。因此如何保证通信双方的安全性，也成为了VoIP领域的一个重要课题。基于上述问题，提出了基于SIP协议、SRTP协议的基本语音功能以及各种业务的安全语音通信，从而满足8048路由器语音组件在各种情况下媒体流的安全性。整个语音系统是基于Linux环境，在Comware V7平台下实现的。系统包括中央管理控制模块、号码管理模块、初始会话协商模块、媒体控制转发模块、电话驱动控制模块等模块。通过对整个VoIP系统的分析，我们需要在媒体控制转发以及初始会话协商两个模块实现上述功能。初始会话协商模块通过SIP协议协商各种不同类型的呼叫，根据语音流程，逐步给出语音协商的各种情况下的时序图，包括基本呼叫，回退呼叫，trunk（中继）功能等协商，通过在不同的SIP报文中携带各自SDP消息，协商建立会话信息。媒体控制转发模块管理RTP、RTCP报文的传输，根据SRTP协议实现对应媒体RTP、RTCP报文的加密。系统通过SIP协议以及SRTP协议，全面分析语音流程，除了满足基本的安全呼叫外，还实现支持更全面的业务，优化了协商和媒体转发，支持更多情况下的媒体流安全；并且按照非安全呼叫性能进行配置，安全呼叫的语音质量也是满足需求的。
[Abstract]:With the rapid development of network, (VoIP) technology based on IP network has been developed into a special communication technology, and its application is more and more extensive. In addition to the quality of service and other issues, the security requirements of VoIP network are increasingly high, especially some customers involved in sensitive information, such as banks, government departments and so on, are facing the threat of eavesdropping or attacking. Therefore, how to ensure the security of both sides of communication has become an important issue in the field of VoIP. Based on the above problems, the basic voice function of SIP protocol and the secure voice communication of various services are proposed to satisfy the security of the 8048 router voice component in various situations. The whole voice system is based on Linux environment and implemented on Comware V7 platform. The system includes the central management control module, the number management module, the initial session negotiation module, the media control and forwarding module, the telephone drive control module and so on. Through the analysis of the whole VoIP system, we need to realize the above functions in the media control forwarding module and the initial session negotiation module. The initial session negotiation module negotiates various kinds of calls through the SIP protocol. According to the speech flow, the sequential diagram of the speech negotiation is given step by step, including the basic call, the trunk function of the back call, and so on. By carrying their own SDP messages in different SIP messages, the session information is established through negotiation. The media control and forwarding module manages the transmission of RTCP packets. According to the SRTP protocol, the corresponding media RTCP packets are encrypted. Through SIP protocol and SRTP protocol, the system comprehensively analyzes the voice flow. Besides satisfying the basic secure call, the system also realizes more comprehensive business, optimizes negotiation and media forwarding, and supports the security of media flow in more cases. The voice quality of the secure call is also satisfied with the unsecured call performance.
【学位授予单位】：华中科技大学
【学位级别】：硕士
【学位授予年份】：2014
【分类号】：TN916.5

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