标题: 网际网路语音协定之行动管理与整合
The Mobility Management and Interworking of VoIP Protocols
作者: 叶一均
I-Chin Yeh
张明峰
Meng-Feng Chang
资讯科学与工程研究所
关键字: 网际网路语音传输协定;网际网路电话服务;整合;行动管理;VoIP;Internet Telephony Service;H.323;SIP;ITDS;interworking;mobility management
公开日期: 2000
摘要: 网际网路电话服务(Internet Telephony Service)提供使用者一套可以在网际网路上传送即时资料和语音的服务。使用者可以使用传统电话、一般连接上网的电脑或是其他通讯设备透过网际网路相互联络。此机制可以减少通话成本并且提供更多使用者便利性。因此如何将现行的网际网路语音传输协定(VoIP),如H.323、SIP、MGCP与传统电话网路(PSTN)彼此整合便成为网际网路电话服务发展的重要因素。
网际网路语音传输协定H.323、SIP和MGCP是目前发展与应用最多的网际网路语音传输协定。MGCP提供网际网路与传统电话间整合的介面;因此MGCP与其他网际网路语音传输协定如H.323或SIP的整合即提供使用者使用传统电话网路及网际网路设备相互沟通的服务。在网际网路端,SIP与H.323的整合也提供SIP使用者与H.323使用者相互沟通的服务。整合H.323、SIP、MGCP/PSTN这三者相互沟通的能力,可以建构一个由网际网路、传统电话网路甚至和行动电话网路组成的整合性网路。
我们将H.323、SIP、MGCP/PSTN结合在一起成为一个整合网路架构,并使用Internet Telephony Directory Service (ITDS) 架构来管理不同网路中的使用者的位址资讯。为了提供使用者行动通话能力,ITDS记录了同一个使用者在不同网域的识别身份与状态。当使用者移动到不同网域切换不同识别身分,通话方依然可以透过ITDS查询到该使用者以建立通话。在整合网路架构中,我们也提出了通话交递(handoff)的机制,让使用者在通话中能够移动到不同的网路通讯环境。
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
显示于类别:Thesis