標題: 以SIP為基礎提供合法監聽之功能
Lawful Interception Based on SIP
作者: 彭俊豪
Chung-Hao Peng
蔡文能
Wen-Nung Tsai
資訊科學與工程研究所
關鍵字: 網路電話;安全;合法監聽;VoIP;security;Lawful Interception
公開日期: 2004
摘要: 隨著網際網路的成長,在網路上的應用也變得越來越多,除了網頁的瀏覽(HTTP)、資料的傳輸(FTP)、信件的傳送(SMTP)這些基本之外,網路電話(VoIP)是近幾年快速興起的應用,有取代傳統電話的趨勢。網路電話指的是將通話語音經由IP網路封包交換技術傳送的新電話服務,跟傳統電話電路交換技術完全不同,因此網路電話有一些特性是傳統電話所沒有的,還有許許多多的優點:通話費便宜、可以直接在電腦上使用、使用方便等等,因此網路電話可說是商機無限。 網路電話有許多好處,也有一些問題需要克服,例如通話品質(QoS)、穿透防火牆或是NAT等等。當VoIP想要商業化並在市場上開始提供通話服務時,都會被國家的法律要求提供犯罪者的電話監聽,但是目前VoIP無法提供合法監聽的功能,傳統PSTN和GSM已經有一套正在運行的架構可以讓司法單位監聽使用傳統電話和行動電話通話的使用者,但是網路電話目前還無法達到此目的,因此在這裡希望可以提出一套讓VoIP有合法監聽功能的架構。 VoIP在建立通話時常用到的標準協定是SIP這種網路協定,這種協定可以建立一通網路電話,修改一通網路電話,或是結束一通網路電話,因此這個合法監聽是以SIP網路協定為基礎來提出一個架構,另外合法監聽有一些基本的要求,例如不能讓監聽目標發現受到監聽等等,這個架構也要能符合這些要求。
With the rapid growth of the internet, applications for the internet has become more and more popular, including HTTP, FTP, SMTP, and VoIP. Voice over IP(VoIP) is the application growing fast in recent years. It has the tendency to replace the traditional telephone. VoIP is the routing of voice conversations over the IP network. The voice data flows over a packet-switched network instead of a circuit-switched network. This brings a lot of advantages: low cost, convenient, and easy to use on the computer. Therefore, VoIP has many opportunities in commerce. Despite of those advantages, VoIP is still facing some challenges, such as, quality of service(QoS) , traversing firewalls and NAT ,etc. For VoIP to be commercialized, it must support Lawful Interception which the Law Enforcement Agency of the country asks for. The conventional PSTN and GSM can support Lawful Interception for LEA, but VoIP does not yet. In this thesis, we proposed an Architecture for VoIP to support Lawful Interception. The standard protocol which is often used in VoIP is the Session Initial Protocol(SIP). The SIP protocol can be used to set up, modify, and terminate an Internet telephone call. The LI-supported VoIP architecture we proposed in this thesis is based on SIP. A Lawful Interception system must fulfill certain requirements. In this thesis, we will illustrate how our VoIP architecture meets these requirements.
URI: http://140.113.39.130/cdrfb3/record/nctu/#GT009117592
http://hdl.handle.net/11536/50324
顯示於類別:畢業論文


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