Study of SIP protocol through VoIP solution of “Asterisk”

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.