Study of SIP protocol through VoIP solution of "asterisk"

Lu Tian*, Nicolas Dailly, Qiao Qiao, Jihua Lu, Jiannan Zhang, Jing Guo, Ji'ao Zhang

*Corresponding author for this work

Research output: Chapter in Book/Report/Conference proceedingConference contributionpeer-review

6 Citations (Scopus)

Abstract

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.

Original languageEnglish
Title of host publication2011 Global Mobile Congress, GMC 2011
DOIs
Publication statusPublished - 2011
Event2011 Global Mobile Congress, GMC 2011 - Shanghai, China
Duration: 17 Oct 201118 Oct 2011

Publication series

Name2011 Global Mobile Congress, GMC 2011

Conference

Conference2011 Global Mobile Congress, GMC 2011
Country/TerritoryChina
CityShanghai
Period17/10/1118/10/11

Keywords

  • Asterisk
  • DTMF
  • SIP
  • VoIP
  • frame exchange

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