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Dissertations / Theses on the topic 'VoIP'
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1
Schildt, Holger. "VoIP mit IAX." Universitätsbibliothek Chemnitz, 2004. http://nbn-resolving.de/urn:nbn:de:swb:ch1-200400518.
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Workshop "Netz- und Service-Infrastrukturen"
Das Inter-Asterisk eXchange (IAX)-Protokoll ermöglicht eine unproblematische Kommunikation zwischen IAX-fähigen VoIP-Systemen. In der Präsentation zu dem Vortrag werden das Protokoll vorgestellt und die Vorteile von IAX skizziert.
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Correia, Ricardo João Luís Marques. "Scrambler para VoIP." Master's thesis, Universidade de Aveiro, 2009. http://hdl.handle.net/10773/7364.
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Mestrado em Engenharia Electrónica e Telecomunicações<br>A necessidade de se preservar a confidencialidade numa conversa telefónica é
um facto dos nossos dias.
Pretende-se com este trabalho dar uma resposta a este problema propondo
um sistema original de scrambling do sinal no domínio das frequências. O
sistema inclui uma troca de chaves públicas que geram uma chave secreta
comum entre o emissor e o receptor, baseado no método de Diffie-Hellman.
Além da implementação apresentam-se resultados de testes efectuados sobre
o sistema de scrambling proposto associado a vários codecs de uso comum.<br>The need to preserve confidentiality in a telephone conversation is, nowadays,
a motive of concern for most of us. This work pretends to address this issue by
proposing an original signal scrambling system in the frequency domain. The
system includes a public key exchange which generates a common secret key,
based on the Diffie-Hellman method, at the transmitter and the receiver.
The implementation and some results of the proposed solution using common
conventional codecs are presented.
3
Pini, Fabio. "VoIP su BlackBerry." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2010. http://amslaurea.unibo.it/1631/.
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Lembard, Tomáš. "Speciální aplikace VoIP." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-219188.
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The aim of this master's thesis is suggestion and following realization of voice transmission over the local network equipment and a description of used circuits and solutions in terms of hardware and software. This thesis deals with digitization of low-frequency signals, the structure of IP and UDP protocols, implementation of TCP/IP stack cIPS
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Kristensveen, David. "Sikkerhet i VoIP-portnere." Thesis, Norwegian University of Science and Technology, Department of Telematics, 2006. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-10267.
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<p>Session Initiation Protocol(SIP) er i ferd med å bli den ledende signaleringsprotokollen i forbindelse med IP-telefoni. SIP benyttes til å initiere, modifisere og terminere interaktive sesjoner. Arkitekturens to hovedkomponenter er servere og brukeragenter. De ulike brukeragentene utveksler forespørsler og tilhørende responsmeldinger. Etter at en sesjon er satt opp av SIP benyttes Real Time Protocol(RTP) til å overføre data i forbindelse med selve samtalen. RTP benytter dynamisk valgte portnumre. Disse utveksles på forhånd mellom brukeragentene ved hjelp av Session Description Protocol(SDP) i meldingskroppen til SIP-meldingene. Når brannmurer eller Network Address Translation(NAT) benyttes sammen med IP-telefoni er det et som regel et problem at IP-adresser og portnumrene som skal benyttes av brukeragentene omskrives. Adresseinformasjonen som er utvekslet på forhånd vil derfor bli ugyldig. Det finnes flere ulike metoder for å tilnærme seg disse problemene. To metoder fra IETF er Simpel Traversal of UDP through NAT(STUN) og Traversal using Relay NAT(TURN). Session Border Controllers(SBC) er lukkede kommersielle nettverksløsninger som benyttes til å løse mange av de samme problemene. Ulempen med SBC er at dette er kostbare løsninger og at prinsippet om at SIP skal være en åpen protokoll brytes. Innefor IP-telefoni ser en utvikling der tilbydere eller organisasjoner har infrastrukturen på plass for å realisere en IP-telefonitjeneste på vegne av sine egne brukere, mens samtrafikk med andre er nødt til å foregå ved hjelp av PSTN-nettet. For å realisere en slik samtrafikk er en nødt til å benytte en Gateway. En slik Gateway har oftest en todelt funksjonalitet. Først må det oversettes mellom signaleringen i IP-telefoni(SIP) og signaleringen i PSTN(som regel ISUP). Dette utføres av en signaliseringsgsgateway(SG). Deretter må mediastrømmen oversettes av en mediagateway(MG) fra RTP til det aktuelle formatet som benyttes av det linjesvitsjede nettverket. En Media Gateway Controller(MGC) benyttes for å samkjøre MG og SG. Telephony Routing over IP(TRIP) er en protokoll som kan benyttes av tjenestetilbydere eller organisasjoner for å utveksle rutingtabeller for sine respektive gatewayer. En annen mekanisme for PSTN til IP samtrafikk er SIP for Telephones(SIP-T). Her kan PSTN-signaleringen enten oversettes til SIP eller pakkes inn i SIP-meldinger. En av hovedgrunnene til at PSTN benyttes til ruting i IP-telefon skyldes at det ikke har eksistert noen fullgod erstatter for Signaleringssystem nummer syv(SS7) i forbindelse med IP-telefoni. E.164 Number Mapping(ENUM) er en metode for å oversette E.164-numre til domenenavn ved hjelp av DNS. Ved en slik løsning vil en være i stand til å realisere samtrafikk mellom ulike typer IP-telefoninettverk uten å benytte PSTN. I motsetning til tradisjonelle løsninger der mange i dag benytter to ulike nettverk for data- og taletjenester vil en fremover kunne se en utvikling der alle tjenester er basert på IP og PSTN vil bli overflødig. Samtidig ser en utvikling der Internett løsninger som Skype blir stadig mer populære. Det er også muligheter for å benytte SIP til tilsvarende løsninger, men her er det fortsatt en del uenighet i om hvordan slike løsninger konkret skal realiseres. The IP Multimedia Subsystem(IMS) er et rammeverk som skal sørge for ytterligere konvergens mellom telefoni- og datatjenester. Telefonikunder skal tilbys nye multimediatjenester på apllikasjonsnivå. Kjernenettverket i IMS skal benytte SIP-baserte løsninger. IMS omtales ofte som neste generasjons nettverk. Etter hvert som 802.11 nettverk er blitt mer utbredt er det et voksende marked for IP-telefoni over 802.11. Unlicensed Mobile Access(UMA) er en teknologi som lar brukeren benytte mobiltelefonen over 802.11 eller Bluetooth. Samtidig tillates det handover mot det mobile nettverket når dette er nødvendig. Sikkerhet i trådløse nettverk er et område under stadig utvikling. Ved innføring av 802.11i vil en få sterkere mekanismer for sikkerhet i form av Extended Authentication Protocol(EAP) for autentisering og Advanced Encryption Standard(AES) for kryptering. Ved introduksjonen av ENUM oppstår det nye sikkerhetsutfordringer. Et system med billigere telefoni og enkel tilgangs til brukerlokasjoner vil potensielt kunne føre til mer Spam over IP-telefoni(SPIT). Ellers er DNS systemet utsatt for mange potensielle trusler.</p>
6
Saad, Amna. "Secure VoIP performance measurement." Thesis, Loughborough University, 2013. https://dspace.lboro.ac.uk/2134/13426.
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This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality.
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Sitolino, Claudio Luis. "VOIP : um estudo experimental." reponame:Biblioteca Digital de Teses e Dissertações da UFRGS, 2001. http://hdl.handle.net/10183/3182.
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Voz sobre IP (VoIP) é uma tecnologia que permite a digitalização e a codificação da voz e o empacotamento em pacotes de dados IP para a transmissão em uma rede que utilize o protocolo TCP/IP. Devido ao volume de dados gerados por uma aplicação VoIP, esta tecnologia se encontra em funcionamento, em redes corporativas privadas. Mas se a rede base para o transporte desta aplicação for a Internet, certamente, não deve ser utilizada para fins profissionais, pois o TCP/IP não oferece padrões de QoS (Qualidade de Serviço) comprometendo desta forma a qualidade da voz. A qualidade da voz fica dependente do tráfego de dados existentes no momento da conversa. Para realizar um projeto de VoIP é necessário conhecer todo o tráfego existente na rede e verificar o quanto isto representa em relação à banda total da rede. Também se deve conhecer o tipo de aplicação que se deseja implantar, verificando a banda a ser utilizada por esta, e então projetar como a rede deverá ser estruturada. Para auxiliar no projeto de VoIP, pretende-se mostrar o que está sendo desenvolvido para que o protocolo TCP/IP ofereça QoS e uma ferramenta para análise do tráfego de voz sobre redes TCP/IP e também análises dos resultados obtidos em experimentos simulando diversas situações práticas.
8
Бубнов, Ігор Васильович, Игорь Васильевич Бубнов, Ihor Vasylovych Bubnov та Н. В. Москаленко. "Практическое использование технологии VoIP". Thesis, Издательство СумГУ, 2008. http://essuir.sumdu.edu.ua/handle/123456789/7477.
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Нами для проведения работ по использованию технологии VoIP была избрана бесплатно распространяемая программа Echolink используемая в радиолюбительской практике. Разработчиком системы является Джонатан Тейлор (K1RFD).
При цитировании документа, используйте ссылку http://essuir.sumdu.edu.ua/handle/123456789/7477
9
Kulička, Vojtěch. "VoIP in Jabber Client." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2011. http://www.nusl.cz/ntk/nusl-237034.
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Práce se zabývá možnostmi implementace VoIP do existujícího XMPP programu se sdílenou tabulí. Analyzuje možnosti využití současných technologií pro podporu VoIP. Cílem je nahrazení stávajících komunikačních knihoven klienta za telepathy. Dále také přidání VoIP.
10
Havelka, Ondřej. "Analyzátor kvality VoIP hovorů." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2011. http://www.nusl.cz/ntk/nusl-237055.
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This master thesis deals with the design and implementation of an application for analyzing Voice over IP quality using NetFlow. In the beginning, there is summarized basic information about VoIP technology and NetFlow - its principles, the most used protocols, factors that have influence on call quality and call quality rating methods. Later there is presented proposal of application and then described its implementation. The created application was tested on samples, which simulate calls in network with delays and packet-loss. Within testing was made the comparison with commercial application and the results are discussed.
11
Di, Muro Antonello. "Un'applicazione VoIP per Symbian." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2010. http://amslaurea.unibo.it/1229/.
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Salucci, Luca. "Un'applicazione VoIP per BlackBerry." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/2713/.
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Rosenberg, Martin. "Vysokorychlostní přenos dat v mobilních a bezdrátových sítích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-218989.
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The goal of the master’s thesis was to propose two laboratory exercises, integrating newly purchased devices HTC Desire and Nokia N900. Designed tasks bring new technologies and services to education process. The first task examines the configuration of branch exchange Asterisk PBX and analyzes SIP protocol. Part of exercise is concentrated on high-speed data transmission in mobile and wireless networks, regarding to usability of VoIP technology. The second exercise introduces to vulnerability of VoIP technology. It contains simulations of attacks on branch exchange Asterisk, DoS attack and discusses methods to secure VoiP communication. The part of this exercise examines usage of HTC Desire phone, instead of ordinary Wi-Fi access point.
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Azfar, Abdullah. "Multiple Escrow Agents in VoIP." Thesis, Norwegian University of Science and Technology, Department of Telematics, 2010. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-10895.
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Using a Key escrow agent in conjunction with Voice over IP (VoIP) communication ensures that law enforcements agencies (LEAs) can retrieve the session key used to encrypt data between two users in a VoIP session. However, the use of a single escrow agent has some drawbacks. A fraudulent request by an evil employee from the LEA can lead to improper disclosure of a session key. After the escrow agent reveals the key this evil person could fabricate data according to his/her needs and encrypt it again (using the correct session key). In this situation the persons involved in the communication session can be accused of crimes that he or she or they never committed. The problems with a single escrow agent becomes even more critical as a failure of the escrow agent can delay or even make it impossible to reveal the session key, thus the escrow agent might not be able to comply with a lawful court order or comply with their escrow agreement in the case of data being released according to this agreement (for example for disaster recovery).This thesis project focused on improving the accessibility and reliability of escrow agents, while providing good security. One such method is based on dividing the session key into m chunks and escrowing the chunks with m escrow agents. Using threshold cryptography the key can be regenerated by gathering any n-out-of-m chunks. The value of m and n may differ according to the role of the user. For a highly sophisticated session, the user might define a higher value for m and n for improved, availability, reliability, and security. For a less confidential or less important session (call), the value of m and n might be smaller. The thesis examines the increased availability and increased reliability made possible by using multiple escrow agents.
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Dzaferagic, Samir. "Secure Session Mobility for VoIP." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91676.
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High data rate wireless packet data networks have made real-time IP based services available through mobile devices. At the same time, differences in the characteristics of radio technologies (802.11/WiFi and 3G networks) make seamless handoff across heterogeneous wireless networks difficult. Despite this, many believe that the ultimate goal of next generation networks (often referred to as the fourth generation) is to allow convergence of such dissimilar heterogeneous networks. Supporting voice over Internet Protocol in next-generation wireless systems is thought by some to require support for mobility and quality of service features. Currently a mobile node can experience interruptions or even sporadic disconnections of an on going real-time session due to handovers between both networks of different types and networks of the same type. Many tests have already been done in this area and one may wonder why it is worth spending even more time investigating it? This thesis focuses on the important problem of providing session security despite handovers between networks (be they operated by the same operator or different operators and be they the same link technologies or different). One of the goals in this thesis is to investigate how an ongoing speech session can continue despite a change in transmission media1. Additionally, a number of security threats that could occur due to the handover will be identified and presented. Finally, the most suitable solution to address these threats will be tested in a real environment. Eventual shortcomings and weaknesses will be identified and presented; along with suggestions for future work. 1 When utilizing IP over carriers such as wired Ethernet, WLAN, and 3G.<br>Trådlösa hög-hastighets datanät har möjliggjort appliceringen av realtids tjänster på mobil utrustning över IP. Samtidigt har skillnaderna i de olika radioteknologierna (802.11/WiFi och 3G näten) introducerat nya problem med att upprätthålla trådlösa kommunikationen tvärs den heterogena trådlösa accessen. Många tror att slutmålet för nästa generations nätverk (ofta refererade som fjärde generationens nätverk) är att tillåta konvergensen av dessa olika heterogena nätverk. Stödet för Voice over Internet Protokollet (VoIP) i nästa generations trådlösa nät tror somliga kräver ett inslag av kombination mellan mobilitet samt upprätthållandet av kvaliteten. För närvarande kan den mobila noden (MN) råka ut för störningar och även sporadiska avbrott av en pågående realtidssessionen på grund av övergångar mellan samma eller olika typer av medier. Många tester har redan gjorts inom det här området och man kan fråga sig varför det är värt att lägga ner ännu mer tid på att undersöka det här? Det här examensarbetet fokuserar på det viktiga problemet som handlar om att kunna erbjuda sessions säkerhet trots övergångar mellan näten (oavsett om dessa drivs av samma eller olika operatörer samt oavsett om de är av samma eller olika nätverks typ). Ett av målen för det här examensarbetet är att undersöka hur en pågående talsession behålls vid byte av transmissionsmedia2. Vidare kommer olika säkerhetsaspekter och hot som kan tänkas uppstå vid bytet att identifieras och presenteras. Slutligen kommer den mest lämpade lösningen till problemet att testas i verklig miljö. Eventuella brister och svagheter kommer att identifieras och redovisas i slutet av rapporten tillsammans med förslag på framtida arbete. 2 Då man nyttjar IP bärare som trådbundet Ethernet, WLAN och 3G.
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Švarc, Lukáš. "VoIP v bezdrátové síti VŠE." Master's thesis, Vysoká škola ekonomická v Praze, 2015. http://www.nusl.cz/ntk/nusl-203821.
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The diploma thesis is focused on exploring the possibility of VoIP service in a wireless network of University of Economics, Prague. This thesis describes the basic principles of VoIP and related wireless technologies necessary for its quality and stable operation. Subsequently, different configurations of wireless network and end clients are tested and compared, including its impact on ordinary users, in a laboratory environment with idle and fully utilized frequency band. Finally, a roaming operation with the use of several advanced 802.11 standards is tested in the real environment of the Old building in Žižkov. In conclusion, the ideal settings for all telecommunication devices are recommended in order to maximize the quality of VoIP operation and to minimize the negative impact on ordinary users.
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Massaccesi, Giulio. "Un'applicazione VoIP per Symbian: architettura." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/1935/.
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Borsari, Marcello. "Multihoming su Symbian per VoIP." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2010. http://amslaurea.unibo.it/1436/.
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Maestrini, Mattia. "Client VoIP su Windows Mobile." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2012. http://amslaurea.unibo.it/3883/.
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Quando si parla di VoIP ci si riferisce ad un insieme di protocolli di comunicazione, tecnologie e metodi di trasmissione che permettono di effettuare conversazioni telefoniche attraverso reti a commutazione di pacchetto basata su IP, come Internet. Si tratta di una tecnologia che ha subito un forte crescita sia in ambito lavorativo che in ambito privato, questo fenomeno è in maggior parte dovuto al successo di applicazioni commerciali come Skype. Anche i dispositivi mobili hanno avuto un grande sviluppo e diffusione, sono passati da essere semplici telefoni cellulari a dispositivi in grado di fornire all’utente funzionalità avanzate come ad esempio navigazione internet, posta elettronica, riproduzione video, possibilità di installare applicazioni aggiuntive. Inoltre anche le reti dati sono migliorate in maniera considerevole negli ultimi anni, offrendo agli utenti una larghezza di banda sempre maggiore anche in mobilità. Tutti questi fattori hanno portato ad una crescente richiesta di applicazioni per dispositivi mobili in grado di sfruttare il VoIP. Per questi motivi si è deciso di progettare e sviluppare un applicazione VoIP per Windows Mobile, che offra tutte le funzioni necessarie ad un uso completo del VoIP e con un’interfaccia utente sia di facile utilizzo, per permettere anche agli utenti meno esperti di poter utilizzare la tecnologia VoIP
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Di, Lullo Giuseppe. "Applicazioni Voip per Android: Analisi." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2012. http://amslaurea.unibo.it/3916/.
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Jilani, Khurram. "Performance Analysis of VOIP and VOIP codec with Economical Policy For Call Centers in Pakistan." Thesis, Mittuniversitetet, Avdelningen för informations- och kommunikationssystem, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:miun:diva-23207.
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As information can be sent via various mediums through different channels, visual or vocal communication can also be carried out though various channels such as public switched networks (PSTN) and packet based switched systems which are commonly known as voice over Internet protocol (VOIP). This particular study emphasizes on voice over Internet protocol and its various aspects. Codec is the most important aspect in VOIP communication there is variety of codecs available. In order to have right quality of voice the right codec must be selected. There are various other factors such as bandwidth and right technical infrastructures which are also present and which must be addressed in order to have a better quality of voice over VOIP. Mostly, VOIP is used in companies where a huge volume of voice calls is needed, especially in call centers which deal in offshore sales and support projects. The main factor which make companies and other users interested in VOIP is the financial aspect as, VOIP cuts costs significantly. Every mode which may be utilised for voice communication has its own advantages and disadvantages so user can chose the mode of communication according to what is required. In order to provide the financial and technical advantages of VOIP, various previous studies have been studied, information has been collected from different call centers and a network is simulated using OPNET MODELLER in order to evaluate the performance of VOIP codec as the codec is basically coder as well as decoder, hence it is the basic aspect in order to have a better quality of services (QOS) and quality of voice (QOV).
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Deng, Xianglin. "Security of VoIP : Analysis, Testing and Mitigation of SIP-based DDoS attacks on VoIP Networks." Thesis, University of Canterbury. Computer Science and Software Engineering, 2008. http://hdl.handle.net/10092/2227.
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Voice over IP (VoIP) is gaining more popularity in today‟s communications. The Session Initiation Protocol (SIP) is becoming one of the dominant VoIP signalling protocol[1, 2], however it is vulnerable to many kinds of attacks. Among these attacks, flood-based denial of service attacks have been identified as the major threat to SIP. Even though a great deal of research has been carried out to mitigate denial of service attacks, only a small proportion has been specific to SIP. This project examines the way denial of service attacks affect the performance of a SIP-based system and two evolutionary solutions to this problem that build on each other are proposed with experimental results to demonstrate the effectiveness of each solution. In stage one, this project proposes the Security-Enhanced SIP System (SESS), which contains a security-enhanced firewall, which evolved from the work of stage one and a security-enhanced SIP proxy server. This approach helps to improve the Quality-of-Service (QoS) of legitimate users during the SIP flooding attack, while maintaining a 100 percent success rate in blocking attack traffic. However, this system only mitigates SIP INVITE and REGISTER floods. In stage two, this project further advances SESS, and proposes an Improved Security-Enhanced SIP System (ISESS). ISESS advances the solution by blocking other SIP request floods, for example CANCEL, OK and BYE flood. JAIN Service Logic Execution Environment (JAIN SLEE) is a java-based application server specifically designed for event-driven applications. JAIN SLEE is used to implement enhancements of the SIP proxy server, as it is becoming a popular choice in implementing communication applications. The experimental results show that during a SIP flood, ISESS cannot only drop all attack packets but also the call setup delay of legitimate users can be improved substantially compared to and unsecured VoIP system.
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Gibeli, Luís Henrique. "Construção de baselines para gerência de sistemas voip = Construction of baselines for voip systems management." [s.n.], 2012. http://repositorio.unicamp.br/jspui/handle/REPOSIP/259471.
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Orientadores: Leonardo de Souza Mendes, Gean Davis Breda<br>Dissertação (mestrado) - Universidade Estadual de Campionas, Faculdade de Engenharia Elétrica e de Computação<br>Made available in DSpace on 2018-08-21T07:12:20Z (GMT). No. of bitstreams: 1
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Previous issue date: 2012<br>Resumo: As modernas redes de comunicações são compostas pela interconexão de um grande número de redes heterogêneas, capazes de suportar múltiplos serviços e aplicações. Muitos destes serviços, como VoIP e videoconferência, são sensíveis a latência. Desta forma, a crescente demanda por este tipo de serviço através da Internet impõe o desenvolvimento de redes capazes de oferecer qualidade de serviço para suportá-los. Estas redes requerem ferramentas específicas de gerenciamento. Quando uma chamada VoIP é realizada através da Internet, um bilhete de tarifação é gerado produzindo informações específicas desta chamada. Estes bilhetes são chamados de IP Detail Records (IPDR). O IPDR gerado em cada chamada VoIP contém informações relacionadas ao seu histórico. Estas informações, além de descreverem o que aconteceu com a chamada, oferecem informações valiosas a respeito do estado da rede. Assim, os IPDRs podem ser utilizados para estabelecer baselines do tráfego VoIP na rede. Este trabalho apresenta um modelo de gerência de sistemas VoIP baseado no desenvolvimento de baselines. O objetivo do trabalho é introduzir uma nova abordagem que procura resgatar os conceito utilizados na telefonia pública comutada para ajudar a gerência VoIP. Um estudo de caso foi desenvolvido na Rede Metropolitana de Pedreira, interior de São Paulo, onde o tráfego de chamadas VoIP pôde ser analisado através dos baselines criados<br>Abstract: Modern communications networks are formed by the interconnection of an immense number of complex and heterogeneous networks, capable of supporting multiple services and applications. Many of these services, like VoIP or videoconferencing, are latency sensitive. Thus, the increasing demand for latency sensitive services through the Internet imposes the development of networks capable of delivering quality of service specific for dealing with synchronous. These networks require the use of specific management tools. When a VoIP call occurs upon the Internet, a ticket (a file record) is generated to register information regarding that specific call. These files are called Internet Protocol Detail Record (IPDR). The IPDR, which is generated for every VoIP call, contain information related to the call's history. The full set of information in the IPDRs carries a very complete description of what happened to the call and can provide valuable information about the state of the network during the history of the call. Therefore, IPDRs can be used to establish baselines for the VoIP traffic within network. This work presents a management VoIP system model based on development of a baseline. The target of this work is to introduce a new methodology based on concepts used in the Public Switched Telephone Network to help VoIP management. A case study was developed in the city Pedreira-SP, where the VoIP traffic could be managed through the baselines created<br>Mestrado<br>Telecomunicações e Telemática<br>Mestre em Engenharia Elétrica
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Broström, Anders, and Niclas Kihlstadius. "Prototyp av en VoIP/PSTN-gateway." Thesis, Karlstad University, Division for Information Technology, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-2646.
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<p>Under de senaste åren har Internettelefonin varit på frammarsch, och i takt med att tekniken mognat har fler och fler börjat se den som ett alternativ till att ringa via telefonnätet. Förutom att det är billigare att ringa över det förstnämnda, så erbjuder Internettelefonin också en rad revolutionerande tjänster. Det är dock troligt att telefonnätet kommer att få tjänstgöra i många år till, och det erbjuder fortfarande överlägset bäst stabilitet och har stor acceptans. Om de två telefoninätverken ska existera sida vid sida, med varsina användarbaser är det lämpligt om de kan fås att samverka, så att användare av det ena kan ringa användare av det andra, och vice versa. Detta kan göras med en VoIP/PSTN-gateway, som översätter kontrollinformation och rösttrafik mellan de två nätverken.</p><p>Uppsatsen handlar om det arbete vi har utfört år TietoEnator i Karlstad. Uppgiften bestod i att utveckla en prototyp av en VoIP/PSTN-gateway. Från början var det avsett att systemet skulle klara uppringning från endera en ”vanlig” telefon, eller en så kallad IP-telefon. Därtill skulle rösttrafiken överföras genom ändamålsenlig hårdvara. För att utföra arbetet behövde vi först studera relevanta kommunikationsprotokoll både för telefonnätet och för Internet, för att se hur dessa kunde fås att samverka. Vi behövde också lära oss tillgängliga system, bibliotek och verktyg för att förstå hur vi skulle skapa vårt eget system i den efterkommande implementeringsfasen. På grund av en lång inläsningsperiod och inledande tekniska problem, samt att nödvändig hårdvara för översättning av rösttrafiken inte anlände i tid begränsades arbetet till att innefatta samtal initierade från den vanliga telefonen till ip-telefonen, utan röstöverföring. Likväl har ett resultatgivande arbete utförts, och det beskrivs i detalj i rapporten.</p><br><p>During the past few years Internet telephony has advanced rapidly, and as the technology has evolved, more and more have come to consider it an alternative to making phone calls through the telephone network. Besides being cheaper, Internet telephony also provides several revolutionary services. It is likely though that the telephone network will remain in use for several years to come, and it still offers by far the best stability and is accepted by most people. If the two networks are to coexist, with their respective users, it would be useful if they could be made to interact, so that users of one network can call users of the other, and vice versa. This can be done with a VoIP/PSTN gateway, which translates control information and voice traffic between the two networks.</p><p>Our dissertation is about the work we have performed for TietoEnator in Karlstad. The assignment was to develop a prototype of a VoIP/PSTN gateway. Initially the system was meant to support phone calls initiated either from an “ordinary” phone or from an IP telephone. Also the voice traffic was supposed to be translated with the use of appropriate hardware. To manage this we first needed to study all the relevant protocols for communication used in the telephone network and on the Internet, to get an idea of how these could be made to interact. We also had to learn existing systems, libraries and tools in order to see how we could create our own system. Due to a long learning period and technical problems in the beginning, and because the necessary hardware equipment for translation of voice traffic did not arrive in time, the assignment was limited to include only calls initiated from the ordinary phone to the IP telephone, without voice transmission. Never the less, the efforts have produced results, and our work is explained in detail in this dissertation.</p>
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Lindecrantz, Mikael, and Marcus Junström. "VoIP som kommunikationsplattform : - Tjänster och möjligheter." Thesis, Linköping University, Department of Management and Engineering, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-9700.
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<p>Begreppet Voice over Internet Protocol (VoIP) är något som nog de flesta har hört talas om, men vad innebär egentligen VoIP. Vi har tittat på tekniken och återger en beskrivning av begreppet VoIP. Vi har valt att titta på hur företag använder tekniken samtidigt som vi tittar på vilka nya tjänster och möjligheter som finns. Hur ser tjänsteutbudet ut idag och hur påverkar detta sättet att kommunicera inom företagen? Vilken funktionalitet används och hur drar man nytta av de fördelar och möjligheter som den nya tekniken erbjuder? Med förändrade kommunikationssätt ser vi hur verksamheter både kan effektiviseras och utvecklas.</p><p>När vi tittar på hur leverantörer bidrar till att utveckla sina kunders verksamheter så ser vi att leverantörerna inte är delaktiga inom detta område i någon större utsträckning. Vi ser dock att man inte hunnit så långt i utvecklingen av VoIP, samt att leverantörer har vissa problem med att nå ut och marknadsföra de tjänster och möjligheter den nya tekniken faktiskt erbjuder. Fokus ligger istället mer på de kostnadsbesparingar man gör genom att kommunicera över internet (IP) istället för med traditionell telefoni (PSTN).</p><p>Tekniken är beroende av kvaliteten på bredbandsuppkoppling samt den interna infrastrukturen hos företagen. Mycket av de problem så som eko och dålig samtalskvalitet beror på undermålig utrustning internt på företagen. Dessa problem ser man i branschen som något övergående då ny hårdvara utvecklas och förses med bättre stöd för VoIP. Många leverantörer erbjuder även sina kunder en helhetslösning för att få kontroll över både telefoni och internetförbindelse.</p><p>Slutligen ser vi att VoIP öppnar upp för en omstöpning av hur företag kommunicerar, möjligheterna har bara börjat utforskas. Som teknik är VoIP idag inget nytt, det som är nytt är att företagen först nu börjar se möjligheterna till att effektivisera, utveckla och förändra sina kommunikationsvägar. För att utfallet ska bli så bra som möjligt krävs ett helhetsgrepp på hur VoIP implementeras, vi menar att man måste se till hur man kan förändra sitt arbetssätt, inte bara hur man ringer till en lägre kostnad.</p>
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Boongerd, Sanhawad, and Fredrik Lindstein. "Analys av datakommunikationssäkerhet för VoIP-protokoll." Thesis, KTH, Data- och elektroteknik, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-105774.
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Voice over IP (VoIP) is a relatively new technology that enables voice calls over data networks.With VoIP it is possible to lower expenses, and increase functionality and flexibility. FromSwedish Armed Forces point of view, the security issue is of great importance, why the focus inthis report is on the security aspect of the two most common open-source VoIP-protocols H.323and SIP, some of the most common attacks, and counter-measures for those attacks.Because of the level of complexity with a network running H.323 or SIP, and the fact that it hasyet to stand the same level of trial as of traditional telephony, a VoIP-system includes manyknown security-issues, and probably at present many unknown security flaws. The conclusion is that it takes great knowledge and insight about a VoIP-network based onH.323 or SIP to make the network satisfyingly safe as it is today, and is therefore perhaps not asuitable solution for the Swedish Armed Forces today for their more sensitive communications.<br>Voice over IP (VoIP) är en datakommunkationsteknik som möjliggör röstsamtal överdatanätverk. Med VoIP är det möjligt att sänka kostnader, utöka funktionalitet och flexibilitet.Från Försvarsmaktens perspektiv är säkerhetsfrågan med VoIP av stor vikt, därför läggs speciellfokus för denna rapport på säkerhetsaspekten av de två största öppna VoIP-protokollen H.323och SIP, några av de vanligaste attackerna, och åtgärder mot dessa attacker. Eftersom uppbyggnaden av ett H.323- eller SIP-baserat nätverk är komplext och inte allsbeprövat i samma utsträckning som traditionell telefoni, innehåller det många kända säkerhetshåloch förmodligen för närvarande många okända säkerhetsbrister. Slutsatsen är att det krävs mycket stor kunskap och insikt hur ett VoIP-nätverk baserat på H.323eller SIP fungerar för att göra nätverket tillräckligt säkert i nuläget, vilket gör det till en tveksamttillfredställande lösning för Försvarsmakten idag för deras kommunikation av känsligare slag.
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Li, Zhang. "Service Improvements for a VoIP Provider." Thesis, KTH, Kommunikationssystem, CoS, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91493.
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This thesis project is on helping a Voice over Internet Protocol (VoIP) service provider by improving server side of Opticall AB's Dial over Data solution. Nowadays, VoIP is becoming more and more popular. People use VoIP to call their family and friends every day. It is cheap, especially when users are abroad, because that they do need to pay any roaming fee. Many companies also like their employees to use VoIP, not only because the cost of calling is cheap, but using VoIP means that the company does not need a hardware Private Branch eXchange (PBX) -- while potentially offering all of the same types of services that such a PBX would have offered. As a result the company can replace their hardware PBX with a powerful PC which has Private Branch eXchange PBX software to connect all the employees and their VoIP provider. At the VoIP provider’s side, the provider can provide cheap calls for all users which are connected by Internet. The users can initialize and tear down a session using a VoIP user agent, but how can they place a VoIP call from a mobile device or other devices without a VoIP user agent? Users want to place cheap VoIP call everywhere. VoIP providers want to provide flexible solution to attract and keep users. So they both want to the users to be able to place cheap VoIP call everywhere. Although VoIP user agent are available for many devices as a software running on a computer, a hardware VoIP phone, and even in some mobile devices. However, there are some practical problems with placing a VoIP call from everywhere. The first problem is that not every device can have a VoIP user agent. But if you do not have a VoIP user agent on your device, then it would seem to be difficult to place a VoIP call. The second problem is that you have to connect to a network (probably Internet) to signal that you want to place a call. Thus at a minimum your device has to support connecting to an appropriate network. If your device is connecting to a mobile network, you can send signaling to set up a VoIP call through General Packet Radio Service (GPRS). However, the bandwidth and delay of the GPRS networks of some mobile operators is not suitable for the transfer of encoded voice data, additionally, some mobile operators charge high fees for using GPRS. All of these problems make placing VoIP calls via a mobile device difficult. However, if your mobile device has a VoIP user agent and you have suitable connectivity, then you can easily use VoIP from your mobile device[.] To provide a flexible solution to VoIP everywhere -- even to devices that do not or can not have a VoIP user agent, Opticall AB has designed Dial over Data (DoD) solution. By using this solution, you can place a VoIP call from your mobile device or even fixed phone -- without requiring that the device that you use have a VoIP user agent. This solution also provides a central Internet Protocol-Private Branch eXchange (IP-PBX) which can connect call to and from to Session Initiation Protocol (SIP) phones. Both individuals and companies can use this solution for call cost savings. Max Weltz created the existing DoD solution in an earlier thesis project. This thesis [1] provides a good description of the existing DoD solution. As a result of continued testing and user feedback, Opticall AB has realized that their DoD solution needs to be improved in several area. This thesis project first identified some of the places where improvement was needed, explains why these improvements are necessary, and finally designs, implements, and evaluates these changes to confirm that they are improvements. An important result of this thesis project was a clear demonstration of improvements in configuration of the solution, better presentation of call data records, correct presentation of caller ID, and the ability to use a number of different graphical user interfaces with the improve DoD solution. These improvements should make this solution more attractive to the persons who have to maintain and operate the solution.<br>Detta examensarbete behandlar förbättringar i serversidan av OptiCall ABs “Dial over Data” (DoD) lösning som tillhandahålls för tjänsteleverantörer av VoIP. VoIP blir mer och mer populärt. Människor använder VoIP för att ringa till sin familj och vänner varje dag. Det är billigt, särskilt när användaren är utomlands, eftersom de inte behöver betala någon roamingavgift. Många företag vill också att deras anställda skall använda IP-telefoni, inte bara därför att kostnaden för att ringa oftast är lägre, utan för att bolaget kan ersätta sin traditionella företagsväxel (PBX) med en kraftfull dator som har PBX programvara för att även ansluta alla anställda till deras VoIP leverantör. VoIP leverantören kan erbjuda billiga samtal till alla användare som också är anslutna via Internet. Användarna kan hantera VoIP samtal med en VoIP user agent, men hur kan de ringa ett VoIP-samtal från en mobil enhet eller andra enheter utan VoIP user agent? Användare vill kunna ringa billiga VoIP-samtal överallt. VoIP-leverantörer vill erbjuda en flexibel lösning för att locka och behålla användare. Även VoIP user agent finns utvecklade till många enheter som en programvara som körs på en dator, en hårdvara VoIP-telefon, och även i vissa mobila enheter. Men det finns vissa praktiska problem med att ringa ett VoIP-samtal från alla platser. Det första problemet är att inte varje enhet kan ha en VoIP user agent. Det andra problemet är att den måste ansluta till ett nätverk (troligen Internet) för att signalera att den vill ringa ett samtal. Om din enhet ansluter till ett mobilnät, kan du skicka signalerar att upprätta ett VoIP-samtal via General Packet Radio Service (GPRS). Dock är bandbredden och fördröjningen i GPRS-nät i vissa operatörers nät inte lämpliga för överföring av tal, dessutom tar vissa mobiloperatörer ut höga avgifter för att använda GPRS. Alla dessa problem gör det svårt att hantera VoIP-samtal via en mobil enhet. Men om din mobila enhet har en VoIP user agent och du har lämplig nätanslutning så kan du enkelt använda VoIP från din mobiltelefon[.] För att erbjuda en flexibel VoIP lösning överallt - även på enheter som inte kan ha en VoIP user agent har OptiCall AB utformad “Dial over Data” (DoD). Genom att använda denna lösning kan du initiera ett VoIP-samtal från din mobiltelefon eller fast telefon - utan att kräva att den enhet som du använder har en VoIP user agent. Denna lösning inkluderar också en central Internet Protocol-Private Branch Exchange (IP-PBX) som kan koppla samtal till och från Session Initiation Protocol (SIP) telefoner. Både privatpersoner och företag kan använda denna lösning för att minska samtalskostnader. Max Weltz vidareutvecklade den befintliga DoD lösning i ett tidigare examensarbete. Denna avhandling [1] ger en god beskrivning av den befintliga DoD lösning. Som ett resultat av fortsatt testning samt synpunkter från användarna har OptiCall AB insett att deras DoD lösning måste förbättras på flera områden. Detta examensarbete har i första hand identifierat några områden där förbättringar behövdes, förklarat varför dessa förbättringar är nödvändiga, och slutligen utvecklat och utvärderat dessa förändringar. Ett viktigt resultat av detta examensarbete visades av en tydlig demonstration av förbättrad utformning av lösningen. Gränssnittet fick bla en bättre presentation av samtalshistorik, mer korrekt nummerpresentation. Dessa förbättringar bör göra denna lösning mer attraktivt för de personer som skall använda och underhålla lösningen.
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Hortling, Johan, Erik Bergh, and Daniel Karlsson. "Konstruktion och penetrationstestning av VoIP-system." Thesis, Högskolan i Halmstad, Sektionen för Informationsvetenskap, Data– och Elektroteknik (IDE), 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:hh:diva-19985.
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VoIP-system inom företag blir mer vanligt. Säkerheten bör då beaktas för att undvika hot som riskerar konfidentialitet, integritet och tillgänglighet. Denna rapport visar resultat från två olika VoIP-systems säkerhet med hjälp av praktiska penetrationstestscenarion i labbmiljö. En redogörelse över verktyg som är använda för säkerhetstesterna mot VoIP och tillvägagångssätt redovisas i rapporten med förklarande text och tabeller.<br>VoIP systems in enterprises are becoming more common. Security should then be followed to avoid threats against confidentiality, integrity and availability. This report shows the results from two different VoIP systems security using practical penetration test scenarios in a laboratory environment. A statement of tools that are used for safety tests on VoIP and methods for this, is presented in the report with explanatory text and tables.
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Shah, Zawar Electrical Engineering & Telecommunications Faculty of Engineering UNSW. "Location tracking architectures for wireless VoIP." Publisher:University of New South Wales. Electrical Engineering & Telecommunications, 2009. http://handle.unsw.edu.au/1959.4/43324.
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A research area that has recently gained great interest is the development of network architectures relating to the tracking of wireless VoIP devices. This is particularly so for architectures based on the popular Session Initiation Protocol (SIP). Previous work, however, in this area does not consider the impact of combined VoIP and tracking on the capacity and call set-up time of the architectures. Previous work also assumes that location information is always available from sources such as GPS, a scenario that rarely is found in practice. The inclusion of multiple positioning systems in tracking architectures has not been hitherto explored. It is the purpose of this thesis to design and test SIP-based architectures that address these key issues. Our first main contribution is the development of a tracking-only SIP based architecture. This architecture is designed for intermittent GPS availability, with wireless network tracking as the back-up positioning technology. Such a combined tracking system is more conducive with deployment in real-world environments. Our second main contribution is the development of SIP based tracking architectures that are specifically aimed at mobile wireless VoIP systems. A key aspect we investigate is the quantification of the capacity constraints imposed on VoIP-tracking architectures. We identify such capacity limits in terms of SIP call setup time and VoIP QoS metrics, and determine these limits through experimental measurement and theoretical analyses. Our third main contribution is the development of a novel SIP based location tracking architecture in which the VoIP application is modified. The key aspect of this architecture is the factor of two increase in capacity that it can accommodate relative to architectures utilizing standard VoIP. An important aspect of all our tracking architectures is the Tracking Server. This server supplies the location information in the event of GPS unavailability. A final contribution of this thesis is the development of novel particle-filter based tracking algorithms that specifically address the GPS intermittency issue. We show how these filters interact with other features of our SIP based architectures in a seamless fashion.
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Talaganov, Goce. "Green VoIP : A SIP Based Approach." Thesis, KTH, Kommunikationssystem, CoS, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-98795.
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This master thesis presents, examines, designs, implements, and evaluates with respect to energy efficiency a secure and robust VoIP system. This system utilizes a Session Initiation Protocol (SIP) infrastructure assisted by a cloud service, specifically focusing on small to medium sized enterprises (SME) and homes. This research focuses on using inexpensive, flexible, commodity embedded hardware (specifically a Linksys WRT54GL wireless router for the local site with a customized operating system, specifically DD-WRT). The idea is to reduce the local site's power consumption to very low levels by examining which functions can be done in a cloud service rather than at the local site. The thesis presents the design of a low-power IP telephony system for the local site and the cloud site. A number of different usage scenarios and desirable features are described. The methodology for conducting a set of experiments is defined to perform stress-testing and to evaluate the low- power IP telephony system's design. The experiments concern the overall power consumption of the local site under various configurations, the VPN link's call capacity, the QoS metrics for the VoIP calls, the session request delay (SRD) and the registration request delay (RRD). The results from these experiments show that there is a potential for significant power savings when using the proposed design for an IP telephony system.<br>Detta examensarbete presenterar, undersöker, utformar, implementerar, och försöker att utvärdera ett säkert och robust VoIP-system med energieffektivitet. Detta system använder en Session Initiation Protocol (SIP)-infrastruktur med hjälp av en molntjänst med särskild inriktning på, små, och medelstora företag (SME) och hemmanvändare. Denna forskning fokuserar att använda en prisvärt, billig, flexibel, med program inbyggd hårdvara (speciellt en Linksys WRT54GL trådlös router för den lokala platsen med ett anpassat operativsystem DD-WRT). Tanken är att minska energiförbrukningen på, den lokala platsen till mycket låga nivåer genom att undersöka vilka funktioner, som kan köras på, ett molntjnst snarare än på, den lokala platsen. Avhandlingen presenterar utformningen av ett IP-telefonisystem på, den lokala platsen med ett lågt strömbehov och på, molntjänsten. Ett antal olika användningsförhållanden och önskvärda egenskaper är beskrivna. Metodiken för att genomföra en rad experiment definieras för att utföra stresstester och för att utvärdera designen av IP-telefonisystem med ett lågt effektbehov. I försöken experimenteras den totala energiförbrukningen av den lokala platsen under olika konfigurationer, VPN-länkens samtalskapacitet, QoS-mätning för VoIP-samtal, Session Request Delay (SRD) och Registration Request Delay (RRD). Resultaten från dessa experiment visar att det finns en potential för betydande energibesparing när du använder den föreslagna designen för en IP-telefoni system.
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Scott, David T. "Extending tactical fleet communications through VoIP." Thesis, Monterey, California: Naval Postgraduate School, 2014. http://hdl.handle.net/10945/43996.
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Approved for public release; distribution is unlimited<br>TheNavy's Fleet is in need of tactical voice communication systems that are highly reliable, protected from cyber threats, and able to operate in a denied or degraded environment. Many of theNavy's current systems rely on outdated and inefficient technology, which reduces the overall effectiveness of our tactical communication channels and also limits the accessibility of these systems to communications challenged areas within ships. This research examines the capabilities, limitations, and overall performance of an integrated Voice over Internet Protocol (VoIP) system using four popular link layer protocols (i.e., Ethernet, 802.11n, 2.4 GHz 802.11ac, and 5 GHz 802.11ac) in an attempt to determine the feasibility of incorporating VoIP technology within Consolidated Afloat Networks and Enterprise Services and digital modular radio communication systems. The specific features compared in this study are VoIP network bandwidth consumption, overall network capacity between the four link layer protocols, VoIP codecs, VoIP call limits, intrusion detection system effects, and the effects of additional non-VoIP network traffic. The results of the study show that afloat tactical communications can be effectively enhanced by integrating VoIP intrusion detection systems monitored VoIP network applications with afloat communications systems, and by extending those systems with wireless devices utilizing the 802.11ac protocol.
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Akeel, Hosam. "Remotely Operated VoIP Radio for Drones." Thesis, KTH, Skolan för elektroteknik och datavetenskap (EECS), 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-290598.
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Nowadays, search and rescue organizations are considering using drones to aid in rescue missions. To ensure the drones work properly, we need to design a reliable and stable systems. One of these systems is the communication system, which creates a link between the drone and the operator. There are many communication technologies used in rescue missions, such as Voice over Internet Protocol (VoIP). VoIP uses the Internet to send and receive traditional telephony services such as voice, video, and other media sessions. In this project, we investigate the performance of Software Defined Radio (SDR) in implementing the VoIP technology on the drones to stream video by using Fourth Generation of Telecommunication Systems (4G) Long-Term Evolution (LTE). In addition, the SDR should also communicate in the sea using marine Very High Frequency (VHF) voice-radio for sea rescue missions. To verify and evaluate the system, we used BladeRF xA4 to run Qradiolink and srsLTE to simulate the communication system as the hardware platform. In this experiment, we use a BladeRF SDR as a base station. The test procedure is designed to evaluate the performance of the 4G LTE and marine VHF links. We conducted tests in three different environments to evaluate the performance of the system and show the impacts of different environments. In the 4G LTE measurements we evaluate the Signal to Noise Ratio (SNR), Reference Signal Received Power (RSRP), jitter, delay, and packet loss. In the marine VHF measurement we evaluate the Received Signal Strength Indicator (RSSI). The experimental result shows that it is possible to use SDR to implement the suggested communication system. However, there are some restrictions that are related to the power source and the BladeRF xA4 hardware design. Based on the experimental results, we create a scaling graph to show the baseline of the hardware specifications to achieve the desired performance. We also discussed in details the cost to implement this communication system. Finally, we conclude that it is cost-effective to use SDR as the communication system if there are no off-the-shelf products that cost 15000 SEK.<br>Nu for tiden overvager sok- och raddningsorganisationer att anvanda dronare iraddningsuppdrag. For att sakerstalla att dronarna fungerar ordentligt maste videsigna palitliga och stabila system. Ett av dessa system ar kommunikationssystemet,som skapar en lank mellan dronaren och operatoren. Det finns mangakommunikationstekniker som anvands i raddningsuppdrag, sasom Rost overInternet Protokoll (VoIP). VoIP anvander Internet for att skicka och ta emottraditionella telefonitjanster som rost-, video- och andra mediasessioner. I dethar projektet undersoker vi prestanda for Mjukvara Definierade Radio (SDR)vid implementering av VoIP-teknik pa dronare for att stromma video med FjardeGenerationen av Telekommunikation System ( 4G) Langsiktig Utveckling(LTE). Dessutom bor SDR ocksa kommunicera i havet med hjalp av marineMycket Hog Frekvens (VHF) rostradio for sjoraddningsuppdrag. For att verifieraoch utvardera systemet anvande vi BladeRF xA4 for att kora Qradiolinkoch srsLTE for att simulera kommunikationssystemet som hardvaruplattform.I detta experiment anvander vi en BladeRF SDR som basstation. Testforfarandetar utformat for att utvardera prestanda for 4G LTE- och marineVHFlankar.Vi genomforde tester i tre olika miljoer for att utvardera systemets prestandaoch visa effekterna av olika miljoer. I 4G LTE-matningarna utvarderarvi signal brusforhallande (SNR), referenssignal mottagen effekt (RSRP), jitter,fordrojning och paketforlust. I VHF-matningen utvarderar vi mottagen signalstyrkaindikator (RSSI).Det experimentella resultatet visar att det ar mojligt att anvanda SDR for attimplementera det foreslagna kommunikationssystemet. Det finns dock vissabegransningar som ar relaterade till stromkallan och BladeRF xA4-maskinvarudesignen.Baserat pa experimentresultaten skapar vi en skalningsdiagram for att visa baslinjenfor hardvaruspecifikationerna for att uppna onskad prestanda. Vi diskuteradeocksa i detaljkostnaderna for att implementera detta kommunikationssystem.Slutligen drar vi slutsatsen att det ar kostnadseffektivt att anvanda SDRsom kommunikationssystem om det inte finns nagra hylla produkter som kostar15 000 SEK.
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Сасага, С. І., та S. I. Sasaha. "Дослідження VoIP-трафіку в комп'ютерних мереж". Thesis, Тернопільський національний технічний університет імені Івана Пулюя, 2015. http://elartu.tntu.edu.ua/handle/lib/21399.
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Сасага С.І. ,,Дослідження VoIP-трафіку в комп'ютерних мереж”
На здобуття освітньо-кваліфікаційного рівня “Магістр” зі спеціальності 8.05010201 “Комп’ютерні системи та мережі”. Тернопільський національний технічний університет імені Івана Пулюя, Тернопіль 2015.
Метою роботи є дослідити VoIP-трафік в комп'ютерних мережах, провести аналіз та порівняльну характеристику для можливості оптимізації трафіку.
Методи дослідження. Для розв’язання поставлених задач у дипломній роботі використано методи метод прогнозування MMSE; метод статичного прогнозування за середнім значенню ряду.
Для проведення досліджень використовувались програма для аналізу мережевих пакетів та відкрита комунікаційна платформа, для розгортання програмних АТС Asterisk.
Результати роботи: були розглянуті питання дослідження сигнального трафіку протоколу сигналізації SIP. У роботі проведений статистичний аналіз трафіку протоколу SIP, що показав наявність в ньому всіх основних властивостей самоподібності, вибраний ефективний метод прогнозування сигнального навантаження SIP, а також розроблений новий механізм управління перевантаженнями в мережі сигналізації SIP, що враховує короткочасний прогноз трафіку.
КЛЮЧОВІ СЛОВА:VoIP, моделі прогнозування, само подібний трафік, протокол SIP, трафік, MMSE, перенавантаження, прогнозування.<br>S.I. Sasaha: "Research VoIP traffic in computer networks "
This is submitted for the Master Degree in specialism 8.05010201 – Computer Networks and Systems. - Ternopil Ivan Pul’uj National Technical University, Ternopil, 2015.
The aim is to explore the VoIP-traffic in computer networks, conduct analysis and comparative characteristics for opportunities to optimize traffic.
Research methods. To solve the tasks of the thesis work is used methods of forecasting method MMSE; static method of predicting the average value of the series.
For research program used to analyze network packets and open communications platform to deploy software PBX Asterisk.
The results: the research question addressed signaling traffic signaling protocol SIP. The work carried out statistical analysis of traffic protocol SIP, which showed the presence of a self-similarity of the basic properties of selected effective method for predicting signal load SIP, and also developed a new mechanism for managing network congestion signaling SIP, taking into account the short-term traffic forecast.
KEYWORDS: VoIP, forecasting models, just like the traffic protocol SIP, traffic, MMSE, overload, prediction.
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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 Telemática<br>Nas 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.<br>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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Nassar, Mohamed. "VoIP Networks Monitoring and Intrusion Detection." Thesis, Nancy 1, 2009. http://www.theses.fr/2009NAN10021/document.
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La Voix sur IP (VoIP) est devenue un paradigme majeur pour fournir des services de télécommunications flexibles tout en réduisant les coûts opérationnels. Le déploiement à large échelle de la VoIP est soutenu par l'accès haut débit à l'Internet et par la standardisation des protocoles dédiés. Cependant, la VoIP doit également faire face à plusieurs risques comprenant des vulnérabilités héritées de la couche IP auxquelles s'ajoutent des vulnérabilités spécifiques. Notre objectif est de concevoir, implanter et valider de nouveaux modèles et architectures pour assurer une défense préventive, permettre le monitorage et la détection d'intrusion dans les réseaux VoIP. Notre travail combine deux domaines: celui de la sécurité des réseaux et celui de l'intelligence artificielle. Nous renforçons les mécanismes de sécurité existants en apportant des contributions sur trois axes : Une approche basée sur des mécanismes d'apprentissage pour le monitorage de trafic de signalisation VoIP, un pot de miel spécifique, et un modèle de corrélation des événements pour la détection d'intrusion. Pour l'évaluation de nos solutions, nous avons développés des agents VoIP distribués et gérés par une entité centrale. Nous avons développé un outil d'analyse des traces réseaux de la signalisation que nous avons utilisé pour expérimenter avec des traces de monde réel. Enfin, nous avons implanté un prototype de détection d'intrusion basé sur des règles de corrélation des événements<br>Voice over IP (VoIP) has become a major paradigm for providing flexible telecommunication services and reducing operational costs. The large-scale deployment of VoIP has been leveraged by the high-speed broadband access to the Internet and the standardization of dedicated protocols. However, VoIP faces multiple security issues including vulnerabilities inherited from the IP layer as well as specific ones. Our objective is to design, implement and validate new models and architectures for performing proactive defense, monitoring and intrusion detection in VoIP networks. Our work combines two domains: network security and artificial intelligence. We reinforce existent security mechanisms by working on three axes: a machine learning approach for VoIP signaling traffic monitoring, a VoIP specific honeypot and a security event correlation model for intrusion detection. In order to experiment our solutions, we have developed VoIP agents which are distributed and managed by a central entity. We have developed an analyzer of signaling network traces and we used it to analyze real-world traces. Finally, we have implemented a prototype of a rule-based event-driven intrusion detection system
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Šolc, Jiří. "Bezpečnost firemních telefonních sítí využívajících VoIP." Master's thesis, Vysoká škola ekonomická v Praze, 2008. http://www.nusl.cz/ntk/nusl-8182.
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This thesis focuses on enterprise VoIP telephony network security. Introduces brief comparison of old analog and digital voice networks and IP telephone networks with special focus on VoIP system security. The goal of the thesis is to identify the risks of implementation and operation of VoIP technologies in enterprise environment and so thesis brings some conclusion how to minimalize or avoid these risks. First two chapters briefly introduce the development of telephony technologies with differentiation of enterprise telephone network from public telephone networks. Further it describes individual technologies, digitalization of voice, processing the signal and VoIP protocols and components. Third chapter focuses on infrastructure of telephony networks with special interest for architecture of IP telephony and ways of establishing call processing. It describes data flows for further security risk analysis, which this technology came with. Fifth chapter is about enterprise security standards in common and is trying to describe information security management system (ISMS) adopting VoIP technology. Individual security threats and risks are described in sixth chapter, along with known methods how to avoid them. Final parts of thesis concludes of two real situation studies of threats and risks of VoIP technologies implemented in environment of small commercial enterprise and medium size enterprise, in this example represented by University of economics. These chapters conclude theoretical problems shown on practical examples.
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Liu, Zuo. "Supporting VoIP in IEEE802.11 distributed WLANs." Thesis, University of Manchester, 2013. https://www.research.manchester.ac.uk/portal/en/theses/supporting-voip-in-ieee80211-distributed-wlans(1a6225c3-770e-4ce1-8fbb-b1e3f05534d2).html.
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Telecommunications is converging on the use of IP based networks. Due to the low cost of VoIP applications, they are being increasingly used instead of conventional telephony services. IEEE802.11 WLANs are already widely used both commercially and domestically. VoIP applications will also expand from usage over wired networks to voice communications over IEEE802.11 WLANs. This is known as VoWLAN. The use of VoWLAN may reach the maximum capacity of a wireless channel if there are many simultaneous VoIP calls operating close to each other. There is much published research based on a single IEEE802.11 infrastructure WLAN concluding that packet loss, transmission efficiency and latency issues are the major challenges limiting the VoWLAN capacity. The VoIP service quality will drop sharply when the demands exceed the WLAN’s capacity. This thesis demonstrates that these challenges also apply to distributed WLANs. To extend these findings from the existing research, the analysis in this thesis indicates that the capacity of a single IEEE802.11 WLAN channel is 12 VoIP calls. When the number of simultaneous VoIP calls is within the capacity, the WLAN can deliver more than 90% of the voice packets to the receiver within 150 ms (the lowest network performance for supporting acceptable VoIP service). However, as soon as the traffic loads are beyond the wireless channel capacity e.g. the number of simultaneous VoIP calls is greater than 13, the VoIP service quality catastrophically collapses. When the capacity is exceeded there are almost no voice packets that can be delivered to the receiver within 150 ms. Our research results indicate that the delay accumulation for voice packets in the transmitter’s outgoing buffer causes this problem. Our research also found that dropping ‘stale’ voice packets that are already late for delivery to the receiver can give more transmission opportunities to those voice packets that may still be delivered in time. This thesis presents a new strategy called Active Cleaning Queue (ACQ) which actively drops ‘stale’ voice packets from the outgoing buffer and prevents the accumulation of delay in congested conditions. When ACQ is applied in a saturated wireless channel the network performance for supporting VoIP traffic was found to gradually decrease proportional to the numbers of simultaneous VoIP calls rather than catastrophically collapse. There is also published research suggesting that the aggregation of packets can improve the efficiency of WLAN transmissions. An algorithm called Small Packet Aggregation for Wireless Networks (SPAWN) is also presented in this thesis to improve transmission efficiency of small voice packets in WLANs without introducing further delay to VoIP traffic. The evaluation result shows that after applying the SPAWN algorithm, the VoIP capacity of a single wireless channel can be extended up to 24 simultaneous calls.
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Nassar, Mohamed Festor Olivier. "VoIP Networks Monitoring and Intrusion Detection." S. l. : Nancy 1, 2009. http://www.scd.uhp-nancy.fr/docnum/SCD_T_2009_0021_NASSAR1.pdf.
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Calma, Giampiero. "Un'applicazione VoIP per Symbian: un'interfaccia utente." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/1940/.
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Nel campo della tecnologia, l’ultimo decennio è stato caratterizzato da significativi sviluppi nel mondo dei dispositivi mobili. Si è passati dal tradizionale telefono cellulare, ai più recenti palmari e Smartphone che integrano al tradizionale stereotipo di telefono cellulare, funzionalità avanzate su hardware molto sofisticato.
Con un moderno dispositivo mobile infatti, è possibile collegarsi ad Internet, leggere mail, guardare video, scaricare applicazioni e installarle per poterne così fruire.
L’International Telecommunications Union (ITU-T) ha stimato che alla fine del 2010 il numero di utenti Internet a livello mondiale ha raggiunto i 2 mi- liardi e che gli accessi alla rete cellulare hanno raggiunto circa 5,3 miliardi di sottoscrizioni. Se si considera inoltre che le reti 2G verranno sostituite da quelle 3G (che consente una connessione alla rete a banda larga tramite dispositivi cellulari), è inevitabile che nel prossimo futuro, gli utilizzatori di Internet tramite rete mobile potranno arrivare ad essere anche qualche miliardo.
Le applicazioni disponibili in rete sono spesso scritte in linguaggio Java che su dispositivi embedded, dove è cruciale il consumo di energia, mettono in crisi la durata della batteria del dispositivo. Altre applicazioni scritte in linguaggi meno dispendiosi in termini di consumi energetici, hanno un’interfaccia scarna, a volte addirittura ridotta a semplice terminale testuale, che non è indicata per utenti poco esperti. Infine altre applicazioni sono state eseguite solo su simulatori o emulatori, perciò non forniscono riscontri su dispositivi reali.
In questa tesi verrà mostrato come su un dispositivo mobile sia possibile utilizzare, tramite un’interfaccia “user-friendly”, una tecnologia già esistente e diffusa come il VoIP in maniera tale che qualunque tipologia di utente possa utilizzarla senza conoscerne i dettagli tecnici.
Tale applicazione, dovendo utilizzare una connessione dati, sfrutterà o una connessione a una rete WLAN o una connessione a una rete cellulare (GPRS, UMTS e HSDPA ad esempio) a seconda della dotazione hardware dell’apparecchio mobile e della locazione dello stesso in una rete accessibile dall’utente.
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Maiani, Lorenzo. "VoIP su Symbian: Sicurezza e Multihoming." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/1893/.
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Holubovský, Petr. "Management výkonnosti a optimalizace VoIP technologie." Master's thesis, Česká zemědělská univerzita v Praze, 2016. http://www.nusl.cz/ntk/nusl-259801.
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The diploma thesis focuses on the VoIP technology optimization and performance management. The diploma thesis presents the theoretical basis of IP telephony and measurement of its quality. The thesis primarily deals with practical measurements of VoIP calls quality. Asterisk softswitch, various types of IP phones and simulated degradation of signal using Linux software router are used for measurements. Procedural diagram of VoIP technology real deployment is designed based on these measurements.
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Binder, Tomáš. "Správa a konfigurace VoIP ústředny Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217252.
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This diploma dissertation is dealing with the VoIP software exchange Asterisk. In the dissertation there are described its abilities and possible ways of its configuration. Special attention is given to the signalling protocol SIP, which is described in one of the chapters. Within this dissertation a dial plan, which demonstrates the technique of dial plan creating, was created. Within the boundaries of the dialplan following services could be found: a voicemail, conference, Interactive Voice Response and call queues. Configuration files, with the help of which the exchange is configurated, are described in my dissertation as well. Finally, three laboratory assignments for purposes of the subject Multimedia Services are mentioned. Their main aim is to familiarise students with the creation of SIP accounts in the exchange, their mutual connections, defining the Interactive Voice Response and forming a new call centre.
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Mrník, Martin. "Lokace volajícího při tísňových hovorech VoIP." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-220295.
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This master thesis deals with the problem of localization of the caller to the emergency line with focus on VoIP calls. It contains general description of the main active and passive geolocation methods, and explains their function. The paper analyzes market share of different VoIP clients and describes the program Skype. The focus of this work is to create an application for geolocation of the caller using analysis of VoIP transmission. The application gains IP address of the caller by capturing packets and displays his/her location on the map. The coordinates are obtained from the MaxMind GeoLite City database which been chosen by evaluating objective parameters as the most suitable for desired use.
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Ulický, Ivan. "Zabezpečení VoIP sítí a jejich testování." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-220321.
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Main goal of creating this diploma thesis is existence of increasingly amount of potential threats against IP voice networks (VoIP). The thesis is devoted to testing of various types of attacks and provides some possible solutions for this systems as well. The work points out to a various types of current attacks against either insecure or very little secure structures. The theoretical part is dedicated to analyse and description of wide spectrum of VoIP protocols including signaling protocols (SIP, IAX2), transport protocols (RTP, RTCP) and security protocols (SRTP, ZRTP, IPsec, SDES). Further attention is dedicated to the one of possible open source IP PBX solutions called Asterisk. There is shown a variety of possible attacks against this system due to its openness, because open systems always tend to be more susceptible for various attacks as they need an advanced administration and endless need for searching of new trends in area of security. The last block of the theoretical part is focused on common threats and types of attacks against VoIP networks. The practical part is about design and creation of web application called ,,VoIP Hacks using PHP” written in PHP scripting language and ist main task is to execute three basic attacks: eavesdropping, call drop and call flood. There is also a possibility of port scanning of selected network which is added as supplementary part of this application. The application can be comfortably managed from web browser user interface. All captured data can be displayed directly into the web browser. Tests of the application were performed on Google Chrome and Mozzila Firefox browsers. There is an accent placed on cooperation between the application and terminal linux programmes such as Tshark, BYE Teardown, INVITE flooder or Nmap, which all accept commands from web interface and interpret gained output values back to the web browser.
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Polášek, Jakub. "Návrh a realizace testeru VoIP protokolů." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2016. http://www.nusl.cz/ntk/nusl-242144.
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This paper deals with ways and means of testing SIP or IAX2 based equipment and protocols itselfs. I analyse terminology and methodology for benchmarking of proxy and registrar servers as it was described in documents RFC 7501 and RFC 7502 from organization IETF. In the practical part is described the tester realization programmed in PHP programming language witch will use described methodology. Aplication is available in web based interface.
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Merell, Robin. "Implementation av in-house applikation med möjligheter tillIP-telefoni." Thesis, Linköpings universitet, Institutionen för datavetenskap, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-93318.
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Detta examensarbete utfördes hos företaget Syntronic i mjärdevi. Uppgiften de ville ha utförd var en undersökning och implementation av ny funktionalitet i en redan existerande android applikation som de har utvecklat. Uppgiften i denna applikation var att undersöka möjligheterna till IP-telefoni, eller VOIP som det kallas, så att de kunde ringa varandra via applikationen genom datanätet istället för det ordinära nätet, samt att implementera detta. Detta examensarbete utfördes hos företaget Syntronic i mjärdevi. Uppgiften de ville ha utförd var en undersökning och implementation av ny funktionalitet i en redan existerande android applikation som de har utvecklat. Uppgiften i denna applikation var att undersöka möjligheterna till IP-telefoni, eller VOIP som det kallas, så att de kunde ringa varandra via applikationen genom datanätet istället för det ordinära nätet, samt att implementera detta. För att möjliggöra detta har dels android API för SIP, samt en Asterisk server som tar hand om SIP samtalen och SIP-adresserna används. Utöver detta finns det en SOA-tjänst som tar hand om databashanteringen för applikationen.
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Antunes, Rothschild Alencastro [UNESP]. "Instalação de uma rede mesh metropolitana utilizando o padrão IEEE 802.11a e implementação do serviço Voip (Wman-Voip)." Universidade Estadual Paulista (UNESP), 2012. http://hdl.handle.net/11449/87068.
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antunes_ra_me_ilha.pdf: 1207394 bytes, checksum: b3386fc3860a3bec889eb34ebdb2afb9 (MD5)<br>PROPG - Programa de Pós-Graduação<br>A utilização da tecnologia de redes sem fio tem ganhado espaço em diversas aplicações na área de Engenharia nos últimos anos, entre os mais recentes encontra-se o uso de rede sem fio em conjunto com as redes smartgrid e serviço de tempo real. Este trabalho tem o objetivo de apresentar a implementação de uma rede mesh metropolitana e as características da primeira rede mesh (malha) sem fio no estado de Mato Grosso aplicada a pesquisa. Essa rede é resultado de uma parceria com o grupo de pesquisa de redes sem fio (GPRS) do Instituto Federal de Educação, Ciência e Tecnologia de Mato Grosso (IFMT), onde foi feito um estudo de caso na rede, denominada Stormesh. A rede abrange as proximidades do campus Cuiabá do IFMT e tem o propósito de servir de objeto de pesquisa e de acesso à Internet para alunos e servidores do campus Cuiabá e também fornecer outros serviços de rede, como por exemplo, serviço de VoIP (Voice over Internet Protocol). Este trabalho detalha os desafios essenciais superados na implementação de uma rede mesh em área metropolitana e demonstra formas otimizadas de instalar e configurar antenas nessas redes. Além disso, esclarece alguns aspectos importantes na montagem e configuração que normalmente não são encontrados na literatura para esse tipo de implementação. Uma forma de avaliação de performance da Stormesh foi feita avaliação do nível de sinal e vazão da rede, e a implementação de VoIP. Embora o serviço de VoIP esteja bem difundido em escala mundial, quando implementado em redes sem fio, podem ocorrer perdas de dados significativas comprometendo a qualidade da voz transmitida via VoIP. Dentre os vários mecanismos utilizados em conexão VoIP para lidar com os dados corrompidos, afim de assegurar qualidade aceitável ao tráfego de voz, destacam-se os codecs que são...<br>The use of wireless networking technology has gained importance in various applications in the field of engineering in recent years, among the most recent is the use of wireless networks in conjunction with SmartGrid and real-time service. This work aims to present the implementation of a metropolitan mesh area network and the characteristics of the first mesh network wireless in the state of Mato Grosso applied research. This network is the result of a partnership with the research group of wireless networks of the Instituto Federal de Educação, Ciências e Tecnologia de Mato Grosso (IFMT), which it was done a case study on the network, called Stormesh. The network covers around Cuiabá campus of the IFMT and it have purpose serving as a research object and internet access to students and employees on Cuiabá campus and also provide other network services, such as service Voice over Internet Protocol (VoIP). This work details the key challenges overcome in implementing a mesh network in the metropolitan area and demonstrates ways to install and configure optimized antennas on those networks. In addition, it clarifies some important aspects in the assembly and configuration that are not normally found in the literature for this type of implementation. One way of assessing the performance Stormesh was the implementation of service Voice over IP, where we was done the evaluated signal level and flow of the network. Although VoIP service is well spread worldwide, when deployed in wireless networks, data loss can occur significantly affecting the quality of voice transmitted via VoIP. Among the various mechanisms used in VoIP connection to handle corrupted data in order ensure acceptable quality of voice traffic, we highlight the codecs that are used in the encoding and decoding of signals transmitted voice. Therefore, we... (Complete abstract click electronic access below)
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Antunes, Rothschild Alencastro. "Instalação de uma rede mesh metropolitana utilizando o padrão IEEE 802.11a e implementação do serviço Voip (Wman-Voip) /." Ilha Solteira, 2012. http://hdl.handle.net/11449/87068.
Full textAbstract:
Orientador: Ailton Akira Shinoda<br>Coorientador: Ruy de Oliveira<br>Banca: Sérgio Azevedo de Oliveira<br>Banca: Tony Inácio da Silva<br>Resumo: A utilização da tecnologia de redes sem fio tem ganhado espaço em diversas aplicações na área de Engenharia nos últimos anos, entre os mais recentes encontra-se o uso de rede sem fio em conjunto com as redes smartgrid e serviço de tempo real. Este trabalho tem o objetivo de apresentar a implementação de uma rede mesh metropolitana e as características da primeira rede mesh (malha) sem fio no estado de Mato Grosso aplicada a pesquisa. Essa rede é resultado de uma parceria com o grupo de pesquisa de redes sem fio (GPRS) do Instituto Federal de Educação, Ciência e Tecnologia de Mato Grosso (IFMT), onde foi feito um estudo de caso na rede, denominada Stormesh. A rede abrange as proximidades do campus Cuiabá do IFMT e tem o propósito de servir de objeto de pesquisa e de acesso à Internet para alunos e servidores do campus Cuiabá e também fornecer outros serviços de rede, como por exemplo, serviço de VoIP (Voice over Internet Protocol). Este trabalho detalha os desafios essenciais superados na implementação de uma rede mesh em área metropolitana e demonstra formas otimizadas de instalar e configurar antenas nessas redes. Além disso, esclarece alguns aspectos importantes na montagem e configuração que normalmente não são encontrados na literatura para esse tipo de implementação. Uma forma de avaliação de performance da Stormesh foi feita avaliação do nível de sinal e vazão da rede, e a implementação de VoIP. Embora o serviço de VoIP esteja bem difundido em escala mundial, quando implementado em redes sem fio, podem ocorrer perdas de dados significativas comprometendo a qualidade da voz transmitida via VoIP. Dentre os vários mecanismos utilizados em conexão VoIP para lidar com os dados corrompidos, afim de assegurar qualidade aceitável ao tráfego de voz, destacam-se os codecs que são... (Resumo completo, clicar acesso eletrônico abaixo)<br>Abstract: The use of wireless networking technology has gained importance in various applications in the field of engineering in recent years, among the most recent is the use of wireless networks in conjunction with SmartGrid and real-time service. This work aims to present the implementation of a metropolitan mesh area network and the characteristics of the first mesh network wireless in the state of Mato Grosso applied research. This network is the result of a partnership with the research group of wireless networks of the Instituto Federal de Educação, Ciências e Tecnologia de Mato Grosso (IFMT), which it was done a case study on the network, called Stormesh. The network covers around Cuiabá campus of the IFMT and it have purpose serving as a research object and internet access to students and employees on Cuiabá campus and also provide other network services, such as service Voice over Internet Protocol (VoIP). This work details the key challenges overcome in implementing a mesh network in the metropolitan area and demonstrates ways to install and configure optimized antennas on those networks. In addition, it clarifies some important aspects in the assembly and configuration that are not normally found in the literature for this type of implementation. One way of assessing the performance Stormesh was the implementation of service Voice over IP, where we was done the evaluated signal level and flow of the network. Although VoIP service is well spread worldwide, when deployed in wireless networks, data loss can occur significantly affecting the quality of voice transmitted via VoIP. Among the various mechanisms used in VoIP connection to handle corrupted data in order ensure acceptable quality of voice traffic, we highlight the codecs that are used in the encoding and decoding of signals transmitted voice. Therefore, we... (Complete abstract click electronic access below)<br>Mestre
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Nakarmi, Prajwol Kumar. "Evaluation of VoIP Security for Mobile Devices." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-43721.
Full textAbstract:
Market research reports by In-Stat, Gartner, and the Swedish Post and Telecom Agency (PTS) reveal a growing worldwide demand for Voice over IP (VoIP) and smartphones. This trend is expected to continue over the coming years and there is wide scope for mobile VoIP solutions. Nevertheless, with this growth in VoIP adoption come challenges related with quality of service and security. Most consumer VoIP solution, even in PCs, analog telephony adapters, and home gateways, do not yet support media encryption and other forms of security. VoIP applications based on mobile platforms are even further behind in adopting media security due to a (mis-)perception of more limited resources. This thesis explores the alternatives and feasibility of achieving VoIP security for mobile devices in the realm of the IP Multimedia Subsystem (IMS).
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Salari, Syed Ghazanfar. "Control traffic overhead for VoIP over LTE." Thesis, KTH, Kommunikationssystem, CoS, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-99045.
Full textAbstract:
With increasing technological advancements more sophisticated mobile devices are being used by end-users. Third generation (3G) mobile communication systems such as Universal Mobile Telecommunication System (UMTS) are not able to satisfy the rising demand for higher throughputs and low latencies. New standards based on Orthogonal Frequency Division Multiplexing (OFDM), such as Long Term Evolution (LTE) and Worldwide Interoperability for Microwave Access (WiMAX), have been proposed and are currently being integrated into existing mobile networks all over the world. LTE specifications are being finalized within the 3rd Generation Partnership Project (3GPP) with the ambitious goals of increased spectral efficiency and end user throughput. Despite the introduction of several high data rate services, voice communication is still an essential part of the overall wireless wide area cellular communication market. In LTE, the core network is purely packet switched, thus voice is transmitted entirely using a Voice over Internet Protocol (VoIP). Like its predecessor standards it is desired that a large number of simultaneous VoIP calls be supported in LTE, while satisfying the desired Quality of Service (QoS) demands. This thesis examines issues related to VoIP capacity for LTE. One of the key challenges is the limited number of schedulable voice packets per sub frame. The main goal of this thesis is to quantify the impact of this limitation. After describing basic LTE concepts, a detailed description of the control channel resource limitations for the scheduling of voice packets is presented. Consequences of these limitations are explained systematically by presenting the problem in a wider context. Simulation results were obtained using the openWNS Simulator, an event driven system level simulation platform developed at the Communication Networks Research Group (ComNets), RWTH Aachen University Germany. Results are presented showing the impact of different scheduling strategies on VoIP capacity. These results illustrate how the limited control channel resources, specifically the Physical Downlink Control Channel (PDCCH) resources, affect the total number of schedulable VoIP user audio media streams.<br>Med ökande tekniska framsteg mer avancerade mobila enheter som används av slutanv ändarna. Tredje generationens (3G) mobila kommunikationssystem såsom Universal Mobile Telecommunication System (UMTS) inte kan tillgodose den ökande efterfrågan på högre genomströmning och låga latenser. Nya standarder som bygger på Orthogonal Frequency Division Multiplexing (OFDM), såsom Long Term Evolution (LTE) och Worldwide Interoperability för Microwave Access (WiMAX), har föreslagits och håller på att integreras I befintliga mobilnät över hela världen. LTE specifikationer håller på att färdigställas inom 3rd Generation Partnership Project (3GPP) med de ambitiösa målen om ökad spektral effektivitet och slutanvändare genomstr ömning. Trots införandet av flera tjänster av hög datahastighet, är röstkommunikation fortfarande en väsentlig del av den totala Wireless Wide Area cellulär kommunikation marknaden. I LTE är kärnnätet rent paketförmedlande därmed röst överförs helt och hållet med hjälp av en Voice over Internet Protocol (VoIP). Precis som sina föregångare standarder är det önskvärt att ett stort antal samtidiga VoIP samtal få stöd i LTE, samtidigt som det uppfyller önskade Quality of Service (QoS) krav. Denna avhandling undersöker frågor relaterade till VoIP kapacitet för LTE. En av de viktigaste utmaningarna är det begränsade antalet schemaläggningsbart röst paket per sub ram. Det huvudsakliga målet med denna avhandling är att kvantifiera effekterna av denna begränsning. Efter att ha beskrivit de grundläggande LTE begrepp, är en detaljerad beskrivning av de resurser kontroll kanal begränsningar för schemaläggning av röst paket presenteras. Konsekvenser av dessa begränsningar förklaras systematiskt genom att presentera problemet i ett större sammanhang. Simulering resultat erhölls med hjälp av openWNS Simulator, en händelse driven systemnivå simulering som utvecklats vid Communication Networks Research Group (ComNets), RWTH Aachen University Tyskland. Resultat presenteras som visar effekterna av olika schemaläggning strategier för VoIP kapacitet. Dessa resultat illustrerar hur de begränsade kontroll kanalresurser, särskilt fysiskt Downlink (PDCCH) resurser, påverkar det totala antalet schemaläggningsbart VoIP användare ljud mediaströmmar.