Context-awareness is a hot topic in mobile computing currently. A lot of importance is being attached to facilitating the user of various mobile computing devices to provide services that are more “user-centric”. One aspect of context-awareness is to perceive variations in available resources, and to make decisions based on the feedback to enable applications to automatically adapt to the current environment. For Voice over IP (VoIP) software phones (softphones), variations in network performance lead to fluctuations in the quality of the communication. Therefore, by making these softphones more adaptive to the network environment will, to some extent, mask such fluctuations. Dynamic voice and video adaptation derives from the fact that different coder-decoders (CODEC) have different characteristics, even the same CODECs with a different configuration can behave quite differently, in terms of bandwidth consumption, packet size, etc. Minisip is a VoIP client application which was implemented on and targeted for a Linux platform. One of my tasks was to port Minisip to Microsoft’s Windows Mobile operating system, running on an HP IPAQ Pocket PC H5550. Such handheld computer enables the user to communication while they are moving about, thus increasing the probability that the characteristics of the network connection will change. Building upon this port, the next task was to add dynamic voice and video CODEC adaptation. Dynamic voice and video CODEC adaptation on Minisip poses several challenges, for example, in what way can the network performance be determined and what adaptation strategy can achieve high call quality while making efficient utilization of available network resources. In order to make the proper design choices, several estimation models will be discussed, these are used to determine an efficient, un-intrusive, and light weight means of dynamic CODEC selection within Minisip. This thesis only implemented audio CODEC adaptation of Minisip, and the evaluation of the resulting prototype shows that such dynamic adaptation is both feasible and practical; further more, video CODEC adaptation would be a more significant extension to this work in the future. / Context-awareness är ett hett i den nuvarande mobila datavärlden. Det finns ett stort värde i att facilitating användare av olika mobila dator anordningar för att kunna förse branschen med användarvänligare tjänster. En aspekt på Context-awareness är att uppmärksamma variationen i de tillgängliga medel som finns tillhanda, och att ta beslut som är baserade på feedback för att applikationen automatiskt ska anpassa sig till den nuvarande miljön. Variationer i nätverksprestanda påverkar kvaliteten på Voice over IP (VoIP), som är en typ utav softwaretelefon, i hög grad. Dessa kvalitets svängningar kan stabiliseras och döljas i högre grad om softwaretelefonen anpassas till nätverksmiljön. Dynamisk voice och video adaptation härleds från faktum att olika coder-decoders (CODEC) har olika karaktärer, även samma CODEC med en annan konfiguration kan bete sig olikt sig själv om vi talar om bandbredds förbrukning och packet storlekar, etc. Minisip är en VoIP klient som är framtagen för Linux plattformen. En av mina huvuduppgifter var att port Minisip till Microsoft’s Windows Mobila operativsystem genom att köra en HP IPAQ Pocket PC H5550. En sådan bärbar dator möjliggör för användaren att kommunicera fastän denne rör på sig, fastän risken finns för att nätverks kontakten ändras. Baserat på denna port, blev min nästa uppgift att anpassa denna CODEC till dynamiskt ljud och bild. Att anpassa denna CODEC till dynamiskt ljud och bild på Minisip medför många utmaningar t.ex. hur nätverks prestandan kan bestämmas och vilken anpassningsstrategi som kan bidra till högkvalitativa samtal samtidigt som nätverks tillgångarna nyttjas på ett effektivt sätt. Denna tes kan endast genomföras på ljud CODEC anpassning av Minisip, och utvärderingen utav prototypen resulterade i att sådan dynamisk anpassning är både genomförbar och praktisk, en video CODEC anpassning skulle bli ett perfekt uppföljningsprojekt till denna studie.

The Voice over IP (VoIP) technology has become prevalent today due to its lower cost than traditional telephony and its ability to support new value-added services. Additionally, the increasing availability of wireless internet access has led to research studies examining the combination of wireless network access with voice over IP. With the widespread availability of advanced mobile phones and Pocket PCs, the need for VoIP applications on these mobile platforms is tangible. To enable this, we need to evaluate the current wireless access technologies to see if they can support the necessary traffic and implement software to offer these VoIP services to users. In order to easily implement an IP-based service on GSM technology, we should use the GPRS service provided by the GSM operators. In this thesis, we evaluate Voice over IP service over GPRS in terms of feasibility and quality. Following this we ported a locally developed VoIP program to a Pocket PC (with GSM SIM-card support) which runs Microsoft’s Windows Mobile in order to provide suitable software as needed to offer the service from such a portable device. / VoIP tekniken har blivit en rådande teknik numera på grund av dess lägre kostnader och mervärdestjänster som erbjuds jämfört med traditional telefoni. Samtidigt som tendensen mot mer tillgänglig trådlöst internet har underlättat och därmed driver mera studier inom dessa områden. Den allt mer utbredda användningen av avancerade mobiltelefoner och handdatorer numera har lett till ökat behov av att använda VoIP tekniken för dessa mobila utrustningar är alltmer kännbar. För att möjliggöra användadet av VoIP tekniken så behöver vi först och främst utvärdera dagens existerande teknologier för att stödja iden och för det andra måste vi kunna implementera en mjukvara vilket kan erbjuda olika typer av tjänster för slutanvändaren. För att kunna använda en IP-baserad tjänst på GSM teknologin så måste vi använda oss utan GPRS tjänster som tillhandahålls av GSM opratörer. I detta examens arbete kommer vi att utvärdera VoIP tjänster på GPRS när det gäller kvalitet och möjligheter. Därefter kommer vi att Portning en VoIP mjukvara till en handdator (utrustad med GSM sim-kort) vilket har windows Mobile operativsystemet som erbjuder en rad olika tjänster.

Master's Thesis is engaged on possibilities connect SIP server to telephone network on FIT. The main reason is that employees can call to university, when they are out of faculty. For resolution we will use SIP server Asterisk, which will be serving as authorization server for users. Next Asterisk will ensure transmission numbers to SIP address with ENUM. In the practical part we will verify the functionality.

Voz sobre IP (ou VoIP) vem sendo adotada progressivamente não apenas por um grande número de empresas mas também por um número expressivo de pessoas, no Brasil e em outros países. Entretanto, essa crescente adoção de VoIP no mundo traz consigo algumas preocupações tais como ameaças e riscos de segurança, sobretudo no que diz respeito à autenticidade, privacidade e integridade da comunicação. Para proteger a sessão de mídia existem protocolos muito eficientes, como o Secure Real-time Transport Protocol (SRTP). Mas ele depende de uma chave secreta para tornar a comunicação segura de fato. Assim, uma boa estratégia é aproveitar o processo de sinalização que estabelece a sessão de mídia e negociar uma chave secreta de sessão que seja comum às partes comunicantes. Esse processo de sinalização é realizado por tipos específicos de protocolo tais como o Session Initiation Protocol (SIP), um protocolo de sinalização muito importante e que vem sendo usado cada vez mais por softphones para comunicação na Internet. Todavia, os riscos e ameaças mencionados já existem no próprio processo de sinalização e, dentre eles, o ataque do tipo man-in-the-middle é o mais perigoso, devido aos prejuízos que ele pode causar. Depois de fazer uma revisão bibliográfica dos riscos e ameaças inerentes ao SIP, bem como de seus mecanismos de segurança (analisando os pontos fortes e de atenção deles), foi possível originar um novo mecanismo de segurança, o qual é apresentado neste trabalho. O mecanismo proposto usa um protocolo para troca segura de informações o protocolo Massey-Omura o qual, quando combinado com emparelhamentos bilineares, provê ao SIP um melhor nível de segurança em todos os aspectos (autenticidade, privacidade e integridade). Além disso, o novo mecanismo é avaliado através de uma prova de conceito, na qual utilizou-se um softphone SIP funcional. A análise de segurança realizada e os resultados obtidos da prova de conceito fazem do mecanismo de segurança proposto uma alternativa viável para o SIP. / Voice over IP (or VoIP) has been progressively adopted not only by a great number of companies but also by an expressive number of people, in Brazil and in other countries. However, this increasing adoption of VoIP in the world brings some concerns such as security risks and threats, mainly on the authenticity, privacy and integrity of the communication. In order to protect the media session, efficient protocols like the Secure Real-time Transport Protocol (SRTP) have been used. However, it depends on a secret key to make the communication secure. Thus, a good strategy is to take advantage of the signaling process to establish the media session, and agree on a common secret session key between the communicating parties. This signaling process is performed by specific types of protocols such as the Session Initiation Protocol (SIP), a very important signaling protocol, which has been used more and more by softphones in the Internet communication. Nevertheless, those risks and threats already exist in the own signaling process and, among them, the man-in-the-middle attack is the worst of all due to its high danger degree. After doing a bibliographical revision of the SIP security risks and threats, as well as its security mechanisms (analyzing their advantages and drawbacks), it was possible to generate a new security mechanism, which is presented in this work. The proposed mechanism uses a protocol for secure information exchange the Massey-Omura protocol which, when combined with bilinear pairings, provides a better security level for SIP in all its aspects (authenticity, privacy and integrity). Besides this, the new mechanism is evaluated by a proof of concept, in the which a functional SIP softphone was used. The security analysis and the results obtained from the proof of concept, make the proposed security mechanism a viable alternative for SIP.

FundaÃÃo Cearense de Apoio ao Desenvolvimento Cientifico e TecnolÃgico / Nos Ãltimos anos, observa-se o surgimento e a rÃpida disseminaÃÃo do serviÃo VoIP, integrando-se ao mercado atual junto à telefonia convencional e Ãs redes celulares. Por ser uma alternativa tecnolÃgica que contribui para minimizar ustos, assiste-se a uma preferÃncia crescente por fazer fegar a voz atravÃs das redes IP. O HSDPA, como sistema celular, permite a transmissÃo de dados em alta velocidade, aumenta a largura de banda da rede e abre novas possibilidades de serviÃos multimÃdia, como o VoIP que utiliza a transmissÃo em banda larga para telefones mÃveis. Exige-se, porÃm, um considerÃvel esforÃo de anÃlise deste serviÃo, pois o atraso inerente a esse sistema à um desafio para a garantia de qualidade de voz. Estes fatos justificam, conseqÃentemente, um esforÃo de anÃlise que se detenha sobre a qualidade de voz no VoIP sobre o HSDPA. Para avaliar a qualidade de voz, neste estudo aplica-se o mÃtodo MOS, que faz corresponder valores numÃricos a categorias como medidas de qualidade e inteligibilidade da voz transmitida, obtendo-se esses dados de forma objetiva e subjetiva. O processo de avaliaÃÃo dividiu-se em etapas de acordo com cada metodologia, seguindo recomendaÃÃes tÃcnicas e atravÃs de simulaÃÃes computacionais dinÃmicas. Na avaliaÃÃo objetiva, utilizou-se o algoritmo PESQ para obtenÃÃo do conceito MOS, enquanto que na avaliaÃÃo subjetiva, arquivos de voz com certo percentual de erro foram colocados em um endereÃo na Internet para escuta e atribuiÃÃo de nota MOS, baseada na percepÃÃo do usuÃrio ouvinte. Os resultados mostraram que os dois mÃtodos de avaliaÃÃo obtiveram conceitos de qualidade satisfatÃrios, que o QoS nas simulaÃÃes à estÃvel e positivo, que uma boa qualidade para os arquivos de voz e a provÃvel satisfaÃÃo dos usuÃrios do serviÃo VoIP sobre o sistema celular HSDPA à garantida para uma taxa de 2% de FER. Finalmente, mostra-se que a metodologia objetiva garante a obtenÃÃo de notas MOS aproximadas da subjetiva, evitando o Ãrduo trabalho de fazerem-se ouvir diversos arquivos de voz por uma quantidade significativa de usuÃrios para ser vÃlida estatisticamente. / In recent years, we can observe the development and fast dissemination of VoIP services, being integrated by the present market, beside conventional telephony and cellular networks. For being a technological alternative that contributes to minimize costs, we see an increasing preference for voice transmission through IP networks. HSDPA, as a cellular system, allows high speed data transmission,increases the network width of band and creates new possibilities for multimedia services, as VoIP that transmits in wideband to mobile telephones. The delay inherent to this system, however, is a challenge for the need to assure good quality of voice transmissions, demanding a considerable effort of analysis of this service.These facts justify a study that focus on the quality of voice in VoIP over HSDPA.To evaluate the voice quality, in this study, we applied MOS method that makes numerical values correspond to categories like quality and intelligibility of transmitted voice, getting these data through objective and subjective methodologies. The evaluation process was divided in fases according to the characteristics of each methodology and to technical recommendations, and was done through dynamic computational simulations. For objective evaluation process,algorithm PESQ was employed to obtain MOS concepts, whereas, for subjective evaluation, voice files with a percentage of error have been placed in Internet for listening and for the attribution of MOS concepts based in the perception of the listener. The results of this research show that both evaluation methods got satisfactory concepts of quality, that QoS is steady and positive in the simulations, that a good quality for the voice files and the probable satisfaction of the users of the VoIP service on cellular system HSDPA is guaranteed for 2% FER rate. Finally, it shows that MOS concepts produced by objective methodology were close enough to those given by subjective evaluation to dispense with the arduous work of making diverse voice files to be heard and subjectively evaluated by a statisticaly valid amount of users.

Voz sobre IP (ou VoIP) vem sendo adotada progressivamente não apenas por um grande número de empresas mas também por um número expressivo de pessoas, no Brasil e em outros países. Entretanto, essa crescente adoção de VoIP no mundo traz consigo algumas preocupações tais como ameaças e riscos de segurança, sobretudo no que diz respeito à autenticidade, privacidade e integridade da comunicação. Para proteger a sessão de mídia existem protocolos muito eficientes, como o Secure Real-time Transport Protocol (SRTP). Mas ele depende de uma chave secreta para tornar a comunicação segura de fato. Assim, uma boa estratégia é aproveitar o processo de sinalização que estabelece a sessão de mídia e negociar uma chave secreta de sessão que seja comum às partes comunicantes. Esse processo de sinalização é realizado por tipos específicos de protocolo tais como o Session Initiation Protocol (SIP), um protocolo de sinalização muito importante e que vem sendo usado cada vez mais por softphones para comunicação na Internet. Todavia, os riscos e ameaças mencionados já existem no próprio processo de sinalização e, dentre eles, o ataque do tipo man-in-the-middle é o mais perigoso, devido aos prejuízos que ele pode causar. Depois de fazer uma revisão bibliográfica dos riscos e ameaças inerentes ao SIP, bem como de seus mecanismos de segurança (analisando os pontos fortes e de atenção deles), foi possível originar um novo mecanismo de segurança, o qual é apresentado neste trabalho. O mecanismo proposto usa um protocolo para troca segura de informações o protocolo Massey-Omura o qual, quando combinado com emparelhamentos bilineares, provê ao SIP um melhor nível de segurança em todos os aspectos (autenticidade, privacidade e integridade). Além disso, o novo mecanismo é avaliado através de uma prova de conceito, na qual utilizou-se um softphone SIP funcional. A análise de segurança realizada e os resultados obtidos da prova de conceito fazem do mecanismo de segurança proposto uma alternativa viável para o SIP. / Voice over IP (or VoIP) has been progressively adopted not only by a great number of companies but also by an expressive number of people, in Brazil and in other countries. However, this increasing adoption of VoIP in the world brings some concerns such as security risks and threats, mainly on the authenticity, privacy and integrity of the communication. In order to protect the media session, efficient protocols like the Secure Real-time Transport Protocol (SRTP) have been used. However, it depends on a secret key to make the communication secure. Thus, a good strategy is to take advantage of the signaling process to establish the media session, and agree on a common secret session key between the communicating parties. This signaling process is performed by specific types of protocols such as the Session Initiation Protocol (SIP), a very important signaling protocol, which has been used more and more by softphones in the Internet communication. Nevertheless, those risks and threats already exist in the own signaling process and, among them, the man-in-the-middle attack is the worst of all due to its high danger degree. After doing a bibliographical revision of the SIP security risks and threats, as well as its security mechanisms (analyzing their advantages and drawbacks), it was possible to generate a new security mechanism, which is presented in this work. The proposed mechanism uses a protocol for secure information exchange the Massey-Omura protocol which, when combined with bilinear pairings, provides a better security level for SIP in all its aspects (authenticity, privacy and integrity). Besides this, the new mechanism is evaluated by a proof of concept, in the which a functional SIP softphone was used. The security analysis and the results obtained from the proof of concept, make the proposed security mechanism a viable alternative for SIP.

Made available in DSpace on 2014-12-17T14:55:42Z (GMT). No. of bitstreams: 1 patriciaAC_DISSERT.pdf: 1727038 bytes, checksum: 6d92b0a685d31e2550d0b963726e444f (MD5) Previous issue date: 2010-02-19 / This work deals with experimental studies about VoIP conections into WiFi 802.11b networks with handoff. Indoor and outdoor network experiments are realised to take measurements for the QoS parameters delay, throughput, jitter and packt loss. The performance parameters are obtained through the use of software tools Ekiga, Iperf and Wimanager that assure, respectvely, VoIP conection simulation, trafic network generator and metric parameters acquisition for, throughput, jitter and packt loss. The avarage delay is obtained from the measured throughput and the concept of packt virtual transmition time. The experimental data are validated based on de QoS level for each metric parameter accepted as adequated by the specialized literature / Este trabalho trata de estudos experimentais a respeito de conex?es VoIP em redes WiFi 802.11b com mobilidade de um dos usu?rios envolvidos na conex?o de voz e, por conseguinte, na presen?a de handoff. Os experimentos s?o realizados em ambientes indoor e outdoor com foco na medi??o dos ar?metros de desempenho usualmente tidos como indicadores da qualidade de servi?o - QoS em aplica??es VoIP: atraso, vaz?o, jitter, e perda de pacotes. Os par?metros de desempenho s?o obtidos com o aux?lio das ferramentas Ekiga, Iperf e Wimanager que possibilitam, respectivamente, simular uma conex?o VoIP, injetar tr?fego controlado em um ambiente de rede WiFi e medir a vaz?o, o jitter e a perda de pacotes. O atraso m?dio ? obtido analiticamente a partir da vaz?o medida e do uso do conceito de tempo de transmiss?o virtual m?dio de um pacote de voz. A aferi??o da aceita??o dos resultados ? feita com base nos n?veis de servi?os tidos como adequados na literatura para cada uma das m?tricas obtidas nos experimentos

Made available in DSpace on 2015-04-11T14:03:12Z (GMT). No. of bitstreams: 1 DISSERTACAO ANDERSON.pdf: 945892 bytes, checksum: 13a676735f1c713ef1251e829d5e41bb (MD5) Previous issue date: 2006-09-13 / Fundação de Amparo à Pesquisa do Estado do Amazonas / More and more the demanding for mobility has been wanted by the users who want to have everything no matter where they are. The applications have been modified in order to attend this needs demanded by the new environment without any predictable infrastructure or topology. The adhoc networks are very popular when concerning about non-infrastructures environment and many studies have been started looking for solutions and strategies to understand and adapt the applications originally created to work over client/server architecture. Voice over IP applications also have to be adapted to work properly over this environment and this study try to identify the factors which can impact on application performance. It is made looking at the look up algorithms used by them to find other nodes around the net. / A medida que a mobilidade se torna uma exigência cada vez maior dos usuários de redes de computadores, as aplicações precisam adaptarem-se ao novo ambiente descentralizado, que por vezes, é totalmente imprevisível e mutável. As redes adhoc estão muito difundidas em cenários totalmente descentralizados e desprovidos de infra-estrutura fixa e por isso diversos estudos estão sendo realizados visando a adaptação de aplicações originalmente construídas sobre a luz da arquitetura cliente/servidor. As aplicações de voz sobre IP (VoIP) também precisam se adaptar a nova realidade, e para isso foram adotadas diversas estratégias de descentralização, sendo uma delas a utilização de algoritmos de busca P2P para prover a independência de pontos centralizadores. Este trabalho realiza um estudo do comportamento desse tipo de algoritmo sobre redes adhoc visando identificar os fatores que podem influênciar direta ou indiretamente a performance de uma aplicação VoIP que os utilize como m´método de localização de seus nós.

Pour avoir une idée sur la qualité du réseau, la majorité des acteurs concernés (opérateurs réseau, fournisseurs de service) se basent sur la Qualité de Service (Quality of Service). Cette mesure a montré des limites et beaucoup d’efforts ont été déployés pour mettre en place une nouvelle métrique qui reflète, de façon plus précise, la qualité du service offert. Cette mesure s’appelle la qualité d’expérience (Quality of Experience). La qualité d’expérience reflète la satisfaction de l’utilisateur par rapport au service qu’il utilise. L’évaluation de la qualité d’expérience est devenue primordiale pour les fournisseurs de services et les fournisseurs de contenus. Cette nécessité nous a poussés à innover et mettre en place des nouvelles méthodes pour estimer la QoE. Dans cette thèse, nous travaillons sur l’estimation de la QoE dans le cas des communications Voix sur IP et dans le cas de la vidéo sur IP. Nous étudions les performances et la qualité des codecs iLBC, Speex et Silk pour la VoIP et les codecs MPEG-2 et H.264/SVC pour la vidéo sur IP. Nous étudions l’impact que peut avoir la majorité des paramètres réseaux, des paramètres sources (au niveau du codage) et destinations (au niveau du décodage) sur la qualité finale. Afin de mettre en place des outils précis d’estimation de la QoE en temps réel, nous nous basons sur la méthodologie Pseudo-Subjective Quality Assessment. La méthodologie PSQA est basée sur un modèle mathématique appelé les réseaux de neurones artificiels. En plus des réseaux de neurones, nous utilisons la régression polynomiale pour l’estimation de la QoE dans le cas de la VoIP. / Quality of Experience (QoE) is the key criteria for evaluating the Media Services. Unlike objective Quality of Service (QoS) metrics, QoE is more accurate to reflect the user experience. The Future of Internet is definitely going to be Media oriented. Towards this, there is a profound need for an efficient measure of the Quality of Experience (QoE). QoE will become the prominent metric to consider when deploying Networked Media services. In this thesis, we provide several methods to estimate the QoE of different media services: Voice and Video over IP. We study the performance and the quality of several VoIP codecs like iLBC, Speex and Silk. Based on this study, we proposed two methods to estimate the QoE in real-time context, without any need of information of the original voice sequence. The first method is based on polynomial regression, and the second one is based on an hybrid methodology (objective and subjective) called Pseudo-Subjective Quality Assessment. PSQA is based on the artificial neural network mathematical model. As for the VoIP, we propose also a tool to estimate video quality encoded with MPEG-2 and with H.264/SVC. We studied also the impact of several network parameters on the quality, and the impact of some encoding parameters on the SVC video quality. We tested also the performance of several SVC encoders and proposed some SVC encoding recommendations.

Voice over Internet Protocol (VoIP) is a service that has recently gained a lot of attention from the telecommunications (telecom) world since both Internet service providers (ISPs) and telecommunications operators have realized the important advantages that it can offer. Although traditional telephony is well established both in the telecom world and in our daily lives, VoIP is now competing with it by offering cost savings, simplicity, and introducing new ways of communicating. Internet service providers have already started deploying efficient VoIP services for their customers and carriers are transforming their network infrastructures in order to be able to accommodate the requirements of VoIP traffic. There are a lot of essential factors that both providers and carriers have to take into consideration in order to efficiently build and operate VoIP technologies. Proper service planning and well-established monitoring and troubleshooting procedures are vital for successful VoIP service. This thesis focuses on commercial VoIP implementation at the carrier’s side and investigates how a carrier can efficiently maintain and troubleshoot their VoIP infrastructure so as to comply with the Service Level Agreements (SLAs) they have signed with their customers (ISP providers), as well as analyses proactive actions that can betaken for minimizing the resources required for customer support. As an outcome, this thesis presents efficient ways of network planning and monitoring, as well as it provides conclusions regarding what are the efficient methods for troubleshooting the carrier’s VoIP products inboth technical and organizational level. / Röst över Internet Protokoll (VoIP) är en tjänst som nyligen har fått ökad uppmärksamhet inom telekommunikations (telecom) branschen eftersom att både Internetleverantörer (ISPs) och telecom operatörer har insett vilka fördelar som tjänsten erbjuder. Även om traditionell telefoni är väl etablerad i både telecombranschen och vår vardag, så kan VoIP konkurrera genom att erbjuda kostnadsbesparingar, förenkling, och introducera nya sätt att kommunicera på. IP leverantörer har redan påbörjat lansering av effektiva VoIP tjänster till sina kunder och telecom carriers bygger om sin nätverksstruktur för att möta kraven av VoIP traffik. Det finns många faktorer att bejaka för både IP leverantörer och telecom carriers för att effektivt bygga och driva VoIP nätverk. Noggrann produktplanering och väletablerad övervakning samt felsökningsprocedurer är en vital del i en framgångsrik VoIP tjänst. Denna avhandling fokuserar på VoIP implementering hos en telecom carrier och hur en telecom carrier effektivt kan underhålla och felsöka VoIP infrastruktur för att möta de servicenivåavtal de har skrivit med sina kunder (IP leverantörer), samt analysera det förebyggande åtgärder som kan tas för att minimera de resurser som behövs till kundtjänst. Denna avhandling presenteras effektiva tillvägagångssätt för planering och övervakning samt erbjuder effektiva,teknisk och organisationella metoder för felsökning av en telecom carriers VoIP produkter.