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Study of SIP protocol through VoIP solution of "asterisk"

  • Lu Tian*
  • , Nicolas Dailly
  • , Qiao Qiao
  • , Jihua Lu
  • , Jiannan Zhang
  • , Jing Guo
  • , Ji'ao Zhang
  • *此作品的通讯作者
  • Beijing Institute of Technology
  • ESIEE
  • Beijing University of Posts and Telecommunications

科研成果: 书/报告/会议事项章节会议稿件同行评审

摘要

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.

源语言英语
主期刊名2011 Global Mobile Congress, GMC 2011
DOI
出版状态已出版 - 2011
活动2011 Global Mobile Congress, GMC 2011 - Shanghai, 中国
期限: 17 10月 201118 10月 2011

丛书

姓名2011 Global Mobile Congress, GMC 2011

会议

会议2011 Global Mobile Congress, GMC 2011
国家/地区中国
Shanghai
时期17/10/1118/10/11

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