SIP(Session Initiation Protocol)[5, 6, 7]是一種用於網路上協助應用程式建立多媒體通訊的通訊協定，其中最重要的應用之一是用於建立網路語音與視訊會議。雖然SIP已經漸漸被廣為應用，然而一但使用者處在於NAT之下時往往便無法順利建立通訊管道。以往解決這個問題的方法不外乎將語音訊息全部轉送(Relay)、修改NAT主機或是增加新的通訊協定去解決。
　　本文的目的便是在階層式SIP的網路環境(Hierarchical SIP Networks Environment)上解決使用網路位址轉換(NAT, Network Address Translation)技術所造成的語音通訊問題。在本文中將提出一個完全以SIP為基礎解決過NAT問題之方法，此方法完全摒棄以往使用STUN[8, 9, 10]、UPnP[12]或是ALG的解決方式，在完全不修改NAT也不採用額外通訊協定之下，完全利用標準SIP通訊協定去解決過NAT之問題。

　　SIP (Session Initiation Protocol) [5,6,7] is a communication protocol which is used in development multimedia communication applications, especially for VoIP applications and media conference systems. Although SIP is been broadly deployed recently, the communication behind NAT management is still a problem. It causes media data transfer failed. Some methods for solving this problems were proposed. Those are relaying all media packets, modifying NAT machine or adding some new communication protocols or components.
　　The main purpose of this thesis is to solve the voice communication problems which caused by NAT management in a hierarchical SIP communication environment. A method which only use SIP standard messages is proposed to solve the problem. The proposed method is different from that of STUN[8,9,10], UPnP[12] , ALG or modifying NAT machines.

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