The Implementation of Real-Time Recording, Monitoring, and Analyzing System for SIP-based Internet Phone

碩士 === 國立臺北科技大學 === 資訊工程系研究所 === 98 === With the progress on communication technology and Internet popularity, various on-line applications and services are supported and operated well. Among those, Internet phone (i.e. Voice over IP, VoIP) is almost to be a killer application at present. In terms...

Full description

Bibliographic Details
Main Authors: Chien-Zhi Wang, 王謙志
Other Authors: 柯開維
Format: Others
Language:zh-TW
Published: 2010
Online Access:http://ndltd.ncl.edu.tw/handle/4yvagw
Description
Summary:碩士 === 國立臺北科技大學 === 資訊工程系研究所 === 98 === With the progress on communication technology and Internet popularity, various on-line applications and services are supported and operated well. Among those, Internet phone (i.e. Voice over IP, VoIP) is almost to be a killer application at present. In terms of control protocol of VoIP, H.323 proposed by ITU-T (International Telecommunication Union-Telecommunication) and SIP (Session Initiation Protocol) given by IETF (Internet Engineering Task Force) receive the most attention from industry. Currently, SIP seems getting more favor than H.323. The objective of the thesis is to design a monitoring and recording system for SIP based internet phone. It’s based upon JAVA-language incorporating with several JAVA libraries such as packet capture (JPcap) and multimedia framework (JMF). This work includes packet capture and buffer management, SIP/RTP/TCP/UDP/IP protocol analysis, voice packets decoding, on-going conversation monitoring, voice call recording and replaying, call history and statistic, and graphic user interface for status and system operation. All functions were justified through three call types, point-to-point, transfer, and conference, and combined stress test. The results present the design meets the requirements we set in advance and operates well.