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Dissertations / Theses on the topic 'Voice over Internet Protocol (VoIP) telephony'
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Tasyumruk, Lutfullah. "Analysis of voice quality problems of Voice Over Internet Protocol (VoIP)." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03sep%5FTasyumruk.pdf.
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Lewis, Rosemary. "Operational benefit of implementing VoIP in a tactical environment." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03Jun%5FLewis.pdf.
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Thesis (M.S. in Information Systems and Operations)--Naval Postgraduate School, June 2003.Thesis advisor(s): Dan C. Boger, Rex Buddenberg. Includes bibliographical references (p. 61-62). Also available online.
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Patton, Mark B. "A case study of Internet Protocol Telephony implementation at United States Coast Guard headquarters." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2005. http://library.nps.navy.mil/uhtbin/hyperion/05Mar%5FPatton.pdf.
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Thesis (M.S. in Information Technology Management)--Naval Postgraduate School, March 2005.Thesis Advisor(s): Dan C. Boger, R. Scott Coté. Includes bibliographical references (p. 134-138). Also available online.
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Svešnikova, Anastasija. "Internetinės telefonijos teisinis reglamentavimas Lietuvoje." Master's thesis, Lithuanian Academic Libraries Network (LABT), 2009. http://vddb.library.lt/obj/LT-eLABa-0001:E.02~2008~D_20090204_113025-13014.
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Šio darbo tema - Internetinės telefonijos teisinis reglamentavimas Lietuvoje. Šiuolaikinis Internetinės telefonijos populiarumas ne tik sukelia vartotojų, bet ir reguliuotojų suinteresuotumą. Būtent ji pastaruoju metu kelia daugybę diskusijų tarptautiniuose bei nacionalinėse forumuose, kurių vienas pagrindinių aspektų – tinkamo Internetinės telefonijos reguliavimo sukūrimas. Pagrindinis baigiamojo magistrinio darbo tikslas – išnagrinėti Internetinės telefonijos reguliavimą tarptautiniu ir Lietuvos mastu, bei apžvelgti su juo susijusias problemas. Darbe nagrinėjama užsienio šalių praktika, remiantis kuria iškeliamos pagrindinės Lietuvos IP telefonijos teisinio reguliavimo gairės. Būtent: telefono numerių skyrimas, numerio perkeliamumas, skambučiai į pagalbos tarnybas, skambučiai kitais telefono numeriais, IP telefonijos skambučių saugumas, bei aprašomos su jų įgyvendinimu susijusios problemos.The topic of the paper is Legal Reglamentation of VoIP Telephony in Lithuania. VoIP telephony’s nowadays spread and popularity scores an interest and debates not only between it’s consumers but also between legal regulators. VoIP is the main and rather often discussed topic of international and national forums, which aim to develop its proper regulation. The main aim of this paper is to internationally analyse VoIP’s legal regulation and to survey its associated problems. Foreign countries’ experience helps to formulate basic guidelines of Lithuanian VoIP legal regulation. Namely: numbering, numbers portability, calls to emergency services, calls to other telephone numbers, safety of VoIP calls and its associated problems.
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Lella, Tuneesh Kumar. "Privacy of encrypted Voice Over Internet Protocol." Thesis, Texas A&M University, 2008. http://hdl.handle.net/1969.1/86009.
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In this research, we present a investigative study on how timing-based traffic analysis attacks can be used for recovery of the speech from a Voice Over Internet Protocol (VOIP) conversation by taking advantage of the reduction or suppression of the generation of traffic whenever the sender detects a voice inactivity period. We use the simple Bayesian classifier and the complex HMM (Hidden Markov Models) classier to evaluate the performance of our attack. Then we describe the usage of acoustic features in our attack to improve the performance. We conclude by presenting a number of problems that need in-depth study in order to be effective in carrying out silence detection based attacks on VOIP systems.
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Dechjaroen, Chaiporn. "Performance evaluation of Voice over Internet Protocol." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2002. http://library.nps.navy.mil/uhtbin/hyperion-image/02Dec%5FDechjaroen.pdf.
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Wallace, David T. Vegter Henry M. "Exploitation of existing Voice over Internet Protocol Technology for Department of the Navy application /." Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2002. http://library.nps.navy.mil/uhtbin/hyperion-image/02sep%5FWallace.pdf.
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Thesis (M.S. in Information Technology Management)--Naval Postgraduate School, September 2002.Thesis advisor(s):Dan Boger, Rex Buddenberg. Includes bibliographical references (p. 101-102). Also available online.
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Vaishnav, Chintan. "Voice over Internet Protocol (VoIP) : the dynamics of technology and regulation." Thesis, Massachusetts Institute of Technology, 2006. http://hdl.handle.net/1721.1/34533.
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Thesis (S.M.)--Massachusetts Institute of Technology, Engineering Systems Division, Technology and Policy Program, 2006.Includes bibliographical references (p. 144-145).
"What Voice over Internet Protocol (VoIP) is going to do is start to weaken the foundation of the way we've done things for 100 years...Congress already should be discussing the next telecom bill," said Federal Communications Commission (FCC) Chairman Michael Powell in February 2004, before the United States Senate. The objective of this thesis is to study how VoIP challenges the incumbent US telecommunications act. The appearance of VoIP comes at a juncture when telecommunications system has already turned into a large-scale, complex system with multiple, competing infrastructures. VoIP, however, greatly augments the nested complexity by affording a technology that enables multiple architectures and business models for delivering the same voice (and often converged voice and data) service, while remaining agnostic to the underlying infrastructure. The VoIP-enabled architectures have very different capabilities and costs from one another. Many do not - or cannot - support social regulations such as emergency 911, wiretapping and disability access. Most exploit the economic arbitrage opportunities by evading access charges and universal service contributions.
(cont.) Added to this is the combination of reduced asset specificity due to VoIP's layered architecture and a global standard based ubiquitous IP technology that frees the service providers of the need to own the delivery infrastructure, and enables them to offer service from anywhere globally. Such a misalignment - between regulatory obligations and technical capabilities - has the potential to incubate large-scale systemic failures due to lack of coordination between the local optimization focused private markets and the highly compartmentalized public institutions. The case of Communications Assistance for the Law Enforcement Act (CALEA) - also known as the wiretapping act - is taken to study its implications on VoIP. A system dynamics model is used for the analysis. Four policy lessons emerge through the process of arriving at the model and the subsequent sensitivity analysis. First, considering peer-to-peer (P2P) VoIP a non-issue for CALEA is exactly what might make it an issue. Second, if P2P VoIP aspires to be a telephony substitute, it will invite the threat of social regulation. Third, arms race between CALEA-compliant and non-compliant technologies may raise the cost of CALEA compliance. Fourth, prohibiting use of certain encryption techniques may help the LEA to keep their ability to wiretap intact, but it also deprives customers of the privacy the prohibited schemes would have offered, and thereby helps the Internet-crime.
by Chintan Vaishnav.
S.M.
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Nugent, Geoffrey J. "Voice Over Internet Protocol (VOIP), Video Games, and the Adolescent's Perceived Experience." ScholarWorks, 2014. https://scholarworks.waldenu.edu/dissertations/155.
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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.
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Chita, Christian. "MAIDS for VoIP : a Mobile Agents-based Intrusion Detection System for Voice over Internet Protocol." Thesis, University of British Columbia, 2008. http://hdl.handle.net/2429/5599.
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Compared to traditional (PSTN) voice networks, a Voice over Internet Protocol
network is a convergence of a signaling network and a data network using Internet
Protocol (IP). The use of shared media by VoIP systems opens the door to some
uncertainty as to the source of a call. While in the traditional voice networks one has to
tap into a specific circuit to eavesdrop, in an IP network any equipment connected to the
target LAN can identify, store and playback the VoIP packets that traverse that LAN.
Unlike traditional voice networks which have only “dumb” end nodes (i.e. simple
telephone receivers), VoIP must, by its very nature, deploy intelligent end point devices
such as computers andlor IP phones, which are connected to open public networks. An
unprotected, unauthenticated IP network makes VoIP susceptible to hostile use, such as
call hijacking, connection tear down, denial of service, or sending computer viruses over
the network.
In this thesis, we perform a series of attacks against a commercial VoIP
application, and prove that they succeed with nothing more than a couple of identity
tokens captured from the network traffic as prerequisites. We then leverage the mobile
agent-based framework introduced by APHIDS to design an Intrusion Detection System
implementing a gradual attack-response procedure, destined to inform and protect the
End-Users of the Application Under Test when specific, internet telephony attacks do
occur, and ultimately to block the capability of the attack perpetrator to induce further
damage.
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Kolan, Prakash. "System and Methods for Detecting Unwanted Voice Calls." Thesis, University of North Texas, 2007. https://digital.library.unt.edu/ark:/67531/metadc5155/.
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Voice over IP (VoIP) is a key enabling technology for the migration of circuit-switched PSTN architectures to packet-based IP networks. However, this migration is successful only if the present problems in IP networks are addressed before deploying VoIP infrastructure on a large scale. One of the important issues that the present VoIP networks face is the problem of unwanted calls commonly referred to as SPIT (spam over Internet telephony). Mostly, these SPIT calls are from unknown callers who broadcast unwanted calls. There may be unwanted calls from legitimate and known people too. In this case, the unwantedness depends on social proximity of the communicating parties. For detecting these unwanted calls, I propose a framework that analyzes incoming calls for unwanted behavior. The framework includes a VoIP spam detector (VSD) that analyzes incoming VoIP calls for spam behavior using trust and reputation techniques. The framework also includes a nuisance detector (ND) that proactively infers the nuisance (or reluctance of the end user) to receive incoming calls. This inference is based on past mutual behavior between the calling and the called party (i.e., caller and callee), the callee's presence (mood or state of mind) and tolerance in receiving voice calls from the caller, and the social closeness between the caller and the callee. The VSD and ND learn the behavior of callers over time and estimate the possibility of the call to be unwanted based on predetermined thresholds configured by the callee (or the filter administrators). These threshold values have to be automatically updated for integrating dynamic behavioral changes of the communicating parties. For updating these threshold values, I propose an automatic calibration mechanism using receiver operating characteristics curves (ROC). The VSD and ND use this mechanism for dynamically updating thresholds for optimizing their accuracy of detection. In addition to unwanted calls to the callees in a VoIP network, there can be unwanted traffic coming into a VoIP network that attempts to compromise VoIP network devices. Intelligent hackers can create malicious VoIP traffic for disrupting network activities. Hence, there is a need to frequently monitor the risk levels of critical network infrastructure. Towards realizing this objective, I describe a network level risk management mechanism that prioritizes resources in a VoIP network. The prioritization scheme involves an adaptive re-computation model of risk levels using attack graphs and Bayesian inference techniques. All the above techniques collectively account for a domain-level VoIP security solution.
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Epiphaniou, Gregory. "Iterative block ciphers' effects on quality of experience for VoIP unicast transmissions under different coding schemes." Thesis, University of Bedfordshire, 2010. http://hdl.handle.net/10547/142250.
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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.
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Darwis, Darwis. "Implementation and Analysis of VoIP CPE Management System using TR-069." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91668.
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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.
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Silva, Veridiano António Fernandes de Carvalho e. "Soluções wireless/VoIP para redes comunitárias." Master's thesis, Universidade de Aveiro, 2010. http://hdl.handle.net/10773/3615.
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Mestrado em Engenharia de Computadores e TelemáticaNas últimas décadas, a evolução das novas Tecnologias de Informação e Comunicação (TIC), contribuiu em larga escala para o crescimento da Internet e da utilização massificada das tecnologias de banda larga. Com essa evolução, surgiram novas formas de comunicar recorrendo a tecnologias inovadoras, baseadas no protocolo IP (Internet Protocol). Contudo, surgiram assim os softphones, que são as primeiras aplicações da tecnologia VoIP, que vieram revolucionar a forma de comunicar, com custos substancialmente reduzidos, que causaram um enorme impacto nas pessoas e nas organizações. Com o presente trabalho, pretende-se elaborar um estudo minucioso das tecnologias VoIP, apresentando algumas soluções de implementação de um sistema de comunicações VoIP para uma rede comunitária de banda larga. Por último, será apresentada uma proposta de arquitectura, descrevendo os possíveis cenários de implementação de um fornecedor de comunicações VoIP numa Mesh network de rede comunitária.
In latest decades, the evolution of new Information and Communication Technologies (ICT) has contributed a large scale for the growth of the Internet and use mass of broadband technologies. With these developments, there were new ways to communicate using innovative technologies, based on the protocol IP (Internet Protocol). However, emerged as the softphone, which are the first applications of the technology VoIP, who came to revolutionize the way of communicating, with costs substantially reduced, which caused a huge impact on people and organizations. With this work, it is intended to prepare a detailed study of the technology VoIP, providing some solutions for implementing a communication system to a VoIP network of community broadband. Finally, will be a proposal for architecture, describing the possible scenarios for implementing a VoIP provider of communications network in a mesh network community.
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Eisazadeh, Ali Akbar, and Nora Espahbodi. "Fast Fault Recovery in Switched Networks for Carrying IP Telephony Traffic." Thesis, Halmstad University, School of Information Science, Computer and Electrical Engineering (IDE), 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:hh:diva-3859.
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One of the most parts of VOIP management is fault management and, in having a good fault management, finding good mechanisms to detect faults in the network have to be considered.
The main focus of this project is to implement different types of fast fault recovery protocols in networks, especially networks that carry IP telephony. Having a complete understanding of some common link failure detection and fault recovery protocols, such as spanning tree protocol (STP), rapid spanning tree protocol (RSTP) and per-VLAN spanning tree protocol (PVSTP), and also having a complete understanding of three other common techniques for fault detection and fault recovery, such as hot standby routing protocol (HSRP), virtual router redundancy protocol (VRRP) and gateway load balancing protocol (GLBP) will be regarded in the project. We are going to test some fault recovery protocols which can be used in IP telephony networks and choose the best. We intend to focus on this issue in LAN environment in theoretical descriptions and practical implementations.
The final outcome of the thesis is implementation in the Halmstad University’s lab environment to obtain the final result. For doing our thesis, we are going to use some technical tools as hardware tools (Cisco L3 and L2 switches, Routers, IP Phones) and tools which are used for network performance monitoring, like as CommVeiw.
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Radhakrishna, Deekonda, and Jannu Keerthipramukh. "OPNET simulation of voice over MPLS With Considering Traffic Engineering." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-3434.
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Multiprotocol Label Switching (MPLS) is an emerging technology which ensures the reliable delivery of the Internet services with high transmission speed and lower delays. The key feature of MPLS is its Traffic Engineering (TE), which is used for effectively managing the networks for efficient utilization of network resources. Due to lower network delay, efficient forwarding mechanism, scalability and predictable performance of the services provided by MPLS technology makes it more suitable for implementing real-time applications such as voice and video. In this thesis performance of Voice over Internet Protocol (VoIP) application is compared between MPLS network and conventional Internet Protocol (IP) network. OPNET modeler 14.5 is used to simulate the both networks and the comparison is made based on some performance metrics such as voice jitter, voice packet end-to-end delay, voice delay variation, voice packet sent and received. The simulation results are analyzed and it shows that MPLS based solution provides better performance in implementing the VoIP application. In this thesis, by using voice packet end-to-end delay performance metric an approach is made to estimate the minimum number of VoIP calls that can be maintained, in MPLS and conventional IP networks with acceptable quality. This approach can help the network operators or designers to determine the number of VoIP calls that can be maintained for a given network by imitating the real network on the OPNET simulator.0046737675303
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Wulff, Tobias. "Evaluation of and Mitigation against Malicious Traffic in SIP-based VoIP Applications in a Broadband Internet Environment." Thesis, University of Canterbury. Computer Science and Software Engineering, 2010. http://hdl.handle.net/10092/5120.
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Voice Over IP (VoIP) telephony is becoming widespread, and is often integrated into computer networks. Because of his, it is likely that malicious software will threaten VoIP systems the same way traditional computer systems have been attacked by viruses, worms, and other automated agents. While most users have become familiar with email spam and viruses in email attachments, spam and malicious traffic over telephony currently is a relatively unknown threat. VoIP networks are a challenge to secure against such malware as much of the network intelligence is focused on the edge devices and access environment.
A novel security architecture is being developed which improves the security of a large VoIP network with many inexperienced users, such as non-IT office workers or telecommunication service customers. The new architecture establishes interaction between the VoIP backend and the end users, thus providing information about ongoing and unknown attacks to all users. An evaluation of the effectiveness and performance of different implementations of this architecture is done using virtual machines and network simulation software to emulate vulnerable clients and servers through providing apparent attack vectors.
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Knoll, James A. "Convergence of the Naval information infrastructure /." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2004. http://library.nps.navy.mil/uhtbin/hyperion/04Jun%5FKnoll.pdf.
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Doria, Priscila Lôbo Gonçalves. "Avaliação de desempenho de variantes dos Protocolos DCCP e TCP em cenários representativos." Universidade Federal de Sergipe, 2012. https://ri.ufs.br/handle/riufs/3332.
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The Datagram Congestion Control Protocol (DCCP) is a prominent transport protocol that has attracted the attention of the scientific community for its rapid progress and good
results. The main novelty of DCCP is the performance priority design, as in UDP, however with congestion control capabilities, as in TCP. Literature about DCCP is still scarce and needs to be complemented to gather enouth scientific elements to support new research properly. In this context, this work joins the efforts of the scientific community to analise, mensure, compare and characterize DCCP in relevant scenarios that cover many real world situations. Three open questions were preliminarly identified in the literature: How DCCP behaves (i) when fighting for the same link bandwidth with other transport protocols; (ii)
with highly relevant ones (e.g., Compound TCP, CUBIC) and (iii) fighting for the same link bandwidth with Compound TCP and CUBIC, adopting multimedia applications (e.g., VoIP). In this work, computational simulations are used to compare the performance of two DCCP variants (DCCP CCID2 and DCCP CCID3) with three highly representative TCP variants
(Compound TCP, CUBIC and TCP SACK), in real world scenarios, including concurrent use of the same link by protocols, link errors and assorted bandwidths, latencies and traffic
patterns. The simulation results show that, under contention, in most scenarios DCCP CCID2 has achieved higher throughput than Compound TCP or TCP SACK. Throughout
the simulations there was a tendency of DCCP CCID3 to have lower throughput than the other chosen protocol. However, the results also showed that DCCP CCID3 has achieved
significanly better throughput in the presence of link errors and higher values of latency and bandwidth, eventualy outperforming Compound TCP and TCP SACK. Finally, there was a
tendency of predominance of CUBIC´ throughtput, which can be explained by its aggressive algorithm (i.e., non-linear) of return of the transmission window to the previous value before
the discard event. However, CUBIC has presented the highest packet drop and the lowest delivery rate.O Datagram Congestion Control Protocol (DCCP) é um proeminente protocolo de transporte que vem atraindo a atenção da comunidade científica pelos seus rápidos avanços e bons resultados. A principal inovação do DCCP é a priorização de desempenho, como ocorre com o UDP, mas com capacidade de realizar controle de congestionamento, como ocorre com o TCP. Entretanto, a literatura sobre o DCCP ainda é escassa e necessita ser complementada para trazer elementos científicos suficientes para novas pesquisas. Neste contexto, este trabalho vem se somar aos esforços da comunidade científica para analisar, mensurar, comparar e caracterizar o DCCP em cenários representativos que incorporem diversas situações de uso. Identificaram-se então três questões alvo, ainda em aberto na literatura: qual é o comportamento do DCCP (i) quando disputa o mesmo enlace com outros protocolos de transporte; (ii) com protocolos de transporte relevantes (e.g., Compound TCP, CUBIC) e (iii) em disputa no mesmo enlace com o Compound TCP e o CUBIC, utilizando aplicações multimídia (e.g., VoIP). Neste trabalho, simulações computacionais são utilizadas para comparar duas variantes do DCCP (CCID2 e CCID3) a três variantes do TCP (Compound TCP, CUBIC e TCP SACK), em cenários onde ocorrem situações de mundo real, incluindo utilização concorrente do enlace pelos protocolos, presença de erros de transmissão no enlace, variação de largura de banda, variação de latência, e variação de padrão e distribuição de tráfego. Os resultados das simulações apontam que, sob contenção, na maioria dos cenários o DCCP CCID2 obteve vazão superior à do Compound TCP, do DCCP CCID3 e do TCP SACK. Ao longo das simulações observou-se uma tendência do DCCP CCID3 a ter vazão inferior à dos demais protocolos escolhidos. Entretanto, os resultados apontaram que o DCCP CCID3 obteve desempenho significativamente melhor na presença de erros de transmissão e com valores maiores de latência e de largura de banda, chegando a ultrapassar a vazão do DCCP CCID2 e do TCP SACK. Por fim, observou-se uma tendência de predominância do protocolo CUBIC no tocante à vazão, que pode ser determinada pelo seu algoritmo agressivo (i.e., não-linear) de retorno da janela de transmissão ao valor anterior aos eventos de descarte. Entretanto, o CUBIC apresentou o maior descarte de pacotes e a menor taxa de entrega.
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Janošík, Martin. "Návrh virtuální lokální počítačové sítě pro edukativní účely." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217529.
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The master’s thesis focuses on the virtual local computer network for laboratory usage. It aims to propose and realize proper network connection in order to monitor expected data flow. Thanks to the network analysers (software ClearSight and hardware NetTool Series II) it plans to pursue in detail the used transmission protocols of TCP/IP layers. The most decisive feature happens to be the right choice of appropriate network components and their precise configuration. Consequently, the thesis formulates a proposal of a laboratory task for the needs of students, which is also closely related to the actual problems. The assignment of the task will serve the teachers as a test pattern for measurement. The results elaborated in the form of the model protocol should enable later comparison of the recorded data. Another part of the diploma thesis is the working-out of well arranged manuals for the network analysers involved.
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Mfupe, Luzango. "A transparent settlement model and network architecture for mobile voice over Internet protocol (VOIP) service provider." Thesis, 2011. http://encore.tut.ac.za/iii/cpro/DigitalItemViewPage.external?sp=1000631.
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M. Tech. Electrical Engineering.A virtual Mobile Voice over IP (MVoIP) service can be implemented by a Mobile VoIP Operator (MVoIPO) in conjunction with a Mobile Network Operator (MNO). MVoIPOs do not operate their own mobile network infrastructure. Instead, they use the MNO's packet-based cellular network. However, the coexistence between the MVoIPO and the MNO raises two related problems: first, how to handle interconnection settlements, and second, how to (inter)connect the two operators to make such settlements. This dissertation uses a game-theoretic modelling approach to show that it is mutually beneficial economically if the MNO allows the MVoIPO to operate on its network. Further, a Service Level Agreement (SLA)-based Transparent Settlement Agreement (TSA) model is proposed to solve the first problem. The TSA model algorithm calculates the MVoIPO's throughput distribution at the edge of a UMTS Core Network (CN). This facilitates the determination of levels of conformance to the pre-set throughput thresholds and, subsequently, the issuing of compensation to the MVoIPO by the MNO after generating an economically acceptable volume of traffic. Further, possible network architecture to solve the second problem is suggested, by combining the TSA model algorithm, the UMTS CN, the IP Multimedia Subsystem (IMS), and the Online Charging System (OCS)
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"Voice-over-IP (VoIP) over wireless local area networks (WLAN)." 2004. http://library.cuhk.edu.hk/record=b5892251.
Full textAbstract:
Wang Wei.Thesis (M.Phil.)--Chinese University of Hong Kong, 2004.
Includes bibliographical references (leaves 80-83).
Abstracts in English and Chinese.
Chapter Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Motivations and Contributions --- p.1
Chapter 1.2 --- Organization of the Thesis --- p.4
Chapter Chapter 2 --- Background --- p.6
Chapter 2.1 --- IEEE 802.11 --- p.6
Chapter 2.1.1 --- Distributed Coordination Function (DCF) / Point Coordination Function (PCF) --- p.7
Chapter 2.1.2 --- Types of Networks --- p.8
Chapter 2.1.3 --- The 802.11 MAC Sublayer Protocol --- p.9
Chapter 2.1.4 --- Why CSMA/CA for Wireless LAN? --- p.11
Chapter 2.2 --- Voice over IP (VoIP) --- p.13
Chapter 2.2.1 --- Speech Codec --- p.13
Chapter 2.2.2 --- The H.323 Standard --- p.13
Chapter 2.3 --- Related Work --- p.15
Chapter 2.3.1 --- Capacity limits of VoIP over WLAN --- p.16
Chapter 2.3.2 --- Methods for increasing VoIP capacity over WLAN --- p.16
Chapter 2.3.3 --- Interference between traffic of VoIP and other applications --- p.18
Chapter Chapter 3 --- VoIP Multiplex-Multicast Scheme --- p.20
Chapter 3.1 --- System Architecture --- p.20
Chapter 3.2 --- Packet Multiplexing and Multicasting --- p.22
Chapter 3.3 --- Header Compression --- p.24
Chapter 3.4 --- Connection Establishment --- p.29
Chapter Chapter 4 --- Capacity Analysis --- p.31
Chapter 4.1 --- VoIP Capacity Analysis for 802. 11b --- p.31
Chapter 4.1.1 --- Capacity of Ordinary VoIP over WLAN --- p.32
Chapter 4.1.2 --- Capacity of Multiplex-Multicast Scheme over WLAN --- p.33
Chapter 4.2 --- "VoIP Capacity Analysis for 802,11a and 802.11g" --- p.34
Chapter 4.3 --- VoIP Capacity with VBR Sources --- p.38
Chapter 4.4 --- Simulations --- p.38
Chapter Chapter 5 --- Delay Performance --- p.41
Chapter 5.1 --- Access Delay --- p.42
Chapter 5.2 --- Extra Delay Incurred by the Multiplex-Multicast Scheme --- p.47
Chapter Chapter 6 --- VoIP Co-existing with TCP Interference Traffic --- p.49
Chapter 6.1 --- Ordinary VoIP co-existing with TCP over WLAN --- p.49
Chapter 6.1.1 --- Problem Caused by TCP Interference --- p.49
Chapter 6.1.2 --- Solutions --- p.52
Chapter 6.2 --- M-M VoIP coexisting with TCP over WLAN --- p.53
Chapter 6.3 --- 802.11e --- p.56
Chapter 6.3.1 --- EDCA --- p.56
Chapter 6.3.2 --- ACK Policies --- p.58
Chapter 6.3.3 --- VoIP over EDCA --- p.58
Chapter Chapter 7 --- Experimental Validation --- p.61
Chapter 7.1 --- Transmission Errors --- p.61
Chapter 7.2 --- Prototype Implementation --- p.62
Chapter Chapter 8 --- VoIP over Ad Hoc Networks --- p.65
Chapter 8.1 --- Mobile Ad Hoc Networks (MANET) and Wireless Distributed System (WDS) --- p.65
Chapter 8.2 --- The M-M Scheme in WDS --- p.67
Chapter 8.2.1 --- Modified System Architecture --- p.67
Chapter 8.2.2 --- Delay Performance --- p.68
Chapter 8.2.3 --- Analysis of M-M Scheme in WDS --- p.69
Chapter 8.2.4 --- Capacity Improvement --- p.70
Chapter 8.2.5 --- Delay Improvement --- p.71
Chapter 8.2.6 --- Spectrum Reuse --- p.71
Chapter Chapter 9 --- Conclusions --- p.76
References --- p.80
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Darmani, Mohammad Yousef. "Lost VOIP packet recovery in active networks." 2004. http://hdl.handle.net/2440/59644.
Full textAbstract:
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
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Loubser, Jacob Bester. "Design of a practical voice over internet protocol network for the multi user enterprise." Thesis, 2005. http://hdl.handle.net/10352/123.
Full textAbstract:
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.
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"Optimization of resources allocation for H.323 endpoints and terminals over VoIP networks." Thesis, 2014. http://hdl.handle.net/10210/8860.
Full textAbstract:
M.Phil. (Electrical & Electronic Engineering)Without any doubt, the entire range of voice and TV signals will migrate to the packet network. The universal addressable mode of Internet protocol (IP) and the interfacing framing structure of Ethernet are the main reasons behind the success of TCP/IP and Ethernet as a packet network and network access scheme mechanisms. Unfortunately, the success of the Internet has been the problem for real-time traffic such as voice, leading to more studies in the domain of Teletraffic Engineering; and the lack of a resource reservation mechanism in Ethernet, which constitutes a huge problem as switching system mechanism, have raised enough challenges for such a migration. In that context, ITU-T has released a series of Recommendation under the umbrella of H.323 to guarantee the required Quality of Service (QoS) for such services. Although the “utilisation” is not a good parameter in terms of traffic and QoS, we are here in proposing a multiplexing scheme with a queuing solution that takes into account the positive correlations of the packet arrival process experienced at the multiplexer input with the aim to optimize the utilisation of the buffer and bandwidth on the one hand; and the ITU-T H.323 Endpoints and Terminals configuration that can sustain such a multiplexing scheme on the other hand. We take into account the solution of the models from the M/M/1 up to G/G/1 queues based on Kolmogorov’s analysis as our solution to provide a better justification of our approach. This solution, the Diffusion approximation, is the limit of the Fluid process that has not been used enough as queuing solution in the domain of networking. Driven by the results of the Fluid method, and the resulting Gaussian distribution from the Diffusion approximation, the application of the asymptotic properties of the Maximum Likelihood Estimation (MLE) as the central limit theorem allowed capturing the fluctuations and therefore filtering out the positive correlations in the queue system. This has resulted in a queue system able to serve 1 erlang (100% of transmission link capacity) of traffic intensity without any extra delay and a queue length which is 60% of buffer utilization when compared to the ordinary Poisson queue length.
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"An asynchronous time division multiplexing scheme for voice over IP." 2000. http://library.cuhk.edu.hk/record=b5895809.
Full textAbstract:
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
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Alvarez, Charles Conceicao, and 安查爾. "New Business Model on IP (Web) based Network - Voice over Internet Protocol (VoIP) Technology (Telecommunications Industry)." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/24714424707511802749.
Full text
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JEON, SUNG-HO, and 全成浩. "Benefits and Risks of Using Voice over Internet Protocol (VoIP) in Taiwan Business: An Exploratory Study." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/36881265238365350767.
Full textAbstract:
碩士銘傳大學
企業管理學系碩士班
99
ABSTRACT This research thesis was presented as groundwork, an analytical data to promote convenience of internet phone users as well as revitalization of internet phone service by understanding the service quality and device preference that internet users prefer. A total of 620 survey studies were handed out to accompany the research with direct approach method for higher percentage of collection. The major purpose of this study is twofold: Firstly, to understand the current state of VoIP adoption and use within Taiwanese business. Secondly, we will discuss our analysis on both the different benefits expected for subscribers of various characteristics as well as their risk and satisfaction levels. In Taiwan the top 3 VoIP service providers by number of subscribers are (in order): “PChome Internet''s Skype", "Microsoft MSN" and "Yahoo! Messenger". In general, the most popular use of VoIP services includes sending text messages as well as voice chat and video chat with other users. For VoIP phone call services, over 24.8% was for international calls. In the enterprise, 40% of enterprise subscribers respond to having used VoIP over the last 3 years. Most respondents agreed that with the added convenience of VoIP service comes certain risks, with the top concerns being reliability (internet connections are often subject to sudden disconnection) and sometimes lower voice quality (highly dependent on environment). However, subscribers generally agree that the benefits of using VoIP outweigh the common risks, the foremost benefit being cheaper costs. Still, even with significant cost savings, many respondents expressed dissatisfaction with the current charge rates of service providers. When analyzing the subscriber base by certain characteristics, we have found that among Taiwanese business subscribers, benefit and risk recognition was noticeably more pronounced in females compared to males. In contrast, satisfaction level was generally more recognized among males than females. With regards to the different service providers, the satisfaction level of Microsoft MSN subscribers was generally lower than with other subscribers. Accordingly, the results from this research and their interpretation must be expropriated with certain considerations including research limitations and restrictions. It is suggested that additional in-depth research with improved conditions should be carried out in the future. What’s more, there’s a need to review VoIP receptivity that must be conducted systematically in a wider spectrum, while measures for the promotion and growth of VoIP service in Taiwanese business through development and sale of additional service must be presented.
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Mandal, Sandipan. "A new alternate routing scheme with endpoint admission control for low call loss probability in VoIP network." Thesis, 2006. http://hdl.handle.net/10057/548.
Full textAbstract:
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."
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Teixeira, João Nuno Melo. "Cobertura VHF suportada por uma rede IP para o uso de VoIP." Master's thesis, 2019. http://hdl.handle.net/10071/20250.
Full textAbstract:
Num mundo cada vez mais tecnológico, são inúmeros os benefícios e as facilidades que se criam com a inserção das tecnologias em diversas áreas. No âmbito da navegação aérea, a adoção das tecnologias atualmente existentes visa elevar o nível da eficiência e da segurança com que os Air Traffic Services (ATS’s) são prestados.
Com o crescimento do tráfego aéreo surgem cada vez mais desafios na prestação de ATS’s. Garantir a segurança das aeronaves e promover a eficiência do fluxo das mesmas no espaço aéreo nunca foi uma tarefa fácil. De forma a atender às novas demandas do mercado, a Comissão Europeia definiu o programa SESAR 2020, o qual tem como missão otimizar a prestação das Air Navigation Service Providers (ANSP’s) no espaço europeu, introduzindo tecnologias de última geração nos sistemas inerentes à prestação dos seus serviços.
No âmbito das comunicações entre os diversos sistemas, a aplicação das novas tecnologias remete para a utilização do Internet Protocol (IP), promovendo os benefícios inerentes à interoperabilidade entre sistemas. Uma vez que a navegação aérea se enquadra num ambiente extremamente regulado e crítico, foram definidas normas de forma a mitigar fragilidades e garantir que qualquer sistema IP implementado sirva a operação de forma segura e eficiente. Nesta dissertação, são explorados e simulados os upgrades a serem implementados nos sistemas de comunicações de voz que a NAV Portugal recorre para a prestação do Serviço Radar.
De forma a elevar o grau de qualidade operacional e tecnológica na prestação deste serviço, são consideradas as etapas fundamentais à transmissão de voz entre os Controladores de Tráfico Aéreo (CTA’s) e as aeronaves. Uma vez que as comunicações do Serviço Radar são efetuadas via Very High Frequency (VHF), numa primeira fase são apresentadas propostas técnicas relativamente à qualidade da cobertura VHF disponível. Posteriormente, são analisadas e discutidas as soluções que visam introduzir o Voice over Internet Protocol (VoIP) nos sistemas de voz inerentes a ao Serviço Radar. Por fim, é projetada uma rede de comunicações IP, a qual será responsável pela ligação entre estes sistemas tendo em consideração o novo ambiente regulatório.In an increasingly technological world, there are many benefits and facilities created by the insertion of modern technologies in many areas. In the air navigation world, the adoption of these technologies aims to raise the level of efficiency and security with which Air Traffic Services (ATS’s) are provided. With the growth of air traffic, more and more challenges arise in the provision of ATS. Ensuring safety and promoting airspace efficiency has never been an easy task. In order to meet the new market demands, the European Commission has defined the SESAR 2020 program, whose mission is to optimize Air Navigation Service Providers (ANSP’s) services delivery in European airspace by introducing state-of-art technologies into the systems inherent to their services. In the context of communications between different systems, the application of new technologies refers to the user of the Internet Protocol (IP), promoting the adjacent benefits of interoperability between systems. Since air navigation fits into an extremely regulated and critical environment, standards have been set to mitigate weaknesses and ensure that any deployed IP system serves the operation in a more efficient and secure way. This dissertation explores the upgrades to be implemented in the voice communication systems used by NAV Portugal to provide the Radar Service In order to increase the operational and technological quality in the provision of this service, the fundamental steps to voice transmission between air traffic controllers (ATC's) and aircrafts are considered in this project. Since Radar Service communications are established by Very High Frequency (VHF), firstly, technical proposals are made regarding the quality of available VHF coverage. Subsequently, solutions aimed at introducing Voice over Internet Protocol (VoIP) into the voice systems inherent to the Radar Service are analysed and discussed. Finally, it will be designed an IP communications network that will support the connections between these systems considering the new regulations.