基于“星号”VoIP解决方案的SIP协议研究

Lu Tian, Nicolas Dailly, Qiao Qiao, Jihua Lu, Jiannan Zhang, Jing Guo, Ji'ao Zhang
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引用次数: 6

摘要

IP语音是一种在基于IP的网络上提供语音通信服务的技术。近年来,它一直是人们关注的焦点。SIP,即会话发起协议,是一个IETF信令协议,用于文本和多媒体交换的会话管理,如VoIP、即时消息、视频、在线游戏和其他服务。各种电话应用程序和服务,如VoIP,在Asterisk中实现,这是一个免费的开源PABX。在本文中,我们重点介绍了Asterisk在本地网络中使用软电话实现正常通话、语音邮件和会议的配置。在实现的基础上,讨论了信令注册交换、呼叫建立和终止交换、DTMF交换。
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Study of SIP protocol through VoIP solution of “Asterisk”
Voice over IP is a technology that offers voice communication service over IP-based networks. It has been in a focus of much attention in recent years. SIP, the Session Initiation Protocol is an IETF signaling protocol for session management for text and multimedia exchanges, like VoIP, instant messaging, video, on-line games and other services. Various telephony applications and services, such as VoIP, are implemented in Asterisk, a free open source PABX. In this paper, we highlight the configuration of Asterisk to implement normal calls, voice mail, and conferences on a local network with soft phones. Based on the implementations, we discuss about the signaling exchanges for registering, call establishment and termination, DTMF exchanges.
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