There are many disadvantages (cost, lack of security, difficult to manage large networks, support to non-sensitive applications, delay, etc.) associated with traditional networking, IP network, ATM and Frame relay networking. To solve this, an MPLS-based VPN networking is introduced that can work with existing deployed backbones and allow organizations to interconnect the dispersed sites and remote workers through secure links by using public internet. In this thesis, we are trying to build a better understanding to MPLS VPN and we researched to analyze the behavior of OSPF and RIPv2 based MPLS-BGP VPN architectures by using intense VoIP traffic. Then it comes with an OPNET simulation process and scenarios for MPLS-BGP VPN. At last, the conclusion is made: OSPF based MPLS-BGP VPN architecture has lower VPN delay, background traffic Flow delay, LSP delay and point-to-point Queuing delay, and has better performance in VPN load and VPN throughput that can acquire customer satisfaction and confidence as compared to the RIPv2 based MPLS-BGP VPN architecture. / Det finns många nackdelar (kostnader, bristande säkerhet, svåra att hantera stora nätverk, stöd till icke-känsliga tillämpningar, delay, etc.) i samband med traditionella nätverk, IP-nätverk, ATM och Frame Relay nätverk. För att lösa detta, är ett MPLS-baserat VPN nätverk införs som kan arbeta med befintliga sättas samman och låter organisationer för att förbinda de spridda platser och distansarbetare genom säkra länkar genom att använda publika Internet. I denna avhandling försöker vi bygga en bättre förståelse för MPLS VPN och vi forskat för att analysera beteendet hos OSPF och RIPv2 baserad MPLS-VPN BGP arkitekturer med hjälp av intensiv VoIP-trafik. Då kommer med en OPNET simulering process och scenarier för MPLS-BGP VPN. Äntligen är den slutsatsen: OSPF bygger MPLS-VPN BGP arkitektur har lägre VPN dröjsmål bakgrund trafikflödet dröjsmål, LSP dröjsmål och punkt-till-punkt Queuing dröjsmål, och har bättre prestanda i VPN-belastning och VPN som kan få kunden tillfredsställelse och förtroende jämfört med RIPv2 baserad MPLS-VPN BGP arkitektur.

Projektet har till syfte att undersöka den revolutionerande tekniken bakom IP-telefoni. Eftersom tekniken är relativt ung så finns det fortfarande mycket att lära om hur det fungerar och hur den används. Projektet behandlar fördelar och nackdelar samt vilka hot som finns och hur man kan känna sig säker när man använder IP-telefoni till vardags. Det primära objektivet med projektet är att få en helhetsbild och förstå hur tekniken verkligen fungerar. Vidare behandlas telefonens historia då det är viktigt att se var tekniken har sina rötter och hur den utvecklats sedan dess. Som en röd tråd från telefonens teknologi och det publika telefonnätets rötter görs en jämförelse mellan de två teknologierna för att se skillanderna och förmodade fördelar och nackdelar med de båda systemen. Olika sorters protokoll som har ett nära förhållande till IP-telefoni så som SIP, RTP, UDP och IP behandlas. För att få en sanningsenlig bild över hur vardagliga användare upplever IP-telefoni så har en enkätundersökning angående vanor och möjliga problem gjorts hos de anställda på Region Skåne.

Depuis plusieurs années, on observe une multiplication des services de communication en temps réel de type Over-The-Top (OTT). Ces solutions utilisent l¿Internet « best-effort » et s¿adaptent aux fluctuations du réseau. Néanmoins, il est discutable que l¿approche OTT soit suffisante pour fournir une qualité de service de communication acceptable quelles que soient les conditions réseaux. Dès lors, est-il possible d¿utiliser l¿assistance réseau pour améliorer la qualité de service des solutions OTT ?Pour traiter cette question, nous étudions tout d¿abord les solutions OTT, et particulièrement la technologie WebRTC. Nous identifions trois stratégies de couplage lâche qui permettent de tirer parti des mécanismes réseaux pour améliorer la qualité de service des solutions OTT.Nous vérifions la pertinence de ces stratégies dans le contexte de la gestion du trafic. On identifie deux approches de gestion du trafic adaptées à WebRTC : 1) qui assure des délais d¿attente courts quel que soit le trafic ou 2) qui isole le trafic sensible. On évalue ces solutions et leur impact sur WebRTC, pour les réseaux d¿accès filaire (uplink, ADSL et fibre). Les résultats obtenus montrent que les pratiques actuelles de gestion du trafic ne sont pas adaptées au trafic WebRTC. De plus, les solutions proposées assurent plus d¿équité entre le trafic WebRTC et TCP et elles permettent d¿éviter que le trafic WebRTC soit désavantagé et elles améliorent la qualité de communication.Enfin, ces solutions de la gestion du trafic sont positionnées dans le contexte des stratégies de couplage proposées. A partir de là, on fournit des recommandations pour améliorer la qualité WebRTC avec l¿assistance du NSP. / The number of real-time Over-The-Top (OTT) communication services has increased in the recent years. OTT solutions use the best-effort Internet delivery and rely on mechanisms built into the endpoints to adapt to underlying network fluctuations. Nevertheless, it is questionable if this approach is enough to provide acceptable quality of communication regardless the network conditions. Therefore, can network assistance be used to improve the quality of OTT real-time communication services?To address this question, we study OTT solutions with a focus on WebRTC. We identify three loose coupling strategies that leverage network mechanisms for improving OTT communication services quality.We verify the pertinence of these coupling strategies in the context of traffic management. We identify two approaches of traffic management solutions adapted to WebRTC traffic: 1) aiming at assuring lower queuing delays regardless the traffic or 2) isolating the sensitive traffic. We study the impact of identified traffic management solutions on WebRTC for wireline access networks (uplink, ADSL and fiber). The obtained results show that current Internet engineering practices are not well adapted to the WebRTC traffic, but are optimized for TCP traffic. Furthermore, the proposed solutions ensure more fairness between WebRTC and TCP flows and consequently enable avoiding WebRTC traffic starvation and improve the overall quality of the communication.In the final analysis, the evaluated traffic management solutions are positioned in the context of identified coupling strategies. Based on this assessment, we provide recommendations of improving WebRTC quality with the assistance of NSP.

This master thesis focuses on the worldwide optical connection situation.Within the respective regions of Asia-Pacific,the Americas and Europe, it identifies the main countries and focuses on the situation and share of FTTx subscribers in them.It tries to answer the question of what technologies are represented the most on particular markets. Within each country,two service providers are chosen and their offer analyzed.For an overview,the price offer for an Internet connection with the highest transmission rate and the price for VoIP service are provided. The analytical part uses the obtained data from individual countries and regions for a price comparison and market forecast.The price comparison deals with the price level of a standard connection in relation to GDP per capita in PPS.This comparison enables the discovery of how expensive the Internet connection via FTTx is in a particular country.Furthermore, the analytical part, in relevant countries, focuses on the calculation of the ratio of costs for the yearly use of a FTTx subscription in relation to the disposable family income. The highest ratio-8.37%-was recorded in India. The thesis also includes a market forecast in the horizon of the end of 2009 and 2010 for the most important countries.For each country,the main growth factor is identified and subsequently the development for the entire year 2009 and 2010 is forecast.The worldwide forecast is 61.37 million (as of the end of 2009) and 72.67 million (as of the end of 2010) FTTx subscriptions.The contribution of this master thesis is the analysis of in what state of development the market of Internet optical connection is in the most important countries, with price comparisons and a market forecast constituting further contributions.

This master’s thesis dissertate the possibilities of the linkage between Asterisk softswitch and the 4th generation private branch exchange. This should create a new generation’s network, so-called NGN, by the convergence of existing telecommunication networks with an IP computer network. This master’s thesis is divided into several chapters. In introduction is described the evolution of the private branch exchanges as well as the principles of the voice digitizing, codecs and signaling commonly used in both TDM and VoIP networks. The main aim of this project is the configuration of Asterisk software exchange for connection with PBX Alcatel 4400 as well as public phone network PSTN. Another goal of this master’s thesis was the configuration of Alcatel PBX and diagnostics of CCS and CAS signaling on E1 interface. In conclusion there are summarized advantages of NGN networks and their utilization in the future.

The work deals with the developmental stages (Releases) of the third generation UMTS net-works. It explains a partial improvement from Release 99 to Release 5 and Release 6 to im-prove the quality parameters of a mobile data transmission. In the practical part, there are measured qualitative parameters of the third generation networks that are available in the Czech Republic: UMTS, HSDPA, CDMA 1xEV-DO Rev.0 and Rev.A and UMTS-TDD. The measurements give reasonably achievable data rates in both directions (downlink & uplunk), a response time, quality of VoIP calls, a dispersion, a data-loss and an ability to transmit video with different data flows. The other aspects especially an availability of coverage and end-user terminals have been assessed wihin the each network. The work is supplemeted with a laboratory, which is focused on a practical comparison of the basic parameters of data transmissions of 2G and 3G networks.

This master's thesis deals with open source PBX security against security attacks. In the theoretical part is detailed description of problematic about attacks that could be used on VoIP systems with high focus on the Denial of Service attack. Furthermore are in theoretical part described methods of security of initialization protocol SIP. Individual chapter is devoted to intrusion detection and prevention of IDS and IPS systems, focusing on Snort and OSSEC. In the practical part of the work was created generator of attacks against various PBX systems, which was subsequently used for detailed testing. Special tests of PBX system are then used against DoS attacks, for which was created protection in form of active elements consisting of IDS Snort & OSSEC. These are capable to provide protection in real-time. The protection was tested on particular PBX systems and in matter of comparison were measured possibilities before and after of security implementation. The output of this work is attacks generator VoIPtester and creation of configuration rules for Snort and OSSEC.

The thesis examines the impact of VoIP transmission channel characteristics on speech parameters. It seeks ways of emulating properties of a selected VoIP transmission channel and creating a network emulation environment. Several scripts have been created in the Matlab development environment and used to modify and divide a continuous recording into parts identical to the original speeches, before passing through the transmission channel. Subsequently a database of speech recordings is created, as affected by selected characteristics - jitter, bandwidth, loss. Within these databases, symptoms are sought as the most evident characteristics of the transmission channel. Using correlation, symptoms are selected that are best suited for automated determination of the properties of transmission channel characteristics such as jitter, loss and bandwidth.

The theses explains the Evolution Packet System (EPS) and subsystem IP Multimedia Subsystem (IMS). It also informs how these systems cooperate and through which interfaces the main data and signals between these two technologies are transfered. The thesis includes practical section, in which installation and configuration of sections of the IMS system including its "core", and application and medial server. For verification of the functionality of the system, two dissimilar clients are installed, and emulator WAN is applied for simulation of Internet. The product of the theses is the design of the laboratory exercise drafted in such a way that it can be solved by students less knowledgeable of the issues.

Šifrování multimediálních datových přenosů v reálném čase je jednou z úloh telekomunikační infrastruktury pro dosažení nezbytné úrovně zabezpečení. Rychlost provedení šifrovacího algoritmu může hrát klíčovou roli ve velikosti zpoždění jednotlivých paketů a proto je tento úkol zajímavým z hlediska optimalizačních metod. Tato práce se zaměřuje na možnosti paralelizace zpracování SRTP pro účely telefonní ústředny s využitím OpenCL frameworku a následnou analýzu potenciálního zlepšení.