| Summary: | 博士 === 國立交通大學 === 資訊科學與工程研究所 === 94 === 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.
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