Thesis (MTech (Electrical Engineering))--Cape Technikon, Cape Town, 2004 / This work describes research towards the development of a micro-controller based, standalone Internet telephone to be used as an alternative to conventional line telephones. Our definition of 'stand-alone' refers to the unit's capability to perform its function wholly without the need for an attached computer. The unit should be low cost and capable of allowing two users to communicate using the units. Bandwidth usage should be kept low to allow the unit to be used over dial up connections which are prevalent in South Africa. The units should be easy to use as the anticipated users may be unskilled. A module containing a 16-bit micro-controller, an Ethernet controller, flash memory and RAM was chosen as the controller. The module came with a real-time operating system and a TCPlIP stack. The session initiation protocol (SIP) was selected to perform the signalling. SIP uses the session description protocol (SDP) to negotiate the attributes of the media session to be established. The real-time transport protocol (RTP) was implemented to transport encoded audio between the end points. The RTP control protocol (RTCP) was implemented to provide basic quality of service parameters. The ITU-T recommendation G.729 annex A was the voice codec selected. Codec ICs were used to encode and decode the audio. The implementations were designed specifically for a two user, direct communication environment. That is two phone units were developed that communicated directly with each other and not through intermediary servers.

As Voice over IP becomes more prevalent, value-adds to the service will become ubiquitous. Voice over IP (VoIP) is no longer a single service application, but an array of marketable services of increasing depth, which are moving into the non-desktop market. In addition, as the range of devices being generally used increases, it will become necessary for all services, including VoIP services, to be accessible from multiple platforms and through varied interfaces. With the recent introduction and growth of the open source software PBX system named Asterisk, the possibility of achieving these goals has become more concrete. In addition to Asterisk, a number of open source systems are being developed which facilitate the development of systems that interoperate over a wide variety of platforms and through multiple interfaces. This thesis investigates Asterisk in terms of its viability to provide the depth of services that will be required in a VoIP environment, as well as a number of other open source systems in terms of what they can offer such a system. In addition, it investigates whether these services can be made available on different devices. Using various systems built as a proof-of-concept, this thesis shows that Asterisk, in conjunction with various other open source projects, such as the Twisted framework provides a concrete tool which can be used to realise flexible and protocol independent telephony solutions for a small to medium enterprise.

<p>This thesis explored how hearing and deaf users are brought together into one communication space where interaction between them is a semi-synchronous form of message exchange. The focus of this thesis was the means by which message delivery between two e</p>

Chui Sai Kit. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2006. / Includes bibliographical references (leaves 73-77). / Abstracts in English and Chinese. / Abstract --- p.i / Acknowledgement --- p.iii / Chapter 1 --- Introduction --- p.1 / Chapter 1.1 --- Introduction --- p.1 / Chapter 1.2 --- Wireless LAN --- p.3 / Chapter 1.2.1 --- Ad Hoc Mode --- p.3 / Chapter 1.2.2 --- Infrastructure Mode --- p.3 / Chapter 1.3 --- Handoff --- p.4 / Chapter 1.3.1 --- IP Layer Handoff --- p.5 / Chapter 1.3.2 --- MAC Layer Handoff --- p.6 / Chapter 1.4 --- Voice over Internet Protocol (VoIP) --- p.6 / Chapter 1.5 --- Significance of Research Outcomes --- p.8 / Chapter 1.6 --- Outline of Thesis --- p.10 / Chapter 2 --- Background Study --- p.11 / Chapter 2.1 --- Handoff Process --- p.12 / Chapter 2.2 --- MAC Layer Handoff --- p.12 / Chapter 2.2.1 --- MAC Layer Handoff Process --- p.12 / Chapter 2.2.2 --- MAC Layer Handoff Scheme --- p.16 / Chapter 2.3 --- IP Layer Handoff --- p.20 / Chapter 2.3.1 --- IP Layer Handoff Process --- p.20 / Chapter 2.3.2 --- IP Layer Handoff Scheme --- p.22 / Chapter 2.4 --- Chapter Summary --- p.25 / Chapter 3 --- AP Coordination System and Performance Analysis for Sync-Scan --- p.26 / Chapter 3.1 --- Introduction --- p.26 / Chapter 3.2 --- Problem Formulation --- p.27 / Chapter 3.3 --- Fast Handoff Scheme --- p.27 / Chapter 3.3.1 --- Access Point Coordination System --- p.28 / Chapter 3.3.2 --- Simulation Results --- p.30 / Chapter 3.3.3 --- Further Discussion --- p.33 / Chapter 3.3.4 --- Improved Handoff Process --- p.34 / Chapter 3.4 --- SyncScan Performance Analysis --- p.36 / Chapter 3.4.1 --- Beacon Delay --- p.36 / Chapter 3.4.2 --- Handoff Latency --- p.38 / Chapter 3.5 --- Chapter Summary --- p.41 / Chapter 4 --- Handoff Control Message Analysis --- p.43 / Chapter 4.1 --- Introduction --- p.43 / Chapter 4.2 --- Problem Formulation --- p.44 / Chapter 4.3 --- Key System Parameters --- p.45 / Chapter 4.4 --- System Model --- p.47 / Chapter 4.4.1 --- Markov Modulated Poisson Process (MMPP) Model --- p.47 / Chapter 4.4.2 --- System Time Distribution --- p.52 / Chapter 4.5 --- Performance Analysis --- p.58 / Chapter 4.6 --- Further Discussion --- p.63 / Chapter 4.6.1 --- Handoff Scheme Strategy --- p.63 / Chapter 4.6.2 --- Channel Reservation for Handoff Process --- p.66 / Chapter 4.7 --- Chapter Summary --- p.68 / Chapter 5 --- Conclusion --- p.70 / Bibliography --- p.73

Yuen Ka-nang. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2001. / Includes bibliographical references (leaves 133-135). / Abstracts in English and Chinese. / Abstract --- p.i / Acknowledgement --- p.iii / Chapter 1. --- INTRODUCTION --- p.1 / Chapter 1.1. --- Background --- p.1 / Chapter 1.2. --- Objectives --- p.1 / Chapter 1.3. --- Overview of Thesis --- p.2 / Chapter 2. --- NETWORK ASPECT OF THE VOIP TECHNOLOGY --- p.3 / Chapter 2.1. --- VoIP Overview --- p.3 / Chapter 2.2. --- Elements in VoIP --- p.3 / Chapter 2.2.1. --- Call Setup --- p.3 / Chapter 2.2.2. --- Media Capture/Playback --- p.4 / Chapter 2.2.3. --- Media Encoding/Decoding --- p.4 / Chapter 2.2.4. --- Media Transportation --- p.5 / Chapter 2.3. --- Performance Factors Affecting VoIP --- p.6 / Chapter 2.3.1. --- Network Bandwidth --- p.6 / Chapter 2.3.2. --- Latency --- p.6 / Chapter 2.3.3. --- Packet Loss --- p.7 / Chapter 2.3.4. --- Voice Quality --- p.7 / Chapter 2.3.5. --- Quality of Service (QoS) --- p.7 / Chapter 2.4. --- Different Requirements of Intranet VoIP and Internet VoIP --- p.8 / Chapter 2.4.1. --- Packet Loss/Delay/Jitter --- p.8 / Chapter 2.4.2. --- Interoperability --- p.9 / Chapter 2.4.3. --- Available Bandwidth --- p.9 / Chapter 2.4.4. --- Security Requirement --- p.10 / Chapter 2.5. --- Some Feasibility Investigations --- p.10 / Chapter 2.5.1. --- Bandwidth Calculation --- p.10 / Chapter 2.5.2. --- Simulation --- p.12 / Chapter 2.5.3. --- Conclusion --- p.17 / Chapter 2.5.4. --- Simulation Restrictions --- p.17 / Chapter 3. --- SOFTWARE ASPECT OF THE VOIP TECHNOLOGY --- p.19 / Chapter 3.1. --- VoIP Client in JMF --- p.19 / Chapter 3.1.1. --- Architecture --- p.20 / Chapter 3.1.2. --- Incoming Voice Stream Handling --- p.23 / Chapter 3.1.3. --- Outgoing Voice Stream Handling --- p.23 / Chapter 3.1.4. --- Relation between Incoming/Outgoing Voice Stream Handling --- p.23 / Chapter 3.1.5. --- Areas for Further Improvement --- p.25 / Chapter 3.2. --- Capture/Playback Enhanced VoIP Client --- p.26 / Chapter 3.2.1. --- Architecture --- p.27 / Chapter 3.2.2. --- Native Voice Playback Mechanism --- p.29 / Chapter 3.2.3. --- Native Voice Capturing Mechanism --- p.31 / Chapter 3.3. --- Win32 C++ VoIP Client --- p.31 / Chapter 3.3.1. --- Objectives --- p.32 / Chapter 3.3.2. --- Architecture --- p.33 / Chapter 3.3.3. --- Problems and Solutions in Implementation --- p.37 / Chapter 3.4. --- Win32 DirectSound C++ VoIP Client --- p.38 / Chapter 3.4.1. --- Architecture --- p.39 / Chapter 3.4.2. --- DirectSound Voice Playback Mechanism --- p.40 / Chapter 3.4.3. --- DirectSound Voice Capturing Mechanism --- p.44 / Chapter 3.5. --- Testing VoIP Clients --- p.45 / Chapter 3.5.1. --- Setup of Experiment --- p.45 / Chapter 3.5.2. --- Experiment Results --- p.47 / Chapter 3.5.3. --- Experiment Conclusion --- p.48 / Chapter 3.6. --- Real-time Voice Stream Mixing Server --- p.48 / Chapter 3.6.1. --- Structure Overview --- p.48 / Chapter 3.6.2. --- Experiment --- p.53 / Chapter 3.6.3. --- Conclusion --- p.54 / Chapter 4. --- EXPERIMENTAL STUDIES --- p.55 / Chapter 4.1. --- Pure IP-side VoIP-based Call Center ´ؤ VoIP in Education --- p.55 / Chapter 4.1.1. --- Architecture --- p.55 / Chapter 4.1.2. --- Client Structure --- p.56 / Chapter 4.1.3. --- Client Applet User Interface --- p.58 / Chapter 4.1.4. --- Observations --- p.63 / Chapter 4.2. --- A Simple PBX Experiment --- p.63 / Chapter 4.2.1. --- Structural Overview --- p.63 / Chapter 4.2.2. --- PSTN Gateway Server Program --- p.64 / Chapter 4.2.3. --- Problems and Solutions in Implementation --- p.66 / Chapter 4.2.4. --- Experiment 1 --- p.66 / Chapter 4.2.5. --- Experiment 2 --- p.68 / Chapter 5. --- A COMPREHENSIVE VOIP PROJECT 一 GRADUATE SECOND PHONE (GSP) --- p.72 / Chapter 5.1. --- Overview --- p.72 / Chapter 5.1.1. --- Background --- p.72 / Chapter 5.1.2. --- Architecture --- p.76 / Chapter 5.1.3. --- Technologies Used --- p.78 / Chapter 5.1.4. --- Major Functions --- p.80 / Chapter 5.2. --- Client --- p.84 / Chapter 5.2.1. --- Structure Overview --- p.85 / Chapter 5.2.2. --- Connection Procedure --- p.89 / Chapter 5.2.3. --- User Interface --- p.91 / Chapter 5.2.4. --- Observations --- p.92 / Chapter 5.3. --- Gateway --- p.94 / Chapter 5.3.1. --- Structure Overview --- p.94 / Chapter 5.3.2. --- Connection Procedure --- p.97 / Chapter 5.3.3. --- Caller ID Simulator --- p.97 / Chapter 5.3.4. --- Observations --- p.98 / Chapter 5.4. --- Server --- p.101 / Chapter 5.4.1. --- Structure Overview --- p.101 / Chapter 5.5. --- Details of Major Functions --- p.103 / Chapter 5.5.1. --- Secure Local Voice Message Box --- p.104 / Chapter 5.5.2. --- Call Distribution --- p.106 / Chapter 5.5.3. --- Call Forward --- p.112 / Chapter 5.5.4. --- Call Transfer --- p.115 / Chapter 5.6. --- Experiments --- p.116 / Chapter 5.6.1. --- Secure Local Voice Message Box --- p.117 / Chapter 5.6.2. --- Call Distribution --- p.118 / Chapter 5.6.3. --- Call Forward --- p.121 / Chapter 5.6.4. --- Call Transfer --- p.122 / Chapter 5.6.5. --- Dial Out --- p.124 / Chapter 5.7. --- Observations --- p.125 / Chapter 5.8. --- Outlook --- p.126 / Chapter 5.9. --- Alternatives --- p.127 / Chapter 5.9.1. --- Netmeeting --- p.127 / Chapter 5.9.2. --- OpenH323 --- p.128 / Chapter 6. --- CONCLUSIONS --- p.129 / Bibliography --- p.133

Cui, Yuanyuan. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2009. / Includes bibliographical references (p. 64-68). / Abstract also in Chinese. / Chapter Chapter 1 --- Introduction --- p.1 / Chapter 1 .1 --- Motivations and Contributions --- p.1 / Chapter 1.2 --- Related Works --- p.3 / Chapter 1.3 --- Organization of the Thesis --- p.4 / Chapter Chapter 2 --- Background --- p.5 / Chapter 2.1 --- IEEE 802.11 --- p.5 / Chapter 2.1.1 --- IEEE 802.11 Topologies --- p.5 / Chapter 2.1.2 --- IEEE 802.11 MAC --- p.8 / Chapter 2.2 --- Voice over Internet Protocol (VoIP) --- p.11 / Chapter 2.2.1 --- A VoIP system --- p.11 / Chapter 2.2.2 --- QoS requirements for VoIP --- p.11 / Chapter 2.2.3 --- VoIP speech codecs --- p.12 / Chapter 2.3 --- VoIP over WLAN --- p.13 / Chapter 2.3.1 --- System Architecture of VoIP over WLAN --- p.14 / Chapter 2.3.2 --- VoIP Capacity over WLAN --- p.15 / Chapter 2.4 --- Skype --- p.16 / Chapter Chapter 3 --- Skype Rate Adaptation Mechanism --- p.17 / Chapter 3.1 --- Experimental Setting --- p.17 / Chapter 3.2 --- Overview --- p.19 / Chapter 3.3 --- Flow Rate Region --- p.20 / Chapter 3.4 --- Feedback: Receiver Report (RR) --- p.21 / Chapter 3.5 --- Bandwidth Usage Target (BM) --- p.24 / Chapter 3.6 --- Summary of Skype Rate Adaptation Mechanism --- p.28 / Chapter 3.7 --- Skype-emulating Traffic Generator --- p.28 / Chapter Chapter 4 --- "Call Admission, Fairness and Stability Control" --- p.32 / Chapter 4.1 --- Unfair and Instability problems for AVoIP --- p.32 / Chapter 4.1.1 --- Analysis --- p.32 / Chapter 4.1.2 --- Simulation Evaluation --- p.34 / Chapter 4.2 --- CFSC scheme --- p.37 / Chapter 4.2.1 --- Pre-admission Bandwidth-reallocation Call Admission Control (PBCAC) --- p.39 / Chapter 4.2.2 --- Fairness Control --- p.42 / Chapter 4.2.3 --- Stability Control --- p.43 / Chapter Chapter 5 --- Performance Evaluation of CFSC --- p.44 / Chapter 5.1 --- Evaluation of Fairness Control --- p.44 / Chapter 5.2 --- Evaluation of Stability Control --- p.46 / Chapter 5.3 --- Evaluation of PBCAC --- p.46 / Chapter 5.4 --- Evaluation of complete CFSC --- p.49 / Chapter Chapter 6 --- Conclusion --- p.51 / Appendices --- p.53 / References --- p.64

Yuan, Yan. / Thesis submitted in: November 2007. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2008. / Includes bibliographical references (leaves 84-87). / Abstracts in English and Chinese. / Chapter 1 --- Introduction --- p.1 / Chapter 1.1 --- Wireless Mesh Network --- p.3 / Chapter 1.1.1 --- Architecture Overview --- p.3 / Chapter 1.1.2 --- Routing Protocols --- p.5 / Chapter 1.2 --- Contribution of this Thesis --- p.7 / Chapter 1.3 --- Organization of this Thesis --- p.8 / Chapter 2 --- Background and Literature Review --- p.9 / Chapter 2.1 --- VoIP on Wireless Mesh Networks --- p.9 / Chapter 2.1.1 --- Performance of VoIP on Wireless Mesh Networks --- p.9 / Chapter 2.1.2 --- Optimizations for VoIP over Wireless Mesh Networks --- p.12 / Chapter 2.1.3 --- Path and Packet Aggregation Scheme --- p.14 / Chapter 2.2 --- Network Coding on Wireless Mesh Networks --- p.15 / Chapter 2.2.1 --- The Concept of Network Coding --- p.15 / Chapter 2.2.2 --- Related Work --- p.16 / Chapter 3 --- Adaptive Path and Packet Aggregation System --- p.19 / Chapter 3.1 --- Overview --- p.19 / Chapter 3.2 --- The Adaptive Path Aggregation Routing Algorithm --- p.20 / Chapter 3.2.1 --- Protocol Overview --- p.20 / Chapter 3.2.2 --- Data Structure --- p.21 / Chapter 3.2.3 --- The Concept of Link Weight and Path Weight --- p.26 / Chapter 3.2.4 --- APA Operations --- p.27 / Chapter 3.3 --- The Packet Aggregation System --- p.39 / Chapter 3.3.1 --- Overview --- p.39 / Chapter 3.3.2 --- Packet structure --- p.40 / Chapter 3.3.3 --- Local Compression --- p.41 / Chapter 3.3.4 --- Packet Aggregation/Disaggregation --- p.42 / Chapter 3.4 --- Performance Analysis --- p.44 / Chapter 3.4.1 --- Integration of the path aggregation routing protocol and the packet aggregation system --- p.46 / Chapter 3.5 --- Performance Evaluation --- p.48 / Chapter 3.5.1 --- Testbed Setup --- p.48 / Chapter 3.5.2 --- Packet aggregation --- p.48 / Chapter 3.5.3 --- Combined scenario: path and packet aggregation --- p.58 / Chapter 3.6 --- Summary --- p.65 / Chapter 4 --- Network Coding System in wireless network --- p.67 / Chapter 4.1 --- Overview --- p.67 / Chapter 4.2 --- System Architecture --- p.68 / Chapter 4.2.1 --- Packet Format --- p.68 / Chapter 4.2.2 --- Encoding and decoding --- p.69 / Chapter 4.3 --- Performance Evaluation --- p.71 / Chapter 4.3.1 --- Experiment Setup --- p.71 / Chapter 4.3.2 --- Performance Metric --- p.72 / Chapter 4.3.3 --- Experiment Results --- p.72 / Chapter 4.4 --- Summary --- p.79 / Chapter 5 --- Conclusions and Future Directions --- p.82

Interactive multimedia sessions combine requirements of traditional telephony services and Internet applications. This requires call setup, call signaling, negotiation, routing, security, and network resources. Seeking to facilitate the use of quality of service (QoS) mechanisms to users of such applications, this thesis presented new signaling architectures that addressed the interaction of the Session Initiation Protocol (SIP) as the session control signaling protocol and current resource management frameworks. The Differentiated Services (DiffServ) architecture is used as the primary example. The new architectures addressed the roles of SIP agents and proxy servers in subjects such as resource negotiation, call authorization, and end-to-end QoS in heterogeneous networks. First, an architecture based on the use of QoS-enhanced SIP proxies and a SIP-based interface between the application and network layers was developed, implemented in a testbed, and performance enhancements demonstrated. Further studying of the Internet Engineering Task Force (IETF) proposal for the integration of SIP and resource management led to the development of a new signaling scheme, Resource management Overlapped with Answering Delay (ROAD). It explores the SIP user agent interaction with the network in a way that takes advantage of parallel user answering delays and reservation delays. An experimental evaluation of the ROAD scheme showed its call setup delay savings and reduced signaling load. Then, on the interaction of SIP and call admission control, an inter-domain call authorization model that implements the concepts of proxies as gate controllers (QoS-enhanced SIP proxies-GC), and that provides call authorization status and adds more granularity to the authorization process is proposed. This model showed to be scalable in terms of the need to add more resources to compensate for the increasing service load on the servers. Finally, an example framework that applies the new signaling architectures to achieve end-to-end QoS in heterogeneous networks is presented.

Thesis (M.S.)--University of Florida, 2002. / Title from title page of source document. Includes vita. Includes bibliographical references.

No description available.