網際網路電話繞送

Abstract

近年來，由於網際網路的興起，使得網際網路電話成為傳統電話的替代方案，藉由網際網路作為電話傳遞的平台，網際網路電話提供了低價以及容易開發新服務的優勢。網際網路電話使用VoIP(Voice over Internet Protocol)作為電話交換的機制，目前主要的VoIP標準包括有ITU-T H.323 Recommendation, Media Gateway Control Protocol(MGCP)及Session Initiation Protocol (SIP)等。但是由於使用的協定不同，不同系統的網路間彼此缺乏一致性的電話繞送機制，因而降低了網際網路電話的普及率。 本論文提出了一個適用於各種不同網際網路電話協定間的繞送機制，讓不同的網際網路電話系統間能夠互通，因而增加網際網路電話的利用性。本架構支援使用者身份的可攜性(Personal Mobility)，使用者可以在不同的網際網路電話系統，使用不同的電話終端設備而不需要更動使用者的身分識別。在本篇論文中，我們亦實作了一個MGCP與H.323間的共同運作單元(Interworking Function)，證實我們的機制確實能達到在不同電話服務網路中的電話繞送。

In recent years, Internet telephony has emerged as an alternative to the traditional telephony service. By utilizing Internet as a bearer to transport telephone calls, Internet telephony offers at least two advantages over its traditional circuit-switching counterpart: lower cost and easy implementation of innovative services. Internet telephony uses VoIP (Voice over Internet Protocol) for the telephone routing mechanism. Current major VoIP standards include ITU-T H.323 Recommendation, Media Gateway Control Protocol (MGCP) and Session Initiation Protocol (SIP). Since each of these protocols provides an exclusive signaling method. This inoperability between these protocols hinders the wide deployment of the Internet telephony service. In this thesis, we propose an Internet telephony routing architecture, which provides a universal routing mechanism between different VoIP standards. The architecture also supports personal mobility. The user can access the Internet telephony service with any endpoint equipment from any service network without changing his user identifier. To demonstrate this architecture, we have implemented an Interworking Function for MGCP and H.323, and verified it using real-world application, thus proved the feasibility of the architecture.