Real-time video communication over wireless networks is a challenging problem because wireless channels suffer from fading, additive noise and interference, which translate into packet loss and delay. Since modern video encoders deliver video packets with decoding dependencies, packet loss and delay can significantly degrade the video quality at the receiver. Many error resilience mechanisms have been proposed to combat packet loss in wireless networks, but only a few were specifically designed for packet switched video telephony over Third Generation (3G) networks. The first part of the thesis presents an error resilience technique for packet switched video telephony that combines application layer Forward Error Correction (FEC) with rateless codes, Reference Picture Selection (RPS) and cross layer optimization. Rateless codes have lower encoding and decoding computational complexity compared to traditional error correcting codes. One can use them on complexity constrained hand-held devices. Also, their redundancy does not need to be fixed in advance and any number of encoded symbols can be generated on the fly. Reference picture selection is used to limit the effect of spatio-temporal error propagation. Limiting the effect of spatio-temporal error propagation results in better video quality. Cross layer optimization is used to minimize the data loss at the application layer when data is lost at the data link layer. Experimental results on a High Speed Packet Access (HSPA) network simulator for H.264 compressed standard video sequences show that the proposed technique achieves significant Peak Signal to Noise Ratio (PSNR) and Percentage Degraded Video Duration (PDVD) improvements over a state of the art error resilience technique known as Interactive Error Control (IEC), which is a combination of Error Tracking and feedback based Reference Picture Selection. The improvement is obtained at a cost of higher end-to-end delay. The proposed technique is improved by making the FEC (Rateless code) redundancy channel adaptive. Automatic Repeat Request (ARQ) is used to adjust the redundancy of the Rateless codes according to the channel conditions. Experimental results show that the channel adaptive scheme achieves significant PSNR and PDVD improvements over the static scheme for a simulated Long Term Evolution (LTE) network. In the third part of the thesis, the performance of the previous two schemes is improved by making the transmitter predict when rateless decoding will fail. In this case, reference picture selection is invoked early and transmission of encoded symbols for that source block is aborted. Simulations for an LTE network show that this results in video quality improvement and bandwidth savings. In the last part of the thesis, the performance of the adaptive technique is improved by exploiting the history of the wireless channel. In a Rayleigh fading wireless channel, the RLC-PDU losses are correlated under certain conditions. This correlation is exploited to adjust the redundancy of the Rateless code and results in higher Rateless code decoding success rate and higher video quality. Simulations for an LTE network show that the improvement was significant when the packet loss rate in the two wireless links was 10%. To facilitate the implementation of the proposed error resilience techniques in practical scenarios, RTP/UDP/IP level packetization schemes are also proposed for each error resilience technique. Compared to existing work, the proposed error resilience techniques provide better video quality. Also, more emphasis is given to implementation issues in 3G networks.

<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>

The aim of this diploma thesis is to get accustomed with the SIP protocol and with the problematics of attacks targeting this protocol, with the emphasis on DoS and DDoS attacks. The thesis focuses on detailed classification of the attacks, possibilities and forms of generation of the attacks and methodics of defense against them. The attacks of the flood type are especially stressed because they are easily generated and the SIP components are very prone to these attacks. Prototype implementations of the most important ideas concerning attack generation and protection against these attacks are also part of this thesis. Practical tests of the implementations performed in a simulated SIP environment are also included. 1

[pt] Este trabalho apresenta um estudo visando a estabelecer uma metodologia para dimensionamento de um sistema VoIP, focalizando, em particular, o dimensionamento de um multiplexador estatístico. Procuramos aplicar modelos e resultados existentes para o problema geral do multiplexador estatístico ao caso específico de um sistema VoIP, levando em conta as características do tráfego, os requisitos de QoS e os princípios das arquiteturas Intserv e Diffserv. Para isto, apresentamos um resumo da tecnologia VoIP, incluindo seus requisitos de qualidade e os protocolos apropriados para transportar este tipo de mídia na Internet. Discorremos sobre os mecanismos de controle de tráfego usuais em redes de pacotes com QoS, assim como sobre as Arquiteturas de QoS definidas pelo IETF . É apresentada uma revisão de modelos de tráfego e modelos aplicáveis à análise de multiplexadores estatísticos, com destaque para o chamado modelo fluido aplicado à descrição do tráfego gerado por um agregado homogêneo de fontes de voz, além de um estudo comparativo entre respostas obtidas analiticamente com aquelas obtidas por meio de simulação. A influência do tipo de codificador e de parâmetros como tamanho de pacote é investigada, mostrando-se a dificuldade em se dispor de um modelo analítico capaz de levar em conta, de forma precisa, os diferentes formatos do sistema VoIP. Por fim, estabelece-se um cenário para aplicação dos modelos a um sistema VoIP. / [en] This work presents a study aiming at to establish a methodology for sizing a VoIP system, focusing, in particular, the sizing of a statistical multiplexer. We apply existing models and results for the general problem of the statistical multiplexer to the specific case of a VoIP system, taking in account the characteristics of the traffic, the requirements of QoS and the principles of the architectures Intserv and Diffserv. For this, we present a summary of the VoIP technology, including its requirements of quality and the protocols appropriate to carry this type of media in the Internet. We discourse on the usual mechanisms of traffic control in packet networks with QoS, as well as on the Architectures of QoS defined by the IETF. A revision of traffic models and applicable models to the analysis of statistical multiplexers, with prominence for the fluid model applied to the description of the traffic generated for a homogeneous aggregate of voice sources, is presented. Besisdes, a comparative study of behavior gotten analytically with those gotten by means of simulation is made. The influence of the coder and parameters as so packet size is investigated, revealing the difficulty in finding an analytical model capable to take in account, with precision, the different formats of the VoIP system. Finally, we establish a scenario for application of the models to a VoIP system.

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

<p>Communication tools such as text-based instant messaging, voice and video relay services, real-time video chat and mobile SMS and MMS have successfully been used among Deaf people. Several years of field research with a local Deaf community revealed that disadvantaged South African Deaf&nbsp / people preferred to communicate with both Deaf and hearing peers in South African Sign Language as opposed to text. Synchronous video chat and video&nbsp / relay services provided such opportunities. Both types of services are commonly available in developed regions, but not in developing countries like South&nbsp / Africa. This thesis reports on a workaround approach to design and develop an asynchronous video communication tool that adapted synchronous video&nbsp / &nbsp / codecs to store-and-forward video delivery. This novel asynchronous video tool provided high quality South African Sign Language video chat at the&nbsp / expense of some additional latency. Synchronous video codec adaptation consisted of comparing codecs, and choosing one to optimise in order to&nbsp / minimise latency and preserve video quality. Traditional quality of service metrics only addressed real-time video quality and related services. There was no&nbsp / uch standard for asynchronous video communication. Therefore, we also enhanced traditional objective video quality metrics with subjective&nbsp / assessment metrics conducted with the local Deaf community.</p>

Feature interactions occur when one feature interferes with the intended operation of another feature. To detect such interactions, each new feature must be tested against existing features. The detected interactions must then be resolved; many existing approaches to resolving interactions require the feature set be prioritized. Unfortunately, the cost to determine a priority ordering for a feature set increases dramatically as the number of features increases. This thesis explores strategies to decrease the cost of prioritizing features, and thus facilitates priority-based solutions to resolving feature interactions. Specifically, this thesis introduces a categorization approach that reduces the complexity of determining priorities for a large set of features by decomposing the prioritization problem. Our categorization approach reduces this cost by using abstraction to divide the system's features into categories based on their main goal or functionality (e.g., block unwanted calls, present call information). Next, in order to detect and resolve the interactions that occur between these seemingly unrelated categories, we identify a set of principles for proper system behaviour that define acceptable behaviour in the global system. For example, a call that should be blocked by a call-screening feature should never result in a voice connection. The categories are then ordered, such that adherence to the principles is optimized. We show that using category priorities, to order a large feature set, correctly resolves interactions between individual features and significantly reduces the cost to determine priority orderings. The four significant contributions that this thesis makes are: 1) the categorization of features, 2) the principles of proper system behaviour, 3) automatic generation of priority orderings for categories, and 4) devising several optimizations that reduce the search space when exploring call simulations during the automatic generation of the priority orderings. These contributions are examined with respect to the telephony domain and result in the identification of 12 feature categories and 9 principles of proper system behaviour. A Prolog model was also created to run call simulations on the categories, using the identified principles as correctness criteria. Our case studies showed the reduced cost of our categorization approach is approximately 1/10^(55) % of the cost of a traditional approach. Given this significant reduction in the cost and the ability of our model to accurately reproduce the manually identified priority orderings, we can confidently argue that our categorization approach was successful. The three main limitations of our categorization approach are: 1) not all features (e.g., 911 features in telephony) can be categorized or some categories will contain a small number of features, 2) the generated priority ordering may still need to be analyzed by a human expert, and 3) the run time for our automatic generation of priority orderings remains factorial with respect to the size of the number of categories. However, these limitations are small in comparison to the savings generated by the categorization approach.