Simulation of two way telephone calls over the IP network using SIP protocol

The primary goal of this research is to simulate and analyse VoIP system that uses SIP as signalling protocol to create, manage and terminate the voice calls. We focused on the use main SIP messages to initiate and terminate the calls also the voice packets loss in the network was analysed.Voice over Internet Protocol (VoIP) defines a way to carry voice calls over an Internet Protocol (IP) network including the digitisation and packetization of the voice streams. Signalling protocols are needed to establish and control sessions or calls in the VOIP system, and one of the main signalling protocols in the IP network is session initiation protocol (SIP). SIP (Session Initiation Protocol) developed by IETF for VoIP signalling is a Communication control protocol capable of running on different transport layers, e.g., TCP, UDP or SCTP.it is used to establish, modify and terminate sessions of the Internet telephony calls. The primary goal of this research is to simulate and analyse VoIP system that uses SIP as signalling protocol to create, manage and terminate the voice calls. We focused on the use main SIP messages to initiate and terminate the calls also the voice packets loss in the network was analysed. In order to simulate and analyse the SIP network, the network simulator2 program with the network animator was used and two simulation scenarios applied to it to explain how sessions are created and terminated. Also XGRAPH program was used to analyse packet loss with time in the network