Made available in DSpace on 2014-06-12T17:37:19Z (GMT). No. of bitstreams: 2 arquivo5321_1.pdf: 5149979 bytes, checksum: 5024e80a3c898e8ef33f4fea465f7318 (MD5) license.txt: 1748 bytes, checksum: 8a4605be74aa9ea9d79846c1fba20a33 (MD5) Previous issue date: 2008 / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / As redes de telefonia IP popularizaram-se nos últimos anos sobretudo por seu baixo custo e facilidade de utilização. Transmitir voz na forma de pacotes IP favorece o desenvolvimento de uma rede integrada, na qual diversos tipos de dados e mídia trafegam segundo um padrão único, que uniformize os sistemas de telecomunicações (Convergência IP). As redes sobrepostas par-a-par são parcial ou totalmente independentes de qualquer servidor centralizado, possuem alta escalabilidade e fornecem meios para que a comunicação atravesse obstáculos impostos por NATs e firewalls. Tais redes oferecem aos pacotes uma maior flexibilidade de roteamento, permitindo que novas estratégias sejam utilizadas no encaminhamento dos pacotes. Essas estratégias proporcionam uma melhor qualidade de voz ao usuário, principalmente durante falhas e congestionamentos. Nesta dissertação são estudados os sistemas de comunicação de voz sobre IP (VoIP) arquitetados em topologias par-a-par sobrepostas. Aspectos de codificação, sinalização, roteamento e tráfego, bem como os protocolos envolvidos em tais sistemas são descritos. Alternativas para obter uma melhor qualidade de voz através do uso dessa configuração são analisadas. Como contribuições dessa dissertação, é realizada uma análise comparativa entre sistemas VoIP e apresentada uma nova forma quantitativa de medição da QoS (Qualidade de Serviço) baseada na correlação entre os sinais transmitidos e recebidos

Diplomová práce ?Proměny konkurence na českém telekomunikačním trhu? je zaměřena především na analýzu nedávného turbulentního vývoje telekomunikací v České republice. Se zaměřením na společnost Telefónica O2 Czech Republic,a.s. práce popisuje hlavní faktory, které telekomunikační trh utvářely v uplynulých letech ? zabývá se významnými tržními aliancemi i otázkou regulace odvětví. Závěr práce je věnován hlavním trendům v telekomunikacích, kterými jsou především fixní a mobilní konvergence, VoIP, sítě třetí generace (3G), mobilní virtuální operátoři (MVNO) a televize přes IP (IPTV).

Komunikace je v dnešní době základním klíčem k úspěchu. Rychlost, dostupnost a kvalita předávaných informací je schopna ovlivnit nejen budoucnost firem, ale také jedinců. Cílem práce je poskytnout ucelený přehled o dostupných IT a SW prostředcích pro firemní komunikaci. E-mail, hlasová komunikace, videokonference, portály, ale také virtuální prostředí jsou postaveny proti klasickým formám komunikace jako je meeting a korespondence. V práci je vytvořena metodika umožňující srovnání jednotlivých forem komunikace z pohledu jejich vlastností, vhodností jejich použití pro různé modelové typy komunikace a možnosti jejich nasazení v interní a externí firemní komunikaci. Poslední kapitola práce demonstruje možnost aplikace zjištěných závěrů a získaných poznatků na středně velkou stavební firmu.

Secured long-distance communication has always been an important topic for people handling sensitive information. Now with the arrival of ``intelligent`` mobile phones eavesdropping and information gathering is as easy as never. Luckily smartphones present not only problems in terms of security but also an opportunity to protect ones privacy. This thesis attempts to construct a generic software architecture of a communicator which could be capable of transferring voice, video and other various forms of binary data in a secure way. It will analyse and use different communication channels to reach a maximum level of data authenticity, integrity and confidentiality in an environment where a central security element needs to be avoided. The resulting architecture will be tested on a Voice-over-IP (VoIP) application prototype for the mobile Google Android platform to show whether the approach is practically usable on currently available phones.

Tese de mestrado. Redes e Serviços de Comunicação. Faculdade de Engenharia. Universidade do Porto. 2006

Dissertação apresentada para obtenção do grau de Mestre em Redes e Serviços de Comunicação, na Faculdade de Engenharia da Universidade do Porto, sob a orientação do Professor Doutor Raúl Filipe Teixeira Oliveira

<p>The traditional voice service in mobile networks is an important service that mobile users expect high quality from. With the convergence of mobile networks towards an all-IP network, an IP-based speech service becomes important which is referred to as Voice-over-IP (VoIP). The traditional voice service is highly optimized and a VoIP service must therefore fulfil strict quality requirements to provide the same speech service quality. The air interface technology, WCDMA, which is used in third generation communication systems in Europe is constantly developed. An improved concept for the mobile-to-network transmission, called the Enhanced Uplink (EUL) provides for higher uplink capacity for packet data services. It also includes features that may provide a sufficient VoIP service quality in mobile networks, when considering the uplink transmission. </p><p>The purpose of this thesis is to evaluate the VoIP capacity over EUL and identify crucial aspects of radio resource management in order to increase the capacity. This is done through dynamic system simulations, using a realistic VoIP traffic model. The VoIP capacity is also estimated by a derived theoretical framework.\newline </p><p>It is shown by simulation results and theoretical estimations, that power control is a vital mechanism in order to increase the capacity. Simulation results indicate that a VoIP over EUL capacity of 65\% of the traditional voice service capacity may be reached. The results also indicate that to improve the capacity for larger cells, the allowed VoIP packet delay must be increased.</p>

<p>Voice over IP technology (VoIP) has recently gained popularity among consumers. Many popular VoIP services exist only as software for PCs. The need of taking such services out of the PC, into a stand-alone device has been discovered, and this thesis work deals with the development of such a device. The thesis work is done for Häger Scandinavia AB, a Swedish telephone manufacturer. This thesis work covers the design of a complete prototype of a table-top VoIP telephone running an embedded Linux Operating system. Design areas include product development, hardware design and software design.The result is a working prototype with hardware and corresponding Linux device drivers. The prototype can host a Linux application adapted to it. Conclusions are that the first hardware version has worked well and that using an open-source operating system is very useful. Further work consists of implementing a complete telephony software application in the system, evaluation of system requirements and adapting the prototype for a commercial design.</p>

<p>Voice over Internet Protocol (VoIP) has become very popular in recent days andbecome the first choice of small to medium companies for voice and data integration inorder to cut down the cost and use the IT resources in much more efficient way. Anotherpopular technology that is ruling the world after the year 2000 is 802.11 wirelessnetworks. The Organization wants to implement the VoIP on the wireless network. Thewireless medium has different nature and requirement than the 802.3 (Ethernet) andspecial consideration take into account while implementing the VoIP over wirelessnetwork.One of the major differences between 802.11 and 802.3 is the bandwidthavailability. When we implement the VoIP over 802.11, we must use the availablebandwidth is an efficient way that the VoIP application use as less bandwidth as possiblewhile retaining the good voice quality. In our project, we evaluated the differentcompression and decompression (CODEC) schemes over the wireless network for VoIP.To conduct this test we used two computers for comparing and evaluatingperformance between different CODEC. One dedicated system is used as Asterisk server,which is open source PBX software that is ready to use for main stream VoIPimplementation. Our main focus was on the end-to-end delay, jitter and packet loss forVoIP transmission for different CODECs under the different circumstances in thewireless network. The study also analyzed the VoIP codec selection based on the MeanOpinion Score (MOS) delivered by the softphone. In the end, we made a comparisonbetween all the proposed CODECs based on all the results and suggested the one Codecthat performs well in wireless network.</p>

Voice over IP (VoIP) is a real time application that allows transmitting voice through the Internet network. Recently there has been amazing progress in this field, mainly due to the development of voice codecs that react appropriately under conditions of packet loss, and the improvement of intelligent jitter buffers that perform better under conditions of variable inter packet delay. In addition, there are other factors that indirectly benefited VoIP. Today, computer networks are faster due to the advances in hardware and breakthrough algorithms. As a result, the quality of VoIP calls has improved considerably. However, the quality of VoIP calls under extreme conditions of packet loss still remains a major problem that needs to be addressed for the next generation of VoIP services. This thesis concentrates in making an analysis of the effects that network impairments, such as: delay, jitter, and packet loss have in the quality of VoIP calls and approaches to solve this problem. Finally, we analyze the impact of introducing forward error correction (FEC) Piggybacking and Reed Solomon codes for VoIP. To measure the mean opinion score of VoIP calls we develop an application based on the E-Model, and utilize perceptual evaluation of speech quality (PESQ).