This study examines the use of SS7 signaling system over IP networks by using the open source PBX YATE. At first it starts with describing the SS7 followed by an explanation of the function of each of its levels and the messages that are used within the SS7 network. The study then sheds some light on the ways of using SS7 inside IP network with the use of some protocols. It also discusses the architecture of YATE and its files, and how it is installed in Linux operating system. Finally, it describes the important files for delivering this task. The study was commenced by using two virtual machines that have two different open source PBX's which are YATE and Asterisk, and after acquiring some results by establishing communication between them via the means of SIP trunk, furthermore the study was extended to the laboratory in order to test it over real servers that have TDM cards, in order to apply the study by the means of SS7 protocols, SIGTRAN, MGCP gateway and SIP-T. The experiments have almost delivered successful communications after conducting a configuration for the files on multiple sides.

<p>Nowadays, VoIP is getting more and more popular. It helps company to reduce cost, extends service to remote area, produce more service opportunities, etc. Besides these advantages, VoIP also put forward security problems.</p><p>In this paper, we introduce the popular protocols in VoIP and their security mechanisms, by introducing threats to VoIP, we point out the vulnerabilities with the security mechanisms of each VoIP protocol, and give recommendation for each VoIP protocol. In the conclusion part, we evaluate the vulnerabilities of each protocol, and point out in the future, with better protocol architecture, enhanced security policies, VoIP will has a brighter future.</p>

Nowadays, VoIP is getting more and more popular. It helps company to reduce cost, extends service to remote area, produce more service opportunities, etc. Besides these advantages, VoIP also put forward security problems. In this paper, we introduce the popular protocols in VoIP and their security mechanisms, by introducing threats to VoIP, we point out the vulnerabilities with the security mechanisms of each VoIP protocol, and give recommendation for each VoIP protocol. In the conclusion part, we evaluate the vulnerabilities of each protocol, and point out in the future, with better protocol architecture, enhanced security policies, VoIP will has a brighter future.

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.

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.