Title: | 網際網路語音協定之行動管理與整合 The Mobility Management and Interworking of VoIP Protocols |
Authors: | 葉一均 I-Chin Yeh 張明峰 Meng-Feng Chang 資訊科學與工程研究所 |
Keywords: | 網際網路語音傳輸協定;網際網路電話服務;整合;行動管理;VoIP;Internet Telephony Service;H.323;SIP;ITDS;interworking;mobility management |
Issue Date: | 2000 |
Abstract: | 網際網路電話服務(Internet Telephony Service)提供使用者一套可以在網際網路上傳送即時資料和語音的服務。使用者可以使用傳統電話、一般連接上網的電腦或是其他通訊設備透過網際網路相互聯絡。此機制可以減少通話成本並且提供更多使用者便利性。因此如何將現行的網際網路語音傳輸協定(VoIP),如H.323、SIP、MGCP與傳統電話網路(PSTN)彼此整合便成為網際網路電話服務發展的重要因素。 網際網路語音傳輸協定H.323、SIP和MGCP是目前發展與應用最多的網際網路語音傳輸協定。MGCP提供網際網路與傳統電話間整合的介面;因此MGCP與其他網際網路語音傳輸協定如H.323或SIP的整合即提供使用者使用傳統電話網路及網際網路設備相互溝通的服務。在網際網路端,SIP與H.323的整合也提供SIP使用者與H.323使用者相互溝通的服務。整合H.323、SIP、MGCP/PSTN這三者相互溝通的能力,可以建構一個由網際網路、傳統電話網路甚至和行動電話網路組成的整合性網路。 我們將H.323、SIP、MGCP/PSTN結合在一起成為一個整合網路架構,並使用Internet Telephony Directory Service (ITDS) 架構來管理不同網路中的使用者的位址資訊。為了提供使用者行動通話能力,ITDS記錄了同一個使用者在不同網域的識別身份與狀態。當使用者移動到不同網域切換不同識別身分,通話方依然可以透過ITDS查詢到該使用者以建立通話。在整合網路架構中,我們也提出了通話交遞(handoff)的機制,讓使用者在通話中能夠移動到不同的網路通訊環境。 IP telephony provides new service for real-time data/voice communication on the Internet, which lowers the cost and potentially offers more convenience to the users. A number of VoIP protocols have been developed to harvest the advantages. One of the main issues of the IP telephony is how to integrate the VoIP protocols. VoIP standards for Internet telephone calls include H.323, SIP and MGCP. MGCP provides an interface between the PSTN and the Internet. An interworking function between MGCP and other VoIP protocols can lead to the interworking between the VoIP protocols and the PSTN. Moreover, the interworking between SIP and H.323 is also proposed to allow SIP user agent to call H.323 terminals, and vice versa. The integration of these interworking functionalities leads to a converged network of the Internet, PSTN, and even the cellular phone network. In this thesis, we present an architecture for the converged network. The architecture MGCP-SIP interworking function, SIP-H.323 interworking function, and an hierarchical user location database to support user mobility across various VoIP domains. Wherever a user moves, others can communicate with the user using the user’s public ID. We also describe how the handoff can be performed across the different domain networks using two methods, “call join” and “call transfer”. In this situation, we assume that the mobile terminal has the functionality that can change into other protocols. |
URI: | http://140.113.39.130/cdrfb3/record/nctu/#NT890392086 http://hdl.handle.net/11536/66877 |
Appears in Collections: | Thesis |