VoIP聚合環境之行動性與互用性的支援

Abstract

隨著網路技術及終端設備的快速發展，未來的網路電話(VoIP)的環境將傾向於發展成為一個支援多類網路的聚合環境。這些網路將包含公眾交換網路(PSTN)、公用陸地行動通訊網(PLMN)、有線(Wire-line)交換網路及無線(Wireless)交換網路。但在目前，要發展這樣的聚合網路，仍存在著一些問題。例如不同通訊信令(Signaling)之間的互通、穿越位址轉換器(NAT)的問題、交遞的問題、加密金鑰的傳遞問題及計費問題都仍需克服。本篇論文重點在探討互通性及行動交遞問題。 不同網路電話信令的互通是最根本需要解決的問題，不同的機構已經發展或定義出許多不同的信令規格，例如H.323、MGCP及SIP。然而，在目前使用不同信令的電話設備仍無法互通。NAT穿越問題也會阻礙同信令或不同信令之間的互通。現今許多的設備因為缺乏實體IP位址(Public IP)，紛紛改採虛擬IP位址(Private IP)而變成隱藏在NAT之後。造成信令交換過程中，不管是通話發起或是接收端，會因無法得到正確的網路位址而致無法通訊。 再則，無線通訊技術的進展以及SIP在行動上的支援，使用者或終端設備可以在通話過程中間，改變其網路連接點而且仍能保持繼續通話。這過程稱為交遞(Handover)，它除了變更網路連接點(一般稱為Access Point，AP)之外，仍包含有網路位址的改變以及通知對方更新網路位址的信令交換。這一連串的過程相當的長久，對於VoIP這類的及時應用會讓使用者感覺到中斷，所以必須要設計一機制來加速交遞過程。 行動通訊中有一個註冊伺服器(Registrar)來記錄使用者目前所在的位置(或網路位址)。交遞時除了通訊中的彼此需要位址更新之外，更必須要將新的位址重新登錄到註冊伺服器，以便新的通話能夠再接通。然而，這個資料庫因為會經常性的被讀取及更新，有可能會產生錯誤或損毀，如此，找不到使用者的位置以致無法建立通話。 本篇論文發表了一系列的解決方案來克服上述的諸多問題。用一個簡單又富有彈性的方法，利用half-call model來簡化設計並減少因新的通訊協定的加入所需的額外修改。在NAT穿越問題上，我們將應用層閘道器(Application level gateway, ALG)的概念從NAT中分離出來，所以此方法僅需更動到Proxy，而不需修改NAT及使用者設備。我們也另外發展了一套利用服務網域的位置資訊進行快速網域交遞的協定，採用跨階層式的設計，大大減少交遞時所需花費的時間。最後，再針對位置資訊資料庫的毀損問題，我們分析了多種資料庫回復的方法，並比較其回存成本(Checkpoint)及通話斷訊成本(Lost-call)之間的關係。我們發現，資料庫的回存與否與此兩成本之間的比重是有相關的，若通話斷訊成本較高，並不需要使用到很複雜的回存機制，使用簡單的復製資料庫(Duplicated)的方法，就能達到最佳效益。

As the network and terminal technologies advance, the future Voice over IP (VoIP) environment is likely to be a converged infrastructure that consists of Public Switch Telecommunication Networks (PSTNs), Public Land Mobile Networks (PLMNs), Wire-line Packet-switched Networks, and Wireless Packet-switched Networks. However, in such VoIP converged environments, there exists several problems, such as signals interoperability, NAT traversal, handover delay, key distribution, and billing, which remain to be solved. This dissertation focus on the mobility and interoperability supports for the VoIP converged environments. The interoperability of different VoIP signaling protocols is one of the most important problems for the future VoIP converged environment. Several signaling protocols, such as H.323, SIP and MGCP, have been developed by different organizations to support VoIP communications. A device using a signaling protocol cannot operate with other devices using a different signaling protocol. NAT traversal is another interoperability problem for SIP-based VoIP applications. In a VoIP converged environment, devices may situate behind an enterprise network with an NAT router due to the lack of public IP addresses and/or the administration purpose. For a device beneath an NAT router, it cannot establish, whether it initiates the communication or not, a VoIP session with another device. Previous solutions to this NAT traversal problem require changes to the NATs and/or SIP user agents. Moreover, wireless technologies and SIP mobility make it possible for a device to change its network attachment from one point to another (henceforth referred to as handover), while retaining its VoIP session. The handoff procedure also includes sending user location update messages to both the correspondent node and the registrar for SIP-based VoIP applications. Such a handover process is considerably long and may cause serious interruption to the real-time VoIP session. Therefore a fast and smooth handover mechanism is a necessity for a VoIP converged environment. Moreover a registrar maintains the locations of VoIP users in a database, called user mobility database; users who wish to communicate with others should query the registrar to acquire the locations of the communication peers first. However, the user mobility database in a registrar may crash; causing call requests to fail. Therefore, failure recovery of the user location databases is another important issue for the mobility supports in VoIP converged environment. In this thesis, we present a series of solutions to the aforementioned problems. We first propose a simple, flexible framework for interworking gateway for different VoIP signaling protocols; the framework is based on a half-call model to reduce the design and implementation effort. For the NAT traversal problem, our method makes SIP proxies act like an application gateway and thus requires modification only to SIP proxies. Therefore our NAT traversal mechanism is more practical because it leaves NAT routers and SIP user agent programs intact. We also propose a novel topology-assisted cross-layer handover mechanism that can effectively reduce the overall handover delay of a VoIP session from several seconds to less than 120 ms. Finally, we study several user mobility database checkpoint methods and find that in most conditions the optimum checkpointing interval is either zero or infinity. That is to say, a user location record should either be always checkpointed at the registration, or be never checkpointed at all, depending on the weighting factor of checkpointing cost and that of lost-call cost.