<p>SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints.</p>

by Yip Chung Sun Danny. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2000. / Includes bibliographical references (leaves 52-54). / Abstracts in English and Chinese. / Chapter Chapter 1 --- Introduction --- p.1 / Chapter 1.1 --- Motivation --- p.1 / Chapter 1.2 --- Organization of Thesis --- p.5 / Chapter Chapter 2 --- Background --- p.6 / Chapter 2.1 --- Speech Codec --- p.6 / Chapter 2.2 --- RTP/UDP/IP Header Compression --- p.7 / Chapter 2.2.1 --- Real-Time Transport Protocol --- p.7 / Chapter 2.2.2 --- RTP/UDP/IP Header Compression --- p.8 / Chapter Chapter 3 --- Scenario and Assumptions --- p.10 / Chapter Chapter 4 --- Asynchronous Time Division Multiplexing Scheme --- p.14 / Chapter 4.1 --- Basic Idea --- p.14 / Chapter 4.1.1 --- Bandwidth Efficiency Improvement --- p.16 / Chapter 4.1.2 --- Delay Reduction --- p.18 / Chapter 4.2 --- Header Compression --- p.19 / Chapter 4.2.1 --- Header Compression Process --- p.21 / Chapter 4.2.2 --- Context Mapping Table --- p.23 / Chapter 4.3 --- Protocol --- p.28 / Chapter 4.3.1 --- UNCOMPRESSED_RTP Mini-Header --- p.30 / Chapter 4.3.2 --- SYNCHRONIZATION Mini-header --- p.31 / Chapter 4.3.3 --- COMPRESSED´ؤRTP Mini-header --- p.32 / Chapter 4.4 --- Connection Establishment --- p.33 / Chapter 4.4.1 --- Addressing Phase --- p.34 / Chapter 4.4.2 --- Connection Phase --- p.36 / Chapter 4.5 --- Software Implementation --- p.38 / Chapter Chapter 5 --- Simulation Results --- p.39 / Chapter 5.1 --- Simulation Model --- p.39 / Chapter 5.2 --- Voice Source Model --- p.41 / Chapter 5.3 --- Simulation Results --- p.43 / Chapter 5.3.1 --- Network Utilization and Delay Performance --- p.43 / Chapter 5.3.2 --- Number of Supported Connections --- p.45 / Chapter Chapter 6 --- Conclusion and Future Work --- p.49 / Bibliography --- p.52

Ho Ho-ching. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2001. / Includes bibliographical references (leaves 104-107). / Abstracts in English and Chinese. / Abstract --- p.i / Acknowledgment --- p.vi / Chapter 1. --- Introduction --- p.1 / Chapter 1.1. --- Introduction to PBX based Call Center --- p.1 / Chapter 1.2. --- The Scenarios --- p.2 / Chapter 1.3. --- Thesis Overview --- p.5 / Chapter 2. --- Features of Collaborative Browsing --- p.8 / Chapter 2.1. --- Feature Synchronization vs Bitmap Screen Transfer --- p.8 / Chapter 2.2. --- Basic Collaborative Features of the Collaborative Browser --- p.9 / Chapter 2.2.1. --- Web Page Pushing --- p.10 / Chapter 2.2.2. --- Screen Widget Synchronization --- p.11 / Chapter 2.2.3. --- Tele-pointing and Shared Whiteboard --- p.12 / Chapter 2.3. --- Collaborative Form Manipulation --- p.13 / Chapter 2.3.1. --- Importance of Electronic Form Collaboration --- p.13 / Chapter 2.3.2. --- Basic Support for Form Collaboration ´ؤ´ؤData Synchronization --- p.14 / Chapter 2.3.3. --- Existence of Form Complexity ´ؤForm Data Dependency --- p.75 / Chapter 2.3.4. --- Hinting --- p.17 / Chapter 2.4. --- Collaborative IVR --- p.23 / Chapter 2.4.1. --- Traditional Interactive Voice Response (IVR) Service --- p.23 / Chapter 2.4.2. --- Abstraction ´ؤCorrelating Form Interaction Mechansim with IVR --- p.25 / Chapter 2.4.3 --- Collaborative IVR by Form Interaction Mechanism --- p.27 / Chapter 3. --- Software Architecture --- p.33 / Chapter 3.1. --- The Three-Tier Architecture --- p.33 / Chapter 3.2. --- The Collaboration Mechanism for Collaborative Browser --- p.37 / Chapter 3.2.1. --- Session Initialization/Termination --- p.37 / Chapter 3.2.2. --- Data Flow of the Basic Collaboration Features --- p.39 / Chapter 3.2.3. --- Control Mechanism --- p.40 / Chapter 3.2.4. --- The Hinting Mechanism for Collaborative Form Manipulation --- p.43 / Chapter 3.3. --- The Collaboration Mechanism for Collaborative IVR --- p.45 / Chapter 4. --- Implementation --- p.51 / Chapter 4.1. --- Shareable Document Object Architecture for Collaboration --- p.51 / Chapter 4.1.1. --- Document Object Architecture --- p.51 / Chapter 4.1.2. --- Generalizing to Shareable Document Object Architecture --- p.53 / Chapter 4.2. --- Whiteboard Mechanism --- p.55 / Chapter 4.3. --- Packet Data Unit for Communication --- p.57 / Chapter 4.4. --- Bridging Different Software Components --- p.60 / Chapter 4.5. --- Hinting Mechanism for Collaborative Form Manipulation --- p.63 / Chapter 4.5.1. --- Relating Form Fields to Table Fields --- p.63 / Chapter 4.5.2. --- Hinting by the Hinting Tables --- p.69 / Chapter 4.6. --- Collaborative IVR --- p.73 / Chapter 4.6.1. --- Using Mediator for Collaborative IVR --- p.73 / Chapter 4.6.2. --- Concept of Telephone Form --- p.74 / Chapter 4.6.3. --- Hinting for Collaborative IVR --- p.78 / Chapter 4.7. --- System Integration --- p.81 / Chapter 5. --- Performance Evaluation and Experiment Results --- p.84 / Chapter 5.1. --- Optimizing the Transmission Methodology --- p.84 / Chapter 5.2. --- Browser Responsiveness Study --- p.86 / Chapter 5.2.1. --- Experiment Details --- p.86 / Chapter 5.2.2. --- The Assumptions --- p.89 / Chapter 5.2.3. --- Experiment Results and Analysis --- p.90 / Chapter 5.3. --- Bandwidth Consumption --- p.94 / Chapter 6. --- Conclusions --- p.97 / Appendix A 一 Government Profit Tax Return Form --- p.101 / Appendix B ´ؤ A Phone Banking IVR Service Tree --- p.103 / Bibliography --- p.104

The IP Multimedia Subsystem (IMS) is the technology that will merge the Internet (packet switching) with the cellular world (circuit switching). It will make Internet technologies, such as the web, email, instant messaging, presence, and videoconferencing available nearly everywhere. Presence is one of the basic services that is likely to become omnipresent in IMS. It is the service that allows a user to be informed about the reachability, availability, and willingness of communication of another user. Push to talk over Cellular (PoC) is another service in IMS that is intended to provide rapid communications for business and consumer customers of mobile networks. In order to become a truly successful mass-market service for the consumer segment, the only realistic alternative is a standardized Push-to-talk solution providing full interoperability between terminals and operators. Instant Messaging (IM) is the service that allows an IMS user to send some content to another user in near real-time. This service works under IETF’s Message Session Relay protocol (MSRP) to overcome the congestion control problem. We believe the efficiency of these services along with the mobility management in IMS session establishment has not been sufficiently investigated.In this research work, we identify the key issues to improve the existing protocols in IMS for better system behaviour. The work is centred on the three services of IMS: (1) Presence Service, (2) Push-to-Talk over cellular and, (3) Instant Messaging and over the issue of (4) IMS session set up. The existing session establishment scenario of IP Multimedia Subsystem (IMS) suffers from triangular routing for a certain period of time when an end IMS user or terminal is mobile. In this thesis, the performance of three possible session establishment scenarios in a mobile environment is compared by using an analytical model. The model is developed based on the expressions of cost functions, which represents system delay and overhead involved in sessions’ establishment. The other problem areas in optimizing presence service, dimensioning a PoC service and analysing service rates of IM relay extensions in IMS are identified. A presence server becomes overloaded when massive number of IMS terminals joins a network to request presence facility. Performance models are developed in this research to mitigate such load during heavy traffic for the presence service. Queuing analyses for different cases are provided while instant messaging chunks go through two consecutive relay nodes. The specific factors such as blocking probability, stability conditions, optimized subscription lifetime etc. in IMS environment parameters have been investigated. We have also elaborated models to dimension a PoC service for service providers with regards to controlling PoC session access, optimal PoC session timer, path optimization and number of allowable simultaneous PoC sessions for given network grade of service.In a nutshell, the contribution of this dissertation are: (a) a proposed robust scheduler to improve performance of the IMS presence service, (b) several derived models to dimension IMS Push-to-talk over cellular service, (c) a new mechanism to reduce cost for the IMS session set ups in mobile environment and (d) evaluation of message blocking and stability in IMS Instant Messaging (IM) service by applying queuing theories. All of these analyses have resulted in recommendations for the performance enhancements with optimal resource utilization in IMS framework.

Telecommunication systems have evolved significantly and the recent convergence of telephony allows users to communicate through landlines, mobile phones and Voice over IP (VoIP) phones. Unfortunately this convergence has resulted in easy manipulation of caller identity, resulting in both VoIP spam and Caller ID spoofing. In this dissertation, we introduce the notion of effective identity which is a combination of mechanisms to (1) establish identity of the caller that is harder to manipulate, and (2) provide additional information about the caller. We first use effective identities to address the VoIP spam problem by proposing CallRank, a novel mechanism built around call duration and social network linkages to differentiate between a legitimate user and a spammer. To ensure that this mechanism is privacy preserving, we create a token framework that allows a user to prove the existence of a social network path between him and the user he is trying to initiate contact with, without actually revealing the path. We then look at the broader issue of determining identity across the entire telecommunication landscape to address Caller ID spoofing. Towards this, we develop PinDr0p, a technique to determine the provenance of a call - the source and the path taken by a call. In the absence of any verifiable metadata, provenance offers a means of uniquely identifying a call source. Finally, we use anomalies in timbre to develop London Calling, a mechanism to identify geography of a caller. Together, the contributions made in this dissertation create effective identities that can help address the new threats in a converged telecommunication infrastructure.

Network Address Translation (NAT) devices are common, particularly in the small-offiee/home-office environment. Typically, a NAT device does not disrupt the oper-ation of a client-server application. even if the application is unaware of the device's presence. Unfortunately, the same does not hold for peer-to-peer applications. such as Voice over IP (VoIP). This thesis investigates the compatibility of VoIP and NAT by making VoIP calls across various NAT device types. Three different test methods are used. differing in the VoIP phone that is used and the degree of automation that is employed. Moreover, the VoIP phones are tested in different configurations, which results in significantly altered traffic and vastly different test results. All test results are presented in detail. along with explanations as to why specific phone/NAT-type combinations succeed or fail.

Location-based service (LBS) is a geographical location-related service that provides highly personalized services for users. It is a platform for network operators to provide new and innovative ways of increasing profits from new services. With the rapidly growing trend toward LBS, there is a need for standard LBS protocols. This thesis started with introducing the Internet Engineering Task Force GEOPRIV working group, which endeavors to provide standard LBS protocols capable of transferring geographic location information for diverse location-aware applications. Through careful observation, it was found that Session Initiation Protocol (SIP) is well suited to the GEOPRIV requirements. The aim of this research was therefore to explore the possibility of the integration of LBS and the SIP protocol and, to some extent fulfill the GEOPRIV requirements.

<p>SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints.</p>

The IP Multimedia Subsystem (IMS) is the technology that will merge the Internet (packet switching) with the cellular world (circuit switching). It will make Internet technologies, such as the web, email, instant messaging, presence, and videoconferencing available nearly everywhere. Presence is one of the basic services that is likely to become omnipresent in IMS. It is the service that allows a user to be informed about the reachability, availability, and willingness of communication of another user. Push to talk over Cellular (PoC) is another service in IMS that is intended to provide rapid communications for business and consumer customers of mobile networks. In order to become a truly successful mass-market service for the consumer segment, the only realistic alternative is a standardized Push-to-talk solution providing full interoperability between terminals and operators. Instant Messaging (IM) is the service that allows an IMS user to send some content to another user in near real-time. This service works under IETF’s Message Session Relay protocol (MSRP) to overcome the congestion control problem. We believe the efficiency of these services along with the mobility management in IMS session establishment has not been sufficiently investigated.In this research work, we identify the key issues to improve the existing protocols in IMS for better system behaviour. The work is centred on the three services of IMS: (1) Presence Service, (2) Push-to-Talk over cellular and, (3) Instant Messaging and over the issue of (4) IMS session set up. The existing session establishment scenario of IP Multimedia Subsystem (IMS) suffers from triangular routing for a certain period of time when an end IMS user or terminal is mobile. In this thesis, the performance of three possible session establishment scenarios in a mobile environment is compared by using an analytical model. The model is developed based on the expressions of cost functions, which represents system delay and overhead involved in sessions’ establishment. The other problem areas in optimizing presence service, dimensioning a PoC service and analysing service rates of IM relay extensions in IMS are identified. A presence server becomes overloaded when massive number of IMS terminals joins a network to request presence facility. Performance models are developed in this research to mitigate such load during heavy traffic for the presence service. Queuing analyses for different cases are provided while instant messaging chunks go through two consecutive relay nodes. The specific factors such as blocking probability, stability conditions, optimized subscription lifetime etc. in IMS environment parameters have been investigated. We have also elaborated models to dimension a PoC service for service providers with regards to controlling PoC session access, optimal PoC session timer, path optimization and number of allowable simultaneous PoC sessions for given network grade of service.In a nutshell, the contribution of this dissertation are: (a) a proposed robust scheduler to improve performance of the IMS presence service, (b) several derived models to dimension IMS Push-to-talk over cellular service, (c) a new mechanism to reduce cost for the IMS session set ups in mobile environment and (d) evaluation of message blocking and stability in IMS Instant Messaging (IM) service by applying queuing theories. All of these analyses have resulted in recommendations for the performance enhancements with optimal resource utilization in IMS framework.

Thesis (Ph. D.)--University of Canterbury, 2008. / Typescript (photocopy). Includes bibliographical references (p. 193-205). Also available via the World Wide Web.