Study of SIP protocol through VoIP solution of “Asterisk”

Author

Tian, Lu ; Dailly, Nicolas ; Qiao, Qiao ; Lu, Jihua ; Zhang, Jiannan ; Guo, Jing ; Zhang, Ji´ao

Author_Institution

Sch. of Inf. & Electron., Beijing Inst. of Technol., Beijing, China

fYear

2011

fDate

17-18 Oct. 2011

Firstpage

1

Lastpage

5

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.

Keywords

IP networks; Internet telephony; signalling protocols; voice communication; Asterisk; IETF signaling protocol; IP-based networks; SIP protocol; VoIP solution; multimedia exchanges; session initiation protocol; session management; voice communication; voice over IP; IP networks; Internet; Multimedia communication; Protocols; Servers; Telephony; Asterisk; DTMF; SIP; VoIP; frame exchange;

fLanguage

English

Publisher

ieee

Conference_Titel

Mobile Congress (GMC), 2011 Global

Conference_Location

Shanghai

Print_ISBN

978-1-4673-0346-0

Type

conf

DOI

10.1109/GMC.2011.6103925

Filename

6103925