Call admission control (CAC) extends the capabilities of Quality of service (QoS) tools which protect voice traffic from the negative effects of other voice traffic. It does not allow oversubscription of a Voice over Internet Protocol (VoIP) network. To achieve better performance for efficient call admission control, various dynamic routings are being proposed. In the dynamic routing mechanism, the condition of the network is learned by observing the network condition via the probe packets and according to the defined threshold, routes are chosen dynamically. In such schemes, various combination of route selection is used such as two routes are used where one is fixed and other is random or two random routes are chosen and after observation one is chosen if it passes the test. Few schemes use a route history table along with the two random routes. But all have some issues like it selects random routes (not considering the number of hops), does not process memorization before admission threshold test, it calculates all selected paths regardless of the fact that they are selected or not, thereby wasting central processing unit (CPU) time and since these uses two routes so obviously the call admission probability is less. In this thesis work, a new dynamic routing scheme is proposed which considers a routing history table with endpoint admission control increasing the call admission probability, makes call establishment time faster and it saves valuable CPU resources. The proposed scheme considers a combination of three routes with routing history table--one is the direct route and the other two are selected randomly from all available routes and the routing history table is used to memorize the rejected calls. CAC tests like Admission Threshold were performed on the selected routes. Various parameters such as delay, packet loss, jitter, latency etc from the probe packets are used to carry out the tests. Performance of the proposed scheme with respect to other dynamic routing schemes is studied using a mathematical / analytical model. Also, effect of arrival rate probe packets on utilization, busy period, waiting period, acceptance probability of calls, probe packets, and the number of successful calls was also studied. / Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical and Computer Engineering. / "July 2006."

Title page, table of contents and abstract only. The complete thesis in print form is available from the University of Adelaide Library. / Current best-effort packet-switched Internet is not a perfect environment for real-time applications such as transmitting voice-over the network (Voice Over Internet Protocol or VOIP). Due to the unlimited concurrent access to the Internet by users, the packet loss problem cannot be avoided. Therefore, the VOIP based applications encompass problems such as "voice quality degradation caused by lost packets". The effects of lost packets are fundamental issues in real-time voice transmission over the current unreliable Internet. The dropped packets have a negative impact on voice quality and concealing their effects at the receiver does not deal with all of the drop consequences. It has been observed that in a very lossy network, the receiver cannot cope with all the effects of lost packets and thereby the voice will have poor quality. At this point the Active Networks, a relatively new concept in networking, which allows users to execute a program on the packets in active nodes, can help VOIP regenerate the lost packets, and improve the quality of the received voice. Therefore, VOIP needs special voice-packing methods. Based on the measured packet loss rates, many new methods are introduced that can pack voice packets in such a way that the lost packets can be regenerated both within the network and at the receiver. The proposed voice-packing methods could help regenerate lost packets in the active nodes within the network to improve the perceptual quality of the received sound. The packing methods include schemes for packing samples from low and medium compressed sample-based codecs (PCM, ADPCM) and also include schemes for packing samples from high compressed frame-based codecs (G.729). Using these packing schemes, the received voice has good quality even under very high loss rates. Simulating a very lossy network using NS-2 and testing the regenerated voice quality by an audience showed that significant voice quality improvement is achievable by employing these packing schemes. / http://proxy.library.adelaide.edu.au/login?url= http://library.adelaide.edu.au/cgi-bin/Pwebrecon.cgi?BBID=1147315 / Thesis (Ph.D.) -- University of Adelaide, School of Electrical and Electronic Engineering, 2004

Approved for public release, distribution is unlimited / In this thesis, Voice over Internet Protocol (VoIP) technology will be explored and a recommendation of the operational benefit of VoIP will be provided. A network model will be used to demonstrate improvement of voice End-to-End delay by implementing quality of service (QoS) controls. An overview of VoIP requirements will be covered and recommended standards will be reviewed. A clear definition of a Battle Group will be presented and an overview of current analog RF voice technology will be explained. A comparison of RF voice technology and VoIP will modeled using OPNET Modeler 9.0. / Lieutenant, United States Navy

Thesis (M. Tech. Engineering: Electrical--Vaal University of Technology. / This dissertation discusses the design and implementation of a voice over internet protocol system for the multi-user enterprise. It is limited to small to medium enterprises of which the Vaal University of Technology is an example. Voice communications over existing Internet protocol networks are governed by standards, and to develop such a system it is necessary to have a thorough understanding of these standards. Two such standards namely the International Telecommunications Unions H.323 and the Internet Engineering Task Force's SIP were evaluated and compared to each other in terms of their complexity, extensibility and scalability as well as the services they offer. Based on these criteria it was decided to implement a SIP system. A SIP network consists of application software that act as clients and servers, as well as hardware components such as a proxy and redirect and registrar or location servers that allow users of this network to call each other on the data network. Gateways enable users of the network to call regular public switched telephone network numbers. A test network was set up in the laboratory that contained all the hardware and software components. This was done to understand the installation and configuration options of the different software components and to determine the suitability and interoperability of the software components. This network was then migrated to the network of the Vaal University of Technology which allowed selected users to test and use it. Bandwidth use is a major point of contention, and calculations and measurements showed that the codec being used during the voice call is the determining factor. This SIP system is being used on a daily basis and the users report excellent audio quality between soft phones and soft phones, soft phones and normal telephones and even cellular phones.

Video games are an everyday experience for adolescents and have changed how adolescents interact with one another. Prior research has focused on positive and negative aspects of video game play in general, without distinguishing Voice Over Internet Protocol (VOIPing) as the mode of play. Grounded in entertainment theory, motivational theory, and psychological distress theory, this cross-sectional, correlational study examined the relationship between VOIPing and quality of life (Pediatric Quality of Life Inventory), Yee's motivation to play video games, and resilience (Child and Youth Resilience Measure). A series of linear regression and multivariate canonical correlation models analyzed self-report responses of 103 adolescents aged 13 to18. Results indicated that VOIPing was not statistically related to quality of life or resilience. However, VOIPing correlated positively with motivation to play video games, particularly with the subscales of socialization and relationships. Canonical analysis of motivation for gaming and quality of life indicated that adolescents with high scores on customization and escapism motivation for gaming subscales tended to also have high scores on each of the emotional, social, and school quality of life subscales. Canonical analysis of motivation for gaming and resilience indicated that adolescents with low scores on the escapism motivation for gaming subscale tended to also have high scores on the individual, relationships, and community resilience subscales. The positive aspects of VOIPing, particularly with increased motivation to play video games, can be effectively used in coaching adolescents in social skills and relationship building.

Issues around Quality of Service (QoS) and security for Voice over IP (VoIP) have been extensively investigated separately, due to the great attention this technology currently attracts. The specific problem this work addresses centres upon the selection of optimal parameters for QoS and security for VoIP streams integrating both network impairments and user perception metrics into a novel empirically-driven approach. Specifically, the simulation model seeks the optimal parameters in terms of variable VoIP payloads, iterative block ciphers, codecs and authentication mechanisms to be used, so that optimum tradeoff between a set of conflicting factors is achieved. The model employs the widely used Transmission Rating Factor, R, as the methodology to predict and measure the perceived QoS based on current transmission and network impairments. The R factor is then used to map perceived QoS to the corresponding Mean Opinion Score value, which gives the average estimation of perceived voice quality (Quality of Experience). Furthermore, a genetic algorithm (GA) has been developed that uses the output from the simulation model as an input into an offline optimisation routine that simultaneously maximises the VoIP call volumes and the Level of Encryption (LoE) per call basis, without degrading the perceived quality of service under a specific threshold as dictated by the R factor. The solutions reflect the optimum combination of parameters for each codec used and due to the small size of the search space the actual speed of GA has been validated against an exhaustive search algorithm. The results extracted from this study demonstrate that under strict and pre-defined parameters the default payload size supported by the codecs is not the optimal selection in terms of call volume maximisation and perceived QoS when encryption is applied.

Over the past decade, the rapid developments and growth of information and communication technology (ICT) have triggered a new wave of customer service. This study utilizes the net-enable business innovation cycle theory with secondary data analysis to analyze the process and outcomes of the implementation of emerging technology, i.e., web-based application and voice over internet protocol (VoIP) for the case company, i.e., A Company. We investigate the characteristics and feature of the emerging technology including the web-based application and VoIP, identify the potential economic opportunity for the A Company, analyze the needed business innovation for its growth, and assess the potential value for its customer. The findings have the potential to contribute to the understanding of impacts occurring in the innovation associated with the implementation of the emerging technology for the A Company and offer rich insights for the company to exploit the economic opportunities, the needed business innovation, and the potential value for the customer. This approach also provides a systematic template that helps an organization to decide whether an emerging technology is worthy to implement.

Customer Premises Equipment (CPE) management is underestimated by the CPE vendors and services providers while it is in fact one of the most important aspects to ensure the high quality of service. Many people still think CPE management is the same as network management. Thus, they use the Simple Network Management Protocol (SNMP) to manage their CPEs. However, SNMP alone was thought not to scale nor to support the provisioning of the types of services which internet services providers must support today. This thesis highlights the importance of CPE management, how it is implemented using the TR-069; a CPE management protocol defined by the DSL Forum, and how a management system can be used for VoIP service management, and whether a CPE should implement TR-069 or SNMP as the management system to support. In the addition, the TR-069 will be compared against the SNMP to determine which one is more suitable for CPE management. An interesting conclusion is that while TR-069 does have some advantages over SNMP for managing services rather than simply managing the device, these advantages are not a large as initially believed nor has TR-069 avoided the problem of proprietary management information which SNMP has demonstrated. / Customer Premises Equipment (CPE) skötseln är undervärderad av CPE försäljarna och tjänste leverantörerna meddans det faktiskt är en av de mest viktiga aspekterna för att tillförsäkra hög quality of service. Många personer tror fortfarande att CPE skötseln är det samma som att sköta ett nätverk. Så, de använder Simple Network Management Protocol (SNMP) för att sköta deras CPE:er. Emellertid, SNMP ensamt var inte tänkt att skala eller att ge stöd vid försörjning av typer av tjänster som internet tjänst leverantörer måste stödja idag. Den här avhandlingen framhäver det väsentliga med CPE skötsel, hur det implementeras vid användande av TR-069; ett CPE skötsel protocol definerat av DSL forum, och hur detta administrations system kan användas för att sköta VoIP tjänster. Tilläggande så kommer avhandligen att jämföra TR-069 och SNMP för att bestämma vilken av dem som är mer lämplig för CPE administration. En intressant sammanfattning är att meddans TR-069 har några fördelar över SNMP för att sköta tjänster hellre än att enkelt sköta enheten, dessa fördelar är inte så stora som man trott från början. Dessutom, TR-069 ser inte ut att kunna övervinna problemet med privatägd (användande av privat MIB) information som SNMP har demonstrerat.

Approved for public release, distribution is unlimited / Recent advances in information technology communications have brought about increases in bandwidth and processing speeds to encourage the growth of Internet Protocol Telephony (IPT), a method of transmitting voice conversations over data networks. Many organizations are replacing portions of their traditional phone systems to gain the benefits of cost savings and enhanced feature sets through the use of IPT. The Coast Guard has an interest in exploiting this technology, and has taken its first steps by implementing IPT at Headquarters Support Command in Washington D.C. This thesis investigates the successful implementation practices and security policies of commercial, educational, and government organizations in order to create recommendations for IPT security policies and implementation practices relevant to the Coast Guard. It includes the discussion of the public switched telephone network, an overview of IPT, IPT security issues, the safeguards available to counter security threats, the tradeoffs (e.g., voice quality, cost) required to mitigate security risks, and current IPT security policy and implementation guidance. It is supported by the study and analysis of the IPT system at Coast Guard Headquarters. The Coast Guard gains an understanding of the advantages, limitations, and security issues that it will face as it considers further implementation of IPT. / Lieutenant, United States Coast Guard

Magister Scientiae - MSc / Telecommunication companies, service providers and large companies are now adapting converged multi-service Next Generation Networks (NGNs). Network management is shifting from managing Network Elements (NE) to managing services. This paradigm shift coincides with the rapid development of Quality of Service (QoS) protocols for IP networks. NEs and services are managed with Policy Based Network Management (PBNM) which is most concerned with managing services that require QoS using the Common Open Policy Service (COPS) Protocol. These services include Voice over IP (VoIP), video conferencing and video streaming. It follows that legacy NEs without support for QoS need to be replaced and/or excluded from the network. However, since most of these services run over IP, and legacy NEs easily supports IP, it may be unnecessary to throw away legacy NEs if it can be made to fit within a PBNM approach. Our approach enables an existing PBNM system to include legacy NEs in its management paradigm. The Proxy-Policy Enforcement Point (P-PEP) and Queuing Policy Enforcement Point (Q-PEP) can enforce some degree of traffic shaping on a gateway to the legacy portion of the network. The P-PEP utilises firewall techniques using the common legacy and contemporary NE management protocol Simple Network Management Protocol (SNMP) while the Q-PEP uses queuing techniques in the form Class Based Queuing (CBQ) and Random Early Discard (RED) for traffic control. / South Africa