- 學位論文
以SIP Phone為基礎之跨平台側錄監聽與分析系統
The Implementation of Real-Time Recording, Monitoring, and Analyzing System for SIP-based Internet Phone
摘要
在現代網際網路蓬勃發展的趨勢下，眾多線上服務也隨之盛行，而網路電話正是其中之一的殺手級應用服務。在現今的環境下，已逐漸變成生活中不可缺少的重要功能。目前最為主要及普遍的網路電話協定，分別是ITU-T(International Telecommunication Union-Telecommunication)組織提出的H.323協定和IETF(Internet Engineering Task Force)組織所提出的SIP(Session Initiation Protocol)協定。 本論文旨在設計與實作一個以SIP網路電話為基礎之錄音與監聽系統，它是以JAVA程式語言為基礎，配合JPcap與JMF實作此系統。本研究藉由捕捉網路上封包，找出其中屬於SIP的訊號，將封包內的資訊進行分析後，上傳至資料庫，達到監看與分析每一部電話在網路上行為的目的。最後將RTP (Real-time Transport Protocol)封包的負載(Payload)取出並且解碼，將聲音內容重組還原，以提供我們隨時進行還原每通電話內容的動作。
並列摘要
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.