無線網際網路語音服務

Abstract

透過網際網路支援語音服務或稱網際網路語音服務（voice over IP; 簡稱VoIP），已被視為電信產業必然的趨勢。網際網路語音服務能夠提供即時與低價的語音通訊，所以將網際網路語音服務加入現有的電信系統是必要的。而其中，如何將行動電話系統結合網際網路語音服務已成為重要的研究議題。在本論文中，我們提出並描述了數個提供行動用戶網際網路服務之機制。此外，我們也發展了數學分析模型探討其系統效能。 我們首先提出了一個方法，以支援全球式行動通訊系統（Global System for Mobile Communications; 簡稱GSM）用戶在使用者移動的情況下存取網際網路語音服務。在這個方法中，GSM網路和VoIP網路彼此之間相互獨立，網路元件的功能亦沒有大幅度的更動。為了支援使用者的移動性，GSM用戶在改變使用的終端機時，必須明確地執行註冊的動作，以通知系統目前行動終端機之位置。假使這個動作沒有執行，則用戶的受話很可能就會連接至錯誤的位置。我們提出了數學分析模型來探討在無線網路網路語音服務的系統中，因使用者移動所造成的受話錯接問題。接著，根據媒體閘道控制協定，我們設計了GSM-IP系統為GSM行動網路提供網路網路語音服務。在GSM-IP的架構下，GSM網路和VoIP網路彼此緊密的結合，但標準的GSM和MGCP通訊協定卻不需要更改。我們提出了GSM-IP註冊、撥話、受話、斷話和跨系統交遞程序之訊息流程，以證明其架構之可行性。 根據發展GSM系統支援網際網路語音服務的經驗，我們針對一般封包式無線電服務（General Packet Radio Service; 簡稱GPRS）網路提出了網際網路語音服務的機制，稱為vGPRS。在這個機制中，一個新的網路元件「網際網路語音行動交換中心」被提出以取代現有的GSM行動交換中心。標準的GSM和GPRS行動終端機（不需具備VoIP終端機之能力）皆可用來接受即時的網際網路語音服務。vGPRS的方法使用標準的H.323、GPRS和GSM通訊協定來實行，因此現有的GPRS和H.323網路元件不需要做任何的修改。最後，我們描述了第三代通用行動電信服務系統的all-IP方法。這個方法結合了無線與IP的技術，以提供第三代行動用戶即時的多媒體通訊服務。

Supporting telephony services over Internet Protocol (IP) network or the so called voice over IP (VoIP) is considered as a promising trend in telecommunication business. VoIP services can provide real-time and low-cost voice communications over the IP network. Thus, incorporating VoIP servicesinto the existing telecommunication system is essential. Particularly, integrating mobile phone services with VoIP becomes an mportant research issue. In this dissertation, we propose and describe several mechanisms to provide VoIP services for mobile subscribers, and develop analytic models to investigate their performance. We first propose an approach to enable Global System for Mobile Communications (GSM) subscribers to access VoIP services. In this approach, a subscriber can use a GSM mobile station (MS) or a VoIP-capable terminal. To support user mobility, when a subscriber switches from one type of terminal to another, he needs to explicitly register the terminal he uses and the location of the terminal. If the subscriber forgets to take this action, call deliveries to the subscriber may be mis-routed. We propose an analytic model to study the mis-routing problem caused by user mobility in the wireless VoIP system. Then based on the Media Gateway Control Protocol (MGCP), we propose GSM-IP, a VoIP service for GSM mobile network. In the GSM-IP architecture, the GSM and VoIP networks are tightly integrated without modifying the standard GSM and MGCP protocols. The message flows for GSM-IP registration, call origination, call delivery, call release and inter-system handoff procedures are presented to show the feasibility of integrating GSM with the MGCP-based VoIP network. Based on our experience of GSM support for VoIP , we propose vGPRS, a VoIP mechanism for General Packet Radio Service (GPRS) network. In this approach, a new network called VoIP mobile switching center (VMSC) is introduced to replace the standard GSM MSC. Both standard GSM and GPRS MSs which need not equipped with the VoIP terminal capabilities can be used to receive real-time VoIP service. The vGPRS approach is implemented using standard H.323, GPRS and GSM protocols. Thus, existing GPRS and H.323 network elements are not modified. Finally, we describe the Universal Mobile Telecommunications System (UMTS) all-IP approach for third generation (3G) mobile systems, which integrates the wireless and IP technologies to provide real-time multimedia communications for 3G subscribers.