Θα περιγράψουμε τις βασικές λειτουργίες ενός VoIP δικτύου, τα συστατικά του μέρη, καθώς και τα πρωτόκολλα που είναι υπεύθυνα για την εγκατάσταση, τον έλεγχο και τον τερματισμό μιας VoIP υπηρεσίας-συνομιλίας. Τα πρωτόκολλα αυτά ονομάζονται πρωτόκολλα σηματοδοσίας. Τα πρωτόκολλα σηματοδοσίας για VoIP εφαρμογές και ιδιαίτερα το πρωτόκολλο SIP (Session Initiation Protocol) είναι το βασικό θέμα της παρούσας εργασίας. Συγκεκριμένα, έγινε ανάπτυξη ενός SIP User Agent, το λογισμικό του οποίου θα εκτελείται στο δικτυακό επεξεργαστή IXP425 της Intel, μαζί με τα απαραίτητα πρωτόκολλα για την κωδικοποίηση-αποκωδικοποίηση και μετάδοση δειγμάτων φωνής σε μορφή πακέτων δεδομένων. Το αποτέλεσμα αναμένεται να είναι ένα ολοκληρωμένο προϊόν (VoIP phone) για την πραγματοποίηση VoIP κλήσεων. / -

The rapid growth of the Internet and the proliferation of its new applications pose a serious challenge in network performance management and monitoring. The current Internet has no mechanism for providing feedback on network congestion to the end-systems at the IP layer. For applications and their end hosts, end-to-end measurements may be the only way of measuring network performance. Understanding the packet delay and loss behavior of the Internet is important for proper design of network algorithms such as routing and flow control algorithms, for the dimensioning of buffers and link capacity, and for choosing parameters in simulation and analytic studies. In this thesis, round trip time (RTT), one-way network delay and packet loss in the Internet are measured at different times of the day, using a Voice over IP (VoIP) device. The effect of clock skew on one-way network delay measurements is eliminated by a Linear Programming algorithm, implemented in MATLAB. Distributions of one-way network delay and RTT in the Internet are determined. It is observed that delay distribution has a gamma-like shape with heavy tail. It is tried to model delay distribution with gamma, lognormal and Weibull distributions. It is observed that most of the packet losses in the Internet are single packet losses. The effect of firewall on delay measurements is also observed.

Σκοπός της παρούσας διπλωματικής εργασίας ήταν η μελέτη των παραμέτρων που επηρεάζουν την ποιότητα εξυπηρέτησης (QoS – Quality of Service) των εφαρμογών VoIP μέσω των διαφόρων τεχνολογιών ευρυζωνικής πρόσβασης. / Aim of the present diplomatic assigment was the study of parameters that affect the quality of service (QoS) of VoIP applications via the various broadband access technologies.

Submitted by Automa??o e Estat?stica (sst@bczm.ufrn.br) on 2015-12-03T23:16:48Z No. of bitstreams: 1 AnaFlaviaLinsSouto_DISSERT.pdf: 1191835 bytes, checksum: f7fea5d276f29ff8f41cd847fd02f335 (MD5) / Approved for entry into archive by Arlan Eloi Leite Silva (eloihistoriador@yahoo.com.br) on 2015-12-09T22:31:31Z (GMT) No. of bitstreams: 1 AnaFlaviaLinsSouto_DISSERT.pdf: 1191835 bytes, checksum: f7fea5d276f29ff8f41cd847fd02f335 (MD5) / Made available in DSpace on 2015-12-09T22:31:31Z (GMT). No. of bitstreams: 1 AnaFlaviaLinsSouto_DISSERT.pdf: 1191835 bytes, checksum: f7fea5d276f29ff8f41cd847fd02f335 (MD5) Previous issue date: 2014-11-10 / O presente trabalho tem como objetivo demonstrar a rela??o do princ?pio da efici?ncia -como expresso na Constitui??o Federal de 1988, atrav?s da Emenda Constitucional n? 19 - com as ag?ncias reguladoras, mais precisamente a ANATEL (Ag?ncia Nacional de Telecomunica??es). Abrange tamb?m a import?ncia que esse princ?pio possui sobre a regula??o - fiscalizar e gerenciar os servi?os p?blicos - e quando uma atividade ser? considerada eficiente, tendo em mente que as ag?ncias sujeitam-se aos demais princ?pios da Administra??o P?blica. O crescente uso da telefonia vem possibilitando um maior desenvolvimento de tecnologias que proporcionam melhorias na presta??o desse servi?o. O VoIP (Voice over IP), nada mais ? que um avan?o tecnol?gico que atinge diretamente as prestadoras do servi?o de telefonia convencional, tanto pela modifica??o dos neg?cios que trabalhavam por um longo per?odo com a mesma tecnologia, quanto pela quantidade de novos concorrentes que entram no mercado. Como ag?ncia reguladora do servi?o de telecomunica??es, a ANATEL, ainda n?o regulamentou o servi?o de telefonia de voz utilizando o protocolo IP. O que se aguarda com o passar dos anos ? que a ANATEL exer?a a sua fun??o regulat?ria para proporcionar melhores condi??es de competi??o entre as empresas prestadoras do VoIP e da telefonia convencional, obviamente que algumas dificuldades s?o esperadas, haja vista, que o VoIP ? uma tecnologia que abrange dois servi?os, tanto a telefonia convencional quanto a utiliza??o da internet / The present work aims to demonstrate the link of the principle of efficiency - as expressed in the Constitution of 1988, by Constitutional Amendment No. 19 - with regulatory agencies, more specifically the ANATEL (National Telecommunications Agency). It also includes this principle?s importance to regulation - to monitor and manage public services - as well as when an activity will be considered efficient, keeping in mind that agencies are subjected to other principles of public administration. The increasing use of telephony has enabled further development of technologies that provide improvements in the provision of this service. The VoIP (Voice over IP), is nothing more than a technological breakthrough that directly targets the providers of conventional telephone service, both by modifying the business working for a long time with the same technology as the amount of new competitors? dispute on market share. It also analyses the difficulty of understanding and definition of what is VoIP telephony, its growth and the threats that the traditional and mostly which is ANATEL?s role concerning this telephony technology. As regulator of the telecommunications service, ANATEL not yet regulated the voice telephony service using the IP protocol. What looks over the years is that ANATEL exercise its regulatory function to provide better conditions for competition among providers of VoIP and traditional telephone companies, obviously some difficulties are expected, given that VoIP is a technology that provides two services, through conventional telephony and using the internet.

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.

Submitted by Alexandre Cappellozza (cappellozza@gmail.com) on 2013-03-19T02:08:05Z No. of bitstreams: 1 TESE_CAPPELLOZZA_EAESP.pdf: 1587882 bytes, checksum: bf6cff1534cf7ab3eeeaf3110f44c6ff (MD5) / Approved for entry into archive by Suzinei Teles Garcia Garcia (suzinei.garcia@fgv.br) on 2013-03-19T14:27:35Z (GMT) No. of bitstreams: 1 TESE_CAPPELLOZZA_EAESP.pdf: 1587882 bytes, checksum: bf6cff1534cf7ab3eeeaf3110f44c6ff (MD5) / Made available in DSpace on 2013-03-19T14:29:05Z (GMT). No. of bitstreams: 1 TESE_CAPPELLOZZA_EAESP.pdf: 1587882 bytes, checksum: bf6cff1534cf7ab3eeeaf3110f44c6ff (MD5) Previous issue date: 2013-02-20 / Scientific studies have shown the existence of several factors that influence the adoption process of Information Technology. Different aspects related to the intended use technologies seem to be present in decisions about technology adoption, for example, relative advantages obtained by the use of technology, knowledge required for the operation of systems, ease of use and usefulness of technology, among others. Consequently, technology suppliers can also pursue business strategies that attract users through offers that include subsidies, discounts, such as phone carriers. However, there are scientific studies showing that the process of technology adoption may not be explained only by economics aspects but, also, include behavioral ones. Theories focused on adoption technology models can explain a portion of the reasons that lead individuals to behave in accordance with the use of a particular technology. For example, individual habit, accessibility and convenience may influence the preferences of a particular use of information technology. In this sense, users may manifest resistance on the use of available technologies, where this resistance can be justified from negative perceptions that would form a barrier to adoption of information systems. Therefore, one of the technology segments that present conditions with multiple features is the technology of voice communication, where the user can communicate via wired lines, mobile phones, Internet, among other ways. Thus, it is possible to analyze influences, including behavioral ones, where individuals can express decisions which show others reasons, besides the exclusive pursuit of economic results, on the use of personal communication technologies. In order to implement those analysis, we chose to contextualize the study focused on the technology of Voice over IP - VOIP as compared to other communication technologies, can be presented as one of the benefits major economic: free-calls between users of a system, coupled with other benefits emanating from the telephony technology itself. The results from this study have demonstrated VOIP individual adoption receives influences from several factors, positioned in different dimensions. For instance, individual perceptions about the characteristics of technology, user network and habits, besides commercial incentives toward concurrent communication technologies usage may create an individual subjective network of perceptions to the VOIP telephony adoption over the benefits that may be gathered from this application. / Estudos científicos têm demonstrado a existência de diversos fatores de influência sobre o processo de adoção de Tecnologia da Informação. Vários aspectos potencializadores das intenções de uso das tecnologias parecem estar presentes nas decisões sobre a adoção de tecnologia como, por exemplo, vantagens relativas obtidas pelo uso da tecnologia, conhecimento requerido para operação dos sistemas, facilidade e utilidade uso da tecnologia, entre outros. No entanto, há estudos que demonstram que o processo de adoção de tecnologia pode não ser explicado, somente, por aspectos financeiros e mercadológicos, mas englobar fatores endógenos que interferem nas decisões dos usuários sobre o uso de tecnologias de informação. Teorias e modelos de adoção de tecnologias conseguem explicar uma parcela dos motivos que levam os indivíduos a se comportarem de acordo com o uso de uma determinada tecnologia. Por exemplo: hábito individual, conectividade e conveniência podem influenciar as preferências de uso de uma determinada tecnologia de informação. De acordo com estas teorias, é possível analisar as influências que os indivíduos percebem e consideram nas decisões como justificativas sobre o uso de tecnologias de comunicação pessoal, além da busca exclusiva por resultados econômicos. Observa-se que um dos segmentos de tecnologia que apresenta condições de diferentes ofertas e múltiplas funcionalidades se refere ao segmento de tecnologias de comunicação de voz no qual o usuário pode se comunicar por meio de linhas telefônicas fixas, móveis, Internet, entre outras formas. Para a operacionalização de uma análise de adoção de tecnologia que englobe múltiplas interações de influências ao usuário, optou-se pela contextualização do estudo com foco na tecnologia de comunicação de voz pela Internet – VOIP, pois quando comparada com outras tecnologias de comunicação, adiciona-se que a gratuidade de ligações entre usuários de um mesmo sistema pode ser apresentada como um dos benefícios econômicos principais, aliada a outros benefícios provindos da telefonia em si. Os resultados obtidos por esta pesquisa confirmam a influência de diversos fatores posicionados em diferentes dimensões e proporcionam conclusões relevantes à adoção das tecnologias de comunicação de voz sobre Internet. Conclui-se que as percepções individuais sobre as características da tecnologia, a rede de contatos do usuário, hábito de uso e incentivos comerciais destinados ao uso de outras tecnologias de comunicação podem formar uma rede de influências à adoção da telefonia VOIP frente às percepções sobre os benefícios que podem ser obtidos com o uso desta aplicação.

Multiprotocol Label Switching (MPLS) is a new paradigm in routing architectures which has changed the way Internet Protocol (IP) packet is transferred in a Network. MPLS ensures the reliability of the communication minimizing the delays and enhancing the speed of packet transfer. One important feature of MPLS is its capability of providing Traffic Engineering (TE) which plays a vital role for minimizing the congestion by efficient load, balancing and management of the network resources. The performance evaluation is done considering the network parameters latency, jitter, packet end to end delay, and packet delay variation. Integration of QoS with the MPLS-TE network may enhance the performance of the network. Various scheduling algorithms can be used for implementing QoS on a network, which may vary the performance of the network. In our study, QoS is implemented on top of the MPLS-TE network using Differentiated Service (DiffServ) architecture. Different basic scheduling algorithms are used for the implementation of QoS and to check their impact on the network and to identify the suitable one among them. Performance evaluation is done considering the network parameters latency, jitter, packet end-to-end delay, and Packet Delay Variation. The simulation was done using OPNET modeler 16.0 and the results were analyzed. The simulation result shows that using TE along with QoS in MPLS network decreases the latency, jitter, packet delay variation and end to end packet delay compared to using TE alone for voice traffic. / +46738732963

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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

Enormous developing electronic technology has brought telecommunication to the sky inspiring popularity. Wireless Network (WN) technology will be emerging so far human. Presently users give the impression loving be connected all the time (everywhere) to the network or Internet through diverse access system, e.g., Universal Mobile Telecommunications System, Mobile Ad-hoc Network (MANET) [9] and Worldwide Interoperability for Microwave Access [30]. In telecommunication; MANET is considered as self-configured unlike nodes creating infrastructure-less network connected by means of WN, as nodes can exchange data packets without a central control [18]. Choosing beyond line of sight (BLOS) communication, MANET can be an intelligent selection, which is flexible for using and cost saving. However, to maintain quality of service (QoS), more or less challenges still have to be resolved [18]. Multimedia as well as VoIP (Voice over Internet Protocol) gaining more popularity as the internet world favored by the huge use of WN (access technologies) [20]. To maintain persistent services in different MANET situation unlike routing protocols (RP) are employed. In this thesis known (MANET) routing protocols OLSR (Optimized Link State Routing) [20], DSR (Dynamic Source Routing) [30] and TORA (Temporally Ordered Routing Algorithm) [14] have been considered for voice traffic as they maintaining dissimilar characteristics in the dissimilar situations as WN factors (like; bandwidth, signal strength, network traffic or load, network size) influencing the voice quality [9]. This thesis work focusing on the impact of node mobility influencing voice quality in unlike RPs in MANET. To decide the best suit RP in the MANET, the OPNET (Optimized Network Engineering Tool) Simulator 16.0 has been brought into play. OLSR is proposed to be best fitting RP for MANETs running VoIP appliance. / Enorma utveckla elektroniska tekniken har fört telekommunikation till himlen inspirerande popularitet. Trådlöst nätverk (WN) teknik kommer att utvecklas så långt mänsklig. För närvarande användare ger intryck kärleksfulla vara ansluten hela tiden (överallt) till nätverket eller Internet via olika system för tillträde, till exempel, Universal Mobile Telecommunications System, Mobil Ad-hoc-nätverk (Manet) [9] och Worldwide Interoperability för Microwave Access-[30 ]. I telekommunikation, är Manet betraktas som själv-konfigurerade skillnad noder skapa infrastruktur mindre nätverk som är anslutet via WN, som noder kan utbyta datapaket utan central styrning [18]. Välja bortom synfältet (Blos) kommunikation, kan Manet vara ett intelligent val, som är flexibel för att använda och kostnadsbesparande. Men för att upprätthålla service (QoS), mer eller mindre problem återstår att lösas [18]. Multimedia samt VoIP (Voice over Internet Protocol) allt mer populärt eftersom internet världen gynnas av den enorma användningen av WN (accesstekniker) [20]. För att behålla ihållande tjänster i olika Manet situationen skillnad routing protokoll (RP) är anställda. I denna kända avhandling (Manet) routingprotokoll OLSR (Optimerad Routing Link State) [20], DSR (Dynamic Source Routing) [30] och Tora (Tidsmässigt ordnad routing algoritm) har [14] ansetts för taltrafik som de upprätthålla olika egenskaper i olika situationer som WN faktorer (som, bandbredd, signalstyrka, nätverkstrafik eller last, nätverkets storlek) Att påverka ljudkvaliteten [9]. Detta examensarbete fokuserar på effekterna av nod rörlighet påverka röstkvalitet i motsats till RPS i Manet. För att avgöra det bäst passar RP i Manet, den OPNET (Optimerad Network Engineering Tool) Simulator 16,0 har kommit in i bilden. OLSR föreslås bli bäst passar RP för MANETs köra VoIP-apparaten. / Bergvägen 18 5tr 196 31 Kungsängen Sweden Mob: 0046(0)737106987