Développement d’un Serveur SIP pour les Réseaux Multimédia

Abstract

As part of our final-year project, we developed a Voice over IP (VoIP) telephony solution based on the Asterisk server running on Debian 12. First, we studied the fundamental concepts of VoIP and the main protocols used in IP voice communications, including SIP, H.323, RTP, and RTCP. This theoretical study enabled us to gain a better understanding of IP telephony networks and the signaling and media transport mechanisms involved in voice communications. For the practical implementation, VMware Workstation was used to create a virtual machine running Debian 12. Within this environment, the Asterisk server was installed and configured, and several SIP accounts were created to establish voice communications between users through the MicroSIP softphone. In addition, security mechanisms were implemented to enhance the reliability of the solution, and an interconnection between the VoIP network and the Public Switched Telephone Network (PSTN) was established. Finally, a series of tests was conducted to verify the proper operation of the system and validate the different configurations performed. The results demonstrate that an Asterisk-based VoIP solution represents a reliable, flexible, and cost-effective alternative to traditional telephony systems while providing significant opportunities for scalability and integration.