战术互联网中VoIP实现技术研究

发布时间：2018-11-19 20:04

【摘要】：VoIP作为下一代网络的核心技术,目前其应用主要局限在固定网络领域,而战术互联网由于采用无线方式通信,使通信链路不稳定、带宽有限,因而在其上承载VoIP业务存在诸多困难。本论文以Doubango和Asterisk技术为基础,研究了恶劣通信环境下语音自适应传输技术、终端KeepAlive机制、组呼业务实现技术和视频流业务实现技术等,并最终实现了战术互联网VoIP通信系统。具体完成的工作如下:(1)介绍了战术互联网的拓扑结构和网络模型,设计了基于SIP协议的战术互联网通信系统,对战术互联网VoIP系统的信令流程和信令内容进行了分析。(2)为了能够根据网络质量调整语音流的发送速率,在对固定速率语音编码算法进行深入分析的基础上,结合RTCP的QoS反馈机制以及自适应速率语音编码模型,提出了基于网络质量的自适应语音传输算法。仿真测试表明,后者的丢包率和网络时延参数均优于前者。(3)为了实现终端设备和车载基站的KeepAlive,设计了基于PING包的终端重注册机制,解决了无线链路不稳定引起的VoIP SIP协议栈无法进行正常状态回转的问题;在不改变现有网络结构的基础上,设计并实现了组呼业务,扩展了原有的点对点通信模型;在带宽足够的场合,利用H.264视频编解码技术,实现了视频流业务,扩展了单一的语音通话功能。(4)利用Asterisk PBX平台构建了VoIP的服务器,分析了Asterisk平台的体系结构、呼叫流程和控制流程,完成了VoIP的软交换功能;利用Doubango技术设计并实现了SIP软终端,并对底层SIP协议栈的实现进行了分析。测试表明,该VoIP系统的信令正确、业务流畅、网络融合性强,适合战术互联网环境。
[Abstract]:As the core technology of the next generation network, VoIP is mainly used in the field of fixed network at present. However, because of the wireless communication mode, the communication link is unstable and the bandwidth is limited. Therefore, it is difficult to carry VoIP service on it. Based on Doubango and Asterisk technology, this paper studies the adaptive voice transmission technology, the terminal KeepAlive mechanism, the group call service realization technology and the video stream service realization technology under the bad communication environment, and finally realizes the tactical Internet VoIP communication system. The main works are as follows: (1) the topology and network model of tactical Internet are introduced, and a tactical Internet communication system based on SIP protocol is designed. The signaling flow and signaling content of tactical Internet VoIP system are analyzed. (2) in order to adjust the transmission rate of voice stream according to the network quality, the fixed-rate speech coding algorithm is deeply analyzed. Based on the QoS feedback mechanism of RTCP and the adaptive rate speech coding model, an adaptive speech transmission algorithm based on network quality is proposed. The simulation results show that both the packet loss rate and the network delay parameters of the latter are better than the former. (3) in order to realize the KeepAlive, of terminal equipment and vehicle base station, a terminal reregistration mechanism based on PING packet is designed. It solves the problem that the VoIP SIP protocol stack can not be rotated in normal state caused by the instability of wireless link. On the basis of not changing the existing network structure, the group call service is designed and implemented, and the original point-to-point communication model is extended. In the case of sufficient bandwidth, the video stream service is realized by using H.264 video coding and decoding technology, and the single voice calling function is extended. (4) the server of VoIP is constructed by using Asterisk PBX platform, and the architecture of Asterisk platform is analyzed. Call flow and control flow, completed the soft switch function of VoIP; The soft terminal of SIP is designed and implemented by using Doubango technology, and the implementation of the underlying SIP protocol stack is analyzed. The test results show that the VoIP system has the advantages of correct signalling, smooth traffic and strong network integration, which is suitable for tactical Internet environment.
【学位授予单位】：长安大学
【学位级别】：硕士
【学位授予年份】：2017
【分类号】：E919;TP393.09

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