Magistrinio darbo tikslas yra ištirti SIP protokolo panaudojimo galimybes mobiliems prietaisams skirtuose internetinės telefonijos sprendimuose ir parengti tokio bendravimo modelį. Atlikto darbo pagrindinis rezultatas yra reikalingos literatūros apžvalga ir analizė bei eksperimentinio IP telefonijos modelio realizacija mobiliose operacinėse sistemose. Pirmame skyriuje trumpai pateikiamas įvadas apie tradicinės telefonijos evoliuciją į balso perdavimą per interneto tinklus. Taip pat, kad palaipsniui IP telefonija vis labiau realizuojama į mobilius prietaisus. Tolimesniame skyriuje, norint argumentuoti IP telefonijos augimą, aprašomas atliktas rinkos tyrimas ir jo išvados apie esamą padėtį IP telefonijos srityje. Trečiame skyriuje pateikiama informacija apie apžvelgtą literatūrą: SIP protokolo struktūrą ir savybes, įkraunamas aplikacijas į mobiliuosius terminalus bei jų palyginimas. Sekančiame skyriuje pateikiama atlikto eksperimentinio modelio realizacija: reikalavimų specifikacija, reikalavimai įrangai, pasirinktų priemonių veikimo demonstravimas mobiliose aplinkose. Penktame skyriuje parašyti pagrindiniai rezultatai ir gautos išvados, o paskutiniame – nurodomi darbe naudoti literatūros šaltiniai. / The goal of this Master Thesis is to research practice possibilities of SIP protocol in voice over IP solutions for mobile devices and to formulate such communication model. The main result of accomplished work is review and analysis of needful literature and implementation of experimental IP telephony model in mobile operating systems. In the first section is introduced the preface about traditional telephony development to a voice transfer over internet networks. Also, that IP telephony is more and more actualized to mobile devices. To prove the growth of IP telephony was done a market research and was made the conclusions about existing position in VoIP sphere. That information can be found in the next chapter. The third chapter is about reviewed literature: the structure and features of SIP protocol, applications which can be installed into mobile terminals and their comparison. In the next section of the Master Thesis is given realization of experimental model: specifications, request for hardware, demonstration of chosen software tools in mobile systems. The main results and obtained conclusions are written in the fifth chapter, literature list – in the last one.

Pour avoir une idée sur la qualité du réseau, la majorité des acteurs concernés (opérateurs réseau, fournisseurs de service) se basent sur la Qualité de Service (Quality of Service). Cette mesure a montré des limites et beaucoup d'efforts ont été déployés pour mettre en place une nouvelle métrique qui reflète, de façon plus précise, la qualité du service offert. Cette mesure s'appelle la qualité d'expérience (Quality of Experience). La qualité d'expérience reflète la satisfaction de l'utilisateur par rapport au service qu'il utilise. Aujourd'hui, évaluer la qualité d'expérience est devenu primordiale pour les fournisseurs de services et les fournisseurs de contenus. Cette nécessité nous a poussés à innover et concevoir des nouvelles méthodes pour estimer la QoE. Dans cette thèse, nous travaillons sur l'estimation de la QoE (1) dans le cas des communications Voix sur IP et (2) dans le cas des services de diffusion Vidéo sur IP. Nous étudions les performances et la qualité des codecs iLBC, Speex et Silk pour la VoIP et les codecs MPEG-2 et H.264/SVC pour la vidéo sur IP. Nous étudions l'impact que peut avoir la majorité des paramètres réseaux, des paramètres sources (au niveau du codage) et destinations (au niveau du décodage) sur la qualité finale. Afin de mettre en place des outils précis d'estimation de la QoE en temps réel, nous nous basons sur la méthodologie Pseudo-Subjective Quality Assessment. La méthodologie PSQA est basée sur un modèle mathématique appelé les réseaux de neurones artificiels. En plus des réseaux de neurones, nous utilisons la régression polynomiale pour l'estimation de la QoE dans le cas de la VoIP.

Internetas bei jo paslaugos – neatskiriama šiuolaikinės visuomenės dalis. Šiuo metu sunkiai įsivaizduojamas yra kompanijų darbas – be elektroninės bankininkystės, deklaracijų, prekybos sistemų ir pan. Viena iš tokių paslaugų yra sparčiai besivystanti internetinė telefonija (IP telefonija) arba VoIP (Voice Over IP – angl.). Esminis ir pagrindinis šios komunikavimo paslaugos pranašumas prieš tradicines priemones – kaina. Tarptautiniai IP telefonijos pokalbių tarifai jau senokai pigumu viršija tradicinės telefonijos įkainius , o neretai šie pokalbiai yra visiškai nemokami. Nepaisant ekonomiškumo, IP telefonijos srityje kyla taip pat nemažai problemų. Kadangi balso informacija yra perduodama paketiniu būdu (SIP, H.323 protokolai) per IP tinklus ir neturi atskiro kanalo duomenų perdavimui, neretai susiduriama su tam tikromis kokybės problemomis. Svarbiausios jų – uždelsimas(vėlinimas) bei paketų praradimas. Šio darbo esmė – IP telefonijos kanalu perduoto balso įrašo kokybės tyrimas, originalaus bei perduoto įrašo kokybiniai skirtumai, todėl darbo problema yra – balso, perduoto IP telefonijos kanalais kokybės nuostoliai. Tyrimo metu bus matuojami kokybės nuostoliai. Šio darbo objektas – IP kanalu perduoto balso kokybė. Darbo tikslas – ištirti bei įvertinti IP telefonijos kanalais perduoto balso kokybę ir nustatyti, ar perduotas balso įrašas yra tinkamas naudoti balso atpažinimo sistemose. Uždaviniai tikslui pasiekti: • Išsiaiškinti IP telefonijos veikimo principus, technologijas; •... [toliau žr. visą tekstą] / VoIP (voice over IP protocol) is quickly growing and expanding technology. Calls and other services are much cheaper than traditional ones. Nevertheless it meets some quality problems such as packet delay and packet loss. The object of this paper is to explore voice transmitted by IP channels quality and to conclude about if that voice could be used in voice recognition systems. To complete this object, there are 4 main tasks: • Find out the technologies and working techniques of the VoIP. • Determine tools for quality analysis of the transmitted voice • Build an algorithm which could help to extract useful data from voice signal. • Summarize the results and conclude about ability to use transmitted voice in voice recognition systems. There was made 150 transmissions („Skype“ system was used) of the voice during the experiment and all of them was recorded. Fast Fourier transform was used to transform those recording to frequency scale. The quality loss was analyzed; original voice record specter was compared with transmitted voice record specter. The transmissions were divided into two parts. Transmissions was made in 2008 February (1st part) and 2008 May (2nd part). In the first part the results are almost the same, quality variations are stable (specter difference average is 1,44), all values are between 1,44 and 1,55 (specter difference average value). It was noticed that transmitted voice signal has large quality variations in the second part of the experiment. The... [to full text]

Αντικείμενο της εργασίας μας, είναι η δημιουργία ενός γραφικού περιβάλλοντος για τη διαχείριση του συστήματος VOIP υπηρεσιών. Αυτό είναι ο Opensips Web Manager, μια ολοκληρωμένη λύση, από άποψη διαμόρφωσης, χρήσιμο για τον διαχειριστή ενός τηλεφωνικού συστήματος. Ο OWM είναι ένα web user interface, που έχει σχεδιαστεί με PHP. Χρησιμοποιείται για τη διαχείριση των πληροφοριών του συστήματος όσο και για τη διαχείριση της κατάστασης λειτουργίας των εξυπηρετητών, που στηρίζουν ένα VOIP σύστημα. O OWM. Ο διαχειριστής έχει πρόσβαση στον OWM μέσω ενός web browser και έτσι η φυσική θέση του διαχειριστή δεν περιορίζεται από την φυσική θέση του SIP εξυπηρετητή. Η αναζήτηση πληροφοριών, που αφορούν συνδρομητές, στοιχεία κλήσεων κ.α. γίνεται με τη δημιουργία «query», προς τη βάση δεδομένων του SIP εξυπηρετητή. Η μέθοδος ανάπτυξης της εφαρμογής μας ήταν δισδιάστατη. Αρχικά έγινε η υλοποίηση συστήματος VOIP υπηρεσιών, που περιελάμβανε τις διαμορφώσεις των εξυπηρετητών DHCPs , DNSs και OpenSIPs. Ακολούθησε η εγκατάσταση του WEBs εξυπηρετητή, που φιλοξενεί τον OWM. Τέλος σχεδιάστηκε το web user interface. / The subject of our project is to create a graphical user interface for managing systems that provide VOIP services. This is the Opensips Web Manager, a integrated solution regarding configuration, useful for the administration of a telephone system. OWM is a web user interface, designed using PHP. Especially OWM is used to manage system’s data and to control server’s status, which support a VOIP system. The administrator has access to the OWM through a web browser, so the physical location of the manager is not limited by SIP server’s physical location. The search for subscriber’ s and calling feature’s data, is performed by creating «query», to SIP server’s database. The method we use to develop our application has been two-dimensional. First was the implementation of VOIP service, which included the configurations of servers like DHCPs, DNSs and OpenSIPs. Next step was the installation of a WEB server that hosts OWM. Finally, we designed the web user interface.

Increasing capacity in wireless mesh networks can be achieved by using multiple channels and radios. By using different channels, two nodes can send packets at the same time without interfering with each other. To utilize diversity of available frequency, a channel assignment scheme is required. Hybrid channel assignment is an interesting approach where at least one radio is tuned to a fixed channel for receiving and the remaining interfaces switch their channels dynamically in order to match the receiving channel at the receiving node. This provides full connectivity, but at the expense of introduced switching costs. Due to hardware limitations it is too costly to switch channels on a per packet basis. Instead, this thesis proposes an anypath routing and forwarding mechanism in order to allow each node along the route to select the best next hop neighbor on a per packet basis. The routing algorithm finds for each destination a set of next hop candidates and the forwarding algorithm considers the state of the channel switch operation when selecting a next hop candidate. Also, in order to allow latency-sensitive packets to be transmitted before other packets, latency-awareness has been introduced to distinguish e.g. VoIP flows from FTP traffic. The ideas have been implemented and tested using real-world experiments, and the results show a significant reduction in latency.

A redução de custo com telefonia através do uso de voz sobre IP tem disparado a busca de soluções que transformem redes IP originalmente dedicadas a transporte de dados em redes para transporte de voz. Esta dissertação tem por objetivo apresentar uma metodologia para sistematizar a avaliação de redes para o tráfego de voz sobre IP de acordo com as possibilidades disponíveis no cenário a ser avaliado. Inicialmente é dada uma visão geral de voz sobre IP, apresentando os protocolos utilizados, os fatores que influenciam na qualidade da voz e os métodos de avaliação de qualidade da voz. Na seqüência são apresentados trabalhos correlatos a avaliação de qualidade de aplicações de voz sobre IP. E por fim descreve-se a proposta de uma metodologia para sistematizar a avaliação de redes com VoIP. / The use of voice over IP telephony was started with solutions to adapt existent data networks to carrier voice streams. The use of monitoring techniques, QoS and signaling protocols can be combined on a such design. Our goal is to present a methodology to evaluate and choose the probing points and the voice quality evaluation techniques to be used in network redesign. An overview about VoIP protocols and parameters that change the voice quality are presented as well as some related works on evaluating voice quality based on network parameters. A proposed methodology is presented, with a case study to show how one can choose the right combination of probing points with some voice quality measurement technique.

A redução de custo com telefonia através do uso de voz sobre IP tem disparado a busca de soluções que transformem redes IP originalmente dedicadas a transporte de dados em redes para transporte de voz. Esta dissertação tem por objetivo apresentar uma metodologia para sistematizar a avaliação de redes para o tráfego de voz sobre IP de acordo com as possibilidades disponíveis no cenário a ser avaliado. Inicialmente é dada uma visão geral de voz sobre IP, apresentando os protocolos utilizados, os fatores que influenciam na qualidade da voz e os métodos de avaliação de qualidade da voz. Na seqüência são apresentados trabalhos correlatos a avaliação de qualidade de aplicações de voz sobre IP. E por fim descreve-se a proposta de uma metodologia para sistematizar a avaliação de redes com VoIP. / The use of voice over IP telephony was started with solutions to adapt existent data networks to carrier voice streams. The use of monitoring techniques, QoS and signaling protocols can be combined on a such design. Our goal is to present a methodology to evaluate and choose the probing points and the voice quality evaluation techniques to be used in network redesign. An overview about VoIP protocols and parameters that change the voice quality are presented as well as some related works on evaluating voice quality based on network parameters. A proposed methodology is presented, with a case study to show how one can choose the right combination of probing points with some voice quality measurement technique.

This diploma thesis focuses on the issue of evaluation VoIP services (from the perspective of QoE) and their quality in university network. It includes theoretical basics of VoIP testing, principals of QoS, and also overview of QoE/MOS measurement methods. Next part describes the test bed and methodology used for measurements and evaluating of results. Site survey results and VoIP quality measurement are evaluated at the end, followed by recommendations that can lead to better VoIP quality in MENDELU wireless network.

Este trabalho tem como objetivo o estudo da tecnologia VoIP (Voz sobre IP) e a sua aplicação em redes corporativas, enfocando o planejamento de capacidade da rede de dados para absorver serviços VoIP. Serão apresentados tópicos sobre a fundamentação teórica de VoIP (Voz sobre IP), os requisitos de arquitetura de rede e QoS (Qualidade de Serviço) exigidos pelo serviço. Mostra-se também como a metodologia para planejamento de capacidade usado em telefonia convencional pode ser adaptada aos serviços VoIP em uma rede corporativa. Foi aplicada a metodologia adaptada através de um estudo de caso em uma rede corporativa real. / This work has as objective the study of capacity planning in corporate networks for the implementation of VoIP (Voice over IP) services. We will presents topics about the theorical background of VoIP, the requirements of architecture of network and QoS (Quality of Service) demanded by the service. It will also reveal how the methodology used for planning capacity in conventional telephony, could be adjusted to the VoIP services in a corporate network. The adjusted methodology was applied in a real corporate network.

VIEIRA, Leandro Kravczuk. Análise de desempenho de uma aplicação VoIP em redes veiculares. 2011. 136 f. Dissertação (Mestrado em ciência da computação)- Universidade Federal do Ceará, Fortaleza-CE, 2011. / Submitted by Elineudson Ribeiro (elineudsonr@gmail.com) on 2016-07-11T17:42:55Z No. of bitstreams: 1 2011_dis_lkvieira.pdf: 2177171 bytes, checksum: b12f234dbae0e70d63d4ffaacdf15aa0 (MD5) / Approved for entry into archive by Rocilda Sales (rocilda@ufc.br) on 2016-07-18T15:59:47Z (GMT) No. of bitstreams: 1 2011_dis_lkvieira.pdf: 2177171 bytes, checksum: b12f234dbae0e70d63d4ffaacdf15aa0 (MD5) / Made available in DSpace on 2016-07-18T15:59:47Z (GMT). No. of bitstreams: 1 2011_dis_lkvieira.pdf: 2177171 bytes, checksum: b12f234dbae0e70d63d4ffaacdf15aa0 (MD5) Previous issue date: 2011 / Vehicular networks have emerged as a particular case of mobile networks and then became a specific field of research in computer networks. They have been the subject of numerous scientific research in recent years, whose main focus is the development of Intelligent Transport System. Furthermore, given that cars are increasingly important in people's lives, smart board software in their cars can substantially improve the quality of life of users. This fact and the significant market demand for more reliability, security and entertainment in vehicles, has led to significant development and support for vehicular networks and their applications. Among these applications we can mention the use of VoIP, however, VoIP applications suffer from problems of delay, packet loss and jitter. These technical challenges are further aggravated when used in wireless networks. One factor that directly influences the use of an application in wireless networks is the routing protocol. Routing is a challenging task due to the high node mobility, the instability of wireless links and the diversity of scenarios. For this reason, several routing protocols have been designed with the goal of solving one or more specific problems of each scenario. However, although there are several proposed solutions to the problem routing in vehicular networks, no general solution was found, in other words, any proposed protocol obtained considerable performance in the various scenarios that exist in vehicular networks. Thus, in this paper, we analyze through simulations the impact of density, of the reach of transmission, the mobility and the type of routing protocol on the performance of a VoIP application in urban and highway scenarios of vehicular networks. / As redes veiculares surgiram como um caso particular de redes móveis e passaram a formar um campo específico de pesquisa na área de redes de computadores. Elas têm sido alvo de inúmeras pesquisas científicas nos últimos anos, cujo principal foco é o desenvolvimento do Sistema Inteligente de Transporte. Além disso, dado que os automóveis são cada vez mais importantes na vida das pessoas, embarcar softwares inteligentes em seus carros pode melhorar substancialmente a qualidade de vida dos usuários. Esse fato, somado à significante demanda do mercado por mais confiabilidade, segurança e entretenimento nos veículos, levou ao desenvolvimento e suporte significantes para as redes veiculares e suas aplicações. Dentre estas aplicações pode-se citar a utilização do VoIP. Entretanto, os aplicativos VoIP sofrem com problemas de atraso, perda de pacotes e jitter. Estes desafios técnicos se agravam ainda mais quando utilizado em redes sem fio. Um fator que influencia diretamente a utilização de uma aplicação em redes em fio é o protocolo de roteamento. O roteamento é uma tarefa desafiadora devido à alta mobilidade dos nós, à instabilidade dos enlaces sem-fio e a diversidade de cenários. Por essa razão, diversos protocolos de roteamento foram projetados com o objetivo de solucionar um ou mais problemas específicos de cada cenário. Entretanto, apesar de existirem várias soluções propostas para o problema do roteamento em redes veiculares, nenhuma solução geral foi encontrada, ou seja, nenhum protocolo proposto obteve desempenho considerável nos diversos cenários existentes nas redes veiculares. Sendo assim, nesta dissertação, analisamos através de simulações o impacto da densidade, do alcance de transmissão, da mobilidade e do tipo de protocolo de roteamento no desempenho de uma aplicação VoIP nos cenários urbano e de rodovia em redes veiculares.