The purpose of this project is to update the tool of Network Traffic Recognition System (NTRS) which is proprietary software of Ericsson AB and Tsinghua University, and to implement the updated tool to finish SIP/VoIP traffic recognition. Basing on the original NTRS, I analyze the traffic recognition principal of NTRS, and redesign the structure and module of the tool according to characteristics of SIP/VoIP traffic, and then finally I program to achieve the upgrade. After the final test with our SIP data trace files in the updated system, a satisfactory result is derived. The result presents that our updated system holds a rate of recognition on a confident level in the SIP session recognition as well as the VoIP call recognition. In the comparison with the software of Wireshark, our updated system has a result which is extremely close to Wireshark’s output, and the working time is much less than Wireshark. In the aspect of practicability, the memory overflow problem is avoided, and the updated system can output the specific information of SIP/VoIP traffic recognition, such as SIP type, SIP state, VoIP state, etc. The upgrade fulfills the demand of this project.

Voice over Internet Protocol (VoIP) over WLAN is one of the most important technologies in today’s world of communication. VoIP is simply a way to make phone calls through the internet because of the convergence of voice and data networks enables new applications and cost reductions. Voice over WLAN phones are already being offered to enterprises by leading vendors. Most of internet services or applications require centralized network to communicate, but with Ad-hoc networks there is no such requirement at all. In this report we have established a VoIP session by forming a network between Android mobile devices without using an Access point. Energy consumption is a major problem for VoIP over wireless LAN applications while using them in hand held devices. We investigated the energy consumption characteristics of our Evaluation kit during VoIP session. We further studied about new technology: “Wi-Fi Direct” which allows Wi-Fi equipped devices to share data without using wireless access points.

Για περιοχές με μικρή ή καθόλου επίγεια τηλεπικοινωνιακή υποδομή η επικοινωνία μέσω δορυφορικής σύνδεσης είναι μία αποτελεσματική και λογικού κόστους λύση. Το DVB-RCS (Digital Video Broadcast - Return Channel System) standard επιτρέπει την αμφίδρομη μετάδοση IP κίνησης πάνω από δορυφορικό κανάλι κάνοντας έτσι δυνατή την VoIP επικοινωνία μέσω δορυφόρου. Η μετάδοση VoIP μέσω μιας δορυφορικής σύνδεσης όμως, μπορεί προκαλέσει ορισμένα προβλήματα τόσο στην ίδια την ποιότητα της VoIP συνδιάλεξης, όσο και σε άλλες υπηρεσίες που ενδεχομένως μοιράζονται με τις VoIP ροές το διαθέσιμο εύρος ζώνης του δορυφορικού καναλιού. Η διπλωματική αυτή εργασία επικεντρώνεται στη μελέτη και ανάπτυξη end to end μηχανισμών ελέγχου του ρυθμού κωδικοποίησης και μετάδοσης στα VoIP τερματικά τηλέφωνα ανάλογα με τον βαθμό συμφόρησης του καναλιού, ώστε να αξιοποιείται πιο αποτελεσματικά το διαθέσιμο εύρος ζώνης του καναλιού μετάδοσης (δορυφορικού ή μη), να αντιμετωπίζεται ενδεχόμενη συμφόρησή του, να βελτιώνεται η ποιότητα της συνδιάλεξης, να αυξάνεται ο μέγιστος αριθμός των ταυτόχρονων συνδιαλέξεων για ένα δεδομένο εύρος ζώνης και να εξασφαλίζεται ο δίκαιος διαμοιρασμός του διαθέσιμου εύρους ζώνης ανάμεσα στις TCP και VoIP ροές. Σε αυτά τα πλαίσια αρχικά εξετάζονται οι παράγοντες που επιδρούν στην ποιότητα της VoIP υπηρεσίας, παρουσιάζονται ορισμένοι τρόποι για την αξιολόγηση της και γίνεται μελέτη της χρήσης του TFRC (TCP- Friendly Rate Control) μηχανισμού σε VoIP εφαρμογές. Προτείνεται ένας νέος, πολύ απλός στην υλοποίηση, μηχανισμός ελέγχου μετάδοσης για VoIP ροές, που αντιθέτως με τους υπάρχοντες μηχανισμούς, στοχεύει ταυτόχρονα στην βελτίωση της ποιότητας της συνδιάλεξης και στην φιλικότητα προς τις TCP ροές. Αναλύονται επίσης η δομή, η λειτουργία και ορισμένα θέματα υλοποίησης του VoIP τερματικού συστήματος που αναπτύχθηκε (σε μορφή λογισμικού) στα πλαίσια αυτής της διπλωματικής εργασίας και που υλοποιεί τον προτεινόμενο μηχανισμό ελέγχου μετάδοσης. / For areas with limited or no terrestrial telecommunication infrastructure, communication via satellite is a cost effective alternative. The DVB-RCS (Digital Video Broadcast - Return Channel System) standard supports the bidirectional transmission of IP data, making VoIP communication via satellite possible. However, the transmission of VoIP through a satellite link raises some serious issues concerning the VoIP quality of service and the plain functionality of other applications that might share the same link with the VoIP flows. This thesis focuses in the study and the development of end to end rate control mechanisms in VoIP terminal phones, which mechanisms can enhance the utilization of the available channel (satellite or not) bandwidth, tackle potential congestion, enhance the conversational quality, increase the maximum number of simultaneous VoIP conversations for a given bandwidth, and ensure that bandwidth is being fairly shared between TCP and VoIP flows. The factors that affect the VoIP quality of service, the ways this quality can be evaluated and the use of TFRC (TCP- Friendly Rate Control) mechanism in VoIP are discussed. A new, easy-to-implement rate control mechanism which, in contrast to the existing mechanisms, targets on both conversational quality enhancement and TCP friendliness is proposed. Finally, some implementation issues, regarding the VoIP terminal software system that it has been developed as part of this thesis and implements the proposed rate control mechanism, are discussed.

Το VoIP (Voice Over IP) αναφέρεται στη μετάδοση και τη σηματοδοσία επικοινωνιών φωνής π.χ τηλεφωνικές κλήσεις πάνω από IP δίκτυα όπως είναι το διαδίκτυο. Σκοπός της παρούσας διπλωματικής ήταν η μελέτη της παρεχόμενης ποιότητας ομιλίας VoIP τηλεφωνικών συσκευών (Sitel,Polycom) κάτω από σενάρια διαφορετικών συνθηκών κίνησης στο δίκτυο. Αρχικά παρουσιάζονται οι εφαρμογές του voip και στη συνέχεια αναλύονται η λειτουργία, η αρχιτεκτονική και τα πρωτόκολλα της τεχνολογίας αυτής. Ακολούθως περιγράφεται η πειραματική διάταξη που απαιτήθηκε για να συγκεντρωθούν οι μετρήσεις από τις συσκευές και στη συνέχεια οι γραφικές αναπαραστάσεις των μετρήσεων αυτών που αφορούν της παρεχόμενη ποιότητα ομιλίας. Γίνεται ανάλυση των γραφικών και σύγκριση με ανάλογες της βιβλιογραφίας. / The VoIP technology refers to the transmission of voice samples over the Internet Protocol. The aim of the thesis is to investigate the QoS of a VoIP call under various networks circumstances when various impairments occur. The devices that we put under test belong to Sitel and Polycom company.

Recentemente, as tecnologias de telecomunicações esão convergindo para a concepção da Next Generation Network, onde propõe-se que todas as informações trocadas sejam classificadas por prioridade e segurança. Porém, como as redes atuais ainda não promovem tais práticas, protocolos VoIP, em conjunto a outras soluçõoes, buscam a melhoria da qualidade das ligações. Como o protocolo VoIP IAX vem ganhando credibilidade na comunidade open source nos úlltimos anos, torna-se relevante compará-lo ao protocolo SIP, o qual é bastante investigado pela literatura. Desta forma, o objetivo deste trabalho é o estudo e avaliação dos protocolos SIP e IAX, através de verificações de qualidade do áudio em ligações VoIP. Para a realização dos experimentos foi desenvolvida uma estrutura que representasse chamadas VoIP no simulador Network Simulator e, para tais ligações, empregou-se método de avaliação de qualidade PESQ. Assim, foi possível a verficação das semelhanças compreendidas entre os protocolos SIP e IAX diante dos problemas de perda de pacotes, atraso, limitação da taxa de dados e jitter / Telecommunications technologies are recently converging to the Next Generation Network conception, where it is proposed that all exchanged information should be classied by security and priority. As the currently available networks do not provide such practices, VoIP protocols, among other solutions, aim for the improvement of the calls quality. As the IAX VoIP protocol had been receiving credibility in the open source community in the last years, it is relevant to compare it to the SIP protocol, which is widely investigated in the literature. In this way, the objective of this work is the study and evaluation of the SIP and IAX protocols through verications of audio quality in VoIP calls. To implement the experiments, a structure that represents VoIP calls was developed in the \"Network Simulator\" software. For these calls, the PESQ method was used to evaluate the calls quality. Using this approach, it was possible to verify similarities between the SIP and IAX protocols regarding the problems of packet loss, delay, limitation in the data rate and jitter

Made available in DSpace on 2014-07-29T15:08:23Z (GMT). No. of bitstreams: 1 Dissertacao Christiane Santos EEC.pdf: 970611 bytes, checksum: a2b7f2cf34500ad6f1d66ffdf3bf9a69 (MD5) Previous issue date: 2009-11-24 / This work is motivated by a growing interest in the power lines' applicability as an alternative means of propagation for communication signals, and presents an analysis of VoIP's (Voice over IP) traffic and data transfer using BPL / PLC (PowerLine Broadband / PowerLine Communication) network. We describe the main characteristics of the BPL / PLC and HomePlug standard. As the physical transmission technology used by the BPL / PLC for data transfer is hostile, and it was not developed for this purpose, traffic modeling can be useful for planning and design these networks. A model is proposed based on MMFM (Markov Modulated Fluid Models) to characterize the traffic data and VoIP into PLC networks. Simulations and comparisons were made with other models such as Poisson and MMPP (Markov Modulated Poisson Process). The results were obtained by experiments in low-voltage PLC networks (indoor environment), using a 4,3MHz to 20,9MHz bandwidth / Este trabalho é motivado por um crescente interesse na aplicabilidade das linhas de energia como meio alternativo de propagação de sinais de comunicação, e apresenta uma análise do tráfego VoIP (Voice over IP) e da transferência de dados utilizando a rede BPL/PLC (Broadband powerLine/ PowerLine Communication). São descritas as principais características da tecnologia BPL/PLC e do padrão Homeplug. Como o meio físico de transmissão utilizado pela tecnologia BPL/PLC para transferência de dados é hostil, visto que não foi desenvolvido para esta finalidade, a modelagem de tráfego pode ser útil para o planejamento e dimensionamento dessas redes. É proposto um modelo baseado no MMFM (Markov Modulated Fluid Models) para caracterizar o tráfego de dados e de VoIP em redes PLC. Simulações e comparações foram realizadas com outros modelos como Poisson e o MMPP (Markov Modulated Poisson Process). Os resultados foram obtidos através de experiências realizadas em redes PLC de baixa tensão (ambiente indoor), utilizando uma largura de faixa entre 4,3MHz a 20,9MHz

Les réseaux autonomes représentent un intérêt certain pour les opérateurs de télécommunications. L’auto-diagnostic, pour la détection des pannes et des dysfonctionnements, est une fonction critique dans le cadre de ces réseaux.Nous avons opté pour l’utilisation d’un diagnostic à base de modèles car il permet un diagnostic automatique, distribué et adapté à l'architecture des réseaux autonomes. Ce diagnostic est basé sur une modélisation explicite des comportements normaux ou anormaux du système. Nous utilisons ensuite un algorithme de diagnostic générique qui s'appuie sur cette modélisation pour réaliser l’auto-diagnostic. La modélisation utilisée est à base de graphe causal. Elle est une représentation intuitive et efficace des relations de causalités qui existent entre les observations et les pannes.Notre algorithme d’auto-diagnostic qui s’appuie sur l’utilisation de graphes causaux, fonctionne sur le principe suivant : lorsqu’une alarme est déclenchée, l’algorithme est lancé et, grâce aux relations de causalité entre l’alarme et les causes, les causes primaires vont pouvoir être localisées. Puisque le graphe causal permet une modélisation modulaire et extensible, il est possible de le séparer ou de le fusionner pour répondre aux besoins des services et architectures de communication. Cette caractéristique nous permet de proposer un algorithme distribué qui s’adapte à l’architecture des réseaux autonomes. Nous avons, ainsi, proposé un algorithme d’auto-diagnostic qui permet de réaliser le diagnostic distribué correspondant à l’architecture du réseau autonome afin de réaliser un diagnostic global.Nous avons implémenté cet algorithme sur une plateforme OpenIMS, et nous avons montré que notre algorithme d'auto-diagnostic pourrait être utilisé pour différents types de service. Les résultats obtenus correspondent bien à ce qui est attendu. / The autonomic networks show certain interest to manufacturers and operators of telecommunications. The self-diagnosis, the detection of failure and malfunction, is a critical issue in the context of these networks.We choose based-model diagnosis because it allows an automatic diagnosis, and is suitable to distributed network architecture. This diagnosis is based on an explicit modeling of normal and abnormal behavior of the system. We then use a generic diagnostic algorithm that uses this modeling to perform self-diagnosis. The modeling used is based on causal graph. It is an intuitive and efficient representation of causal relationships between observations and failures.The self-diagnosis algorithm we proposed based on the use of causal graphs. The principle is: when an alarm is triggered, the algorithm is run and, with the causal relationships between alarms and causes, the principal causes will be located. Since the causal graph modeling allows a modular and extensible model, it is possible to separate or merge according to the needs of services and communication architectures. This feature allows us to propose a distributed algorithm that adapts to autonomic network architecture. We have thus proposed a self-diagnosis algorithm that allows for the diagnosis corresponding to the autonomic network architecture to realize a global diagnosis.We have implemented this algorithm on a platform OpenIMS, and we showed that our self-diagnostic algorithm could be used for different types of services. The results of implement correspond to what is expected.

Vivemos atualmente uma era de convergência de tecnologias, motivada por questões tanto econômicas como de caráter operacional, na qual os serviços de dados, voz e vídeo estão migrando rapidamente para uma plataforma IP. Particularmente considerando o paradigma da telefonia IP neste processo de convergência, ocorrem desafios tecnológicos, pois temos de um lado os usuários finais que já possuem uma referência sobre a qualidade da fala, fruto das décadas de uso do sistema telefônico tradicional, e na outra extremidade as operadoras de telecomunicações que em sua última milha dependem de redes estatísticas, sem mecanismos adequados para a garantia de QoS. Assim, se torna vital para o sucesso da operação a devida identificação das relações entre os diversos componentes existentes entre terminais e sua contribuição para a qualidade de fala, percebida pelo assinante, de forma a entregar um serviço com qualidade similar ao Sistema Telefônico Fixo Comutado. Neste contexto, este trabalho busca identificar por meio de técnicas de estatística multivariada uma correlação entre métricas objetivas de Qualidade de Serviços aplicáveis em redes IP e a qualidade subjetiva da fala predita pelo algoritmo Modelo-E definido na recomendação ITU-T G.107. Um método de coleta e análise estatística de informações foi desenvolvido para atingir o objetivo proposto. Para sua validação um ambiente de testes foi criado, dados de operação foram coletados e ferramentas computacionais foram aplicadas para o tratamento analítico e estatístico. Os resultados obtidos pelo método foram então aplicados em campo durante as etapas de testes e homologação de um PABX-IP-IMS desenvolvido para o mercado corporativo. A correlação entre os diversos fatores envolvidos, suas métricas e como todo este sistema impacta na qualidade relativa, percebida pelo usuário final permitirá aos provedores de serviços avaliarem quais as melhores estratégias a serem empregadas em seus segmentos de rede de forma a garantir a excelência no nível de serviço oferecido ao consumidor final. / We live now in an convergence of technologies era, driven by economic and operational issues, where the data services, voice and video are quickly moving to an IP platform. Particularly considering the paradigm of IP telephony in the process of convergence, there are technological challenges. We have subscribers who already have a reference about the quality of speech, derived from decades of using the traditional phone system. At the other end telecom operators that rely on statistical networks, with no possibility to guarantee QoS. So it becomes vital to the operations success the proper identification of the relationships between the various components between the terminals and their contribution to the speech quality perceived by the subscriber in order to deliver a quality service close to the PSTN. In this context, this study sought to identify a correlation between objective metrics for Quality of Service applicable to IP networks and subjective quality of speech predicted by the algorithm \"Model-E\" defined in ITU-T G.107 through multivariate statistical techniques. A method of collecting and analyzing statistical information was developed to achieve the proposed objective. To validate a test environment was created, operation data were collected and computational tools were applied to the analytical and statistical treatment. The results obtained by the method were then applied in the field during the stages of testing and approval of an IMS-IP-PBX designed for the corporate market. The correlation between the various factors involved, their metrics and how the whole system impacts on the quality perceived by end users will enable service providers to assess what the best strategies to use in their network segments to ensure an adequate level of service offered to consumers.

Internet telephony has rapidly becoming an integral part of life. Due to its low incremental cost and the wide availability of voice over IP (VoIP) based services these services being used by nearly everyone. Today there are many VoIP applications available in the market, but most of them lack basic security features. Because people use VoIP services via public hotspots and shared local area networks these VoIP applications are vulnerable to attacks, such as eavesdropping. Today, there is a great need for VoIP applications with high quality security. MiniSIP is an open-source VoIP application platform, initially developed at KTH. High quality security has been a major focus of MiniSIP developments by several students, including the first public implementations of the secure real-time protocol (SRTP) and the Multimedia Key Exchange (MIKEY) protocol. MiniSIP implements secure end-to-end VoIP services. In addition, MiniSIP implements features such as dynamically choosing the most appropriate CODEC during a call, implementing calling policies, etc. However, it suffers from having a complicated GUI that requires the use of many libraries, rendering it both hard to build and hard support – both of which make it unsuitable for commercial purposes. Web browser plug-ins are shared libraries that users install to extend the functionality of their browser. For example, a plug-in can be used to display content that the browser itself cannot display natively. For example, Adobe's reader plugin displays PDF files directly within the web browser. Real Network’s Streaming video player utilizes a browser plug-in to provide support for live video streaming within a web page. Adobe’s Flash player plugin is required to load or view any Flash contents – such as video or animations. The goal of this thesis project is remove the problem of the existing MiniSIP GUIs by developing a Firefox browser plug-in for the MiniSIP application that will utilize a web-browser based GUI. The prototype that will be designed, implemented, and evaluated will implement an open-source VoIP application that is easy for a Firefox browser user to install and will be easy to use via a web interface. The long term goal is to facilitate an ordinary user to utilize VoIP communication via their web browser. A secondary goal is to re-use the code within MiniSIP, while using the web-browser to provide the GUI. / Internettelefoni har snabbt blivit en integrerad del av livet. På grund av dess låga marginalkostnaden och den breda tillgången på Röst över IP (VoIP) tjänster dessa tjänster används av nästan alla. Idag finns det många VoIP-applikationer som finns på marknaden, men de flesta av dem saknar grundläggande säkerhetsfunktioner. Eftersom människor använder VoIP tjänster via offentliga hotspots och delade lokala nätverk dessa VoIP-applikationer är sårbara för attacker, såsom avlyssning. Idag finns det ett stort behov av VoIP-applikationer med hög kvalitet säkerhet. MiniSIP är ett open-source VoIP-program plattform, ursprungligen utvecklats vid KTH. Hög kvalitet säkerhet har varit ett stort fokus på MiniSIP utvecklingen genom att flera studenter, däribland de första offentliga implementeringar av den säkra realtid protokoll (SRTP) och Multimedia Key Exchange (MIKEY) protokollet. MiniSIP implementerar säker början till slut VoIP tjänster. Dessutom genomför MiniSIP funktioner som dynamiskt välja den lämpligaste CODEC under ett samtal, genomföra samtalsstrategier, osv. Men lider den från att ha en komplicerad GUI som kräver användning av många bibliotek, vilket gör det både svårt att bygga och hård stöd - som båda gör det olämpligt för kommersiella ändamål. Webbläsare plug-ins delas bibliotek som användare installerar för att utöka funktionerna i sin webbläsare. Till exempel kan en plug-in kan användas för att visa innehåll som webbläsaren inte själv kan visa inföding. Till exempel visar Adobes Reader plugin PDF-filer direkt i webbläsaren. Real Networks strömmande videospelare använder en plugin-att ge stöd för levande video strömning i en webbsida. Adobe Flash Player plugin krävs för att ladda eller visa en Flash innehåll - såsom video eller animationer. Målet med denna avhandling projektet är bort problemet med befintliga MiniSIP GUI genom att utveckla en Firefox webbläsare plug-in för att MiniSIP programmet som kommer att använda en webbläsare baserad GUI. Prototypen som kommer att utformas, genomföras och utvärderas kommer att genomföra en öppen källkod VoIP-program som är lätt för en Firefox webbläsare användaren att installera och kommer att vara lätt att använda via ett webbgränssnitt. Det långsiktiga målet är att underlätta en vanlig användare att använda VoIP-kommunikation via sin webbläsare. En sekundär målsättning är att återanvända kod i MiniSIP, medan du använder webbläsare för att ge det grafiska gränssnittet.

Användandet av IP-telefoni har de senaste åren ökat och marknaden förutspås fortsätta att växa globalt. Många vill ta del av de fördelar som den nya tekniken har i form av ny funktionalitet och minskade kostnader. Men att migrera telefoni till datanät medför både nya möjligheter men också nya utmaningar. I den här rapporten undersöks hur Asterisk kan användas som ett bra IP-PBX alternativ. Rapporten behandlar även säkerheten i att ansluta ett system med Asterisk till internet genom intrusionstester som utförs på systemet i grundkonfiguration. Dessa tester resulterar i rekommendationer om hur systemet kan konfigureras för att hålla en hög säkerhetsnivå. / The usage of IP-telephony has increased in recent past and the market is expected to continue to grow globally. Many want to take part in the advantages that the new technology brings in form of functionality and reduced costs. But to migrate telephony to data networks brings both new possibilities but also new challenges. This report examines how Asterisk can be used as a good IP-PBX alternative. The report also addresses the security aspect of connecting a system based on Asterisk to the internet through conducting intrusion tests on the system in standard configuration. These tests result in recommendations on how the system can be configured to keep a high security standard.