Open source PBX Kamailio and OpenSIPS diploma thesis covers familiarization with appointed SIP exchanges and with their power comparing. A detailed installation instructions on the operating system Ubuntu is the aim of this work too. The work includes the historical development of telephone exchanges with a focus on the latest generation. The following is SIP protocol basic description and components that can be composed SIP exchanges. Another part is devoted to the development of exchanges Kamailio and OpenSIPS. The thesis contain the archutecture and configuration file description. The practical part of the thesis deals with high-capacity switches, and comparing it in terms of memory and computational demands. Selected measurements are compared with the Asterisk PBX.

Αντικείμενο της εργασίας μας, είναι η δημιουργία ενός γραφικού περιβάλλοντος για τη διαχείριση του συστήματος VOIP υπηρεσιών. Αυτό είναι ο Opensips Web Manager, μια ολοκληρωμένη λύση, από άποψη διαμόρφωσης, χρήσιμο για τον διαχειριστή ενός τηλεφωνικού συστήματος. Ο OWM είναι ένα web user interface, που έχει σχεδιαστεί με PHP. Χρησιμοποιείται για τη διαχείριση των πληροφοριών του συστήματος όσο και για τη διαχείριση της κατάστασης λειτουργίας των εξυπηρετητών, που στηρίζουν ένα VOIP σύστημα. O OWM. Ο διαχειριστής έχει πρόσβαση στον OWM μέσω ενός web browser και έτσι η φυσική θέση του διαχειριστή δεν περιορίζεται από την φυσική θέση του SIP εξυπηρετητή. Η αναζήτηση πληροφοριών, που αφορούν συνδρομητές, στοιχεία κλήσεων κ.α. γίνεται με τη δημιουργία «query», προς τη βάση δεδομένων του SIP εξυπηρετητή. Η μέθοδος ανάπτυξης της εφαρμογής μας ήταν δισδιάστατη. Αρχικά έγινε η υλοποίηση συστήματος VOIP υπηρεσιών, που περιελάμβανε τις διαμορφώσεις των εξυπηρετητών DHCPs , DNSs και OpenSIPs. Ακολούθησε η εγκατάσταση του WEBs εξυπηρετητή, που φιλοξενεί τον OWM. Τέλος σχεδιάστηκε το web user interface. / The subject of our project is to create a graphical user interface for managing systems that provide VOIP services. This is the Opensips Web Manager, a integrated solution regarding configuration, useful for the administration of a telephone system. OWM is a web user interface, designed using PHP. Especially OWM is used to manage system’s data and to control server’s status, which support a VOIP system. The administrator has access to the OWM through a web browser, so the physical location of the manager is not limited by SIP server’s physical location. The search for subscriber’ s and calling feature’s data, is performed by creating «query», to SIP server’s database. The method we use to develop our application has been two-dimensional. First was the implementation of VOIP service, which included the configurations of servers like DHCPs, DNSs and OpenSIPs. Next step was the installation of a WEB server that hosts OWM. Finally, we designed the web user interface.

This work explores, analyzes and rate infuence of VoIP attacks on open source pbx functionality. It describes how voice over IP attacks are achieved according to security standards. There are described concepts and basics of VoIP networks with orientation on facts necessary to understand analyzed actions and measurements in theoretical part. In practical part, there is described realization of attack tests according to instructions with first orientation on initial checking of testing device, pbx’s and security attack possibilities and then complex creation and testing of attack scenarios types such as fuzzing and denial of service attacks.

Over the course of last decade, the demand for VoIP (Voice over Internet Protocol) applications has increased significantly among enterprises and individuals due to its low cost. This increasing demand resulted in a significant increase in users who require reliable VoIP communication systems. QoS (Quality of Service) is a major issue in VoIP implementation and is a method to impel the development of real-time multimedia services like VoIP and videoconferencing. However, there are certain challenges in achieving QoS for VoIP application, which need special attentions; like latency and packet loss. The VoIP servers which are functioning on single core software/hardware model have high latency and packet loss issues due to their limited processing bandwidth. A multicore software/hardware model is the solution to cope up with the increasing demands of VoIP and yet an active research area in telecommunication. Using a multicore software/hardware model for VoIP has several challenges, one of the challenges is to design and implement QoS Benchmarking module for VoIP client and server on multicore. In this thesis the focus is on latency and packet loss of SIP messages on OpenSIPS server. This is done by performing stress testing for QoS benchmarking, where delay and call drop rate is calculated for SIP (Session Initiation Protocol) signaling messages on parallel VoIP client server model. The model is built in C for multicore and is used as a simulation tool. SIP is widely deployed protocol for call establishment, maintenance and termination in VoIP.