Internet telephony

Wireless Internet access has recently gained significant attention as wireless/mobile communications and networking become widespread. Voice over IP service is likely to play a key role in the convergence of IP based Internet and mobile cellular networks. The mobility management performance for Mobile IP and Session Initiation Protocol is the focus of this thesis. After illustrating the operation of the protocols, the discrete event simulator, Network Simulator 2 (ns2), is used to compare the performance of the two protocols. The comparison of the protocols is done by comparing average end-to-end delay and the ratio of the number of packets received to the number of packets originally sent (Packet Delivery Fraction). The impact of mobility is analyzed by studying the performance of the protocols, for various mobility scenarios. The effect of an increase in the number of nodes and increase in velocity of the mobile node on the performance of the Mobile IP and SIP is compared. The performance of the Mobile IP and SIP is compared by measuring the performance metrics of the two protocols for similar simulations. The results obtained as a result of the simulations leads us to some interesting conclusions.

This thesis proposes a design for a signaling gateway to address the issues of interoperability between H.323 and SIP. The two IP Telephony standards currently compete for the dominance of IP telephony protocols. We investigate and study these two protocols in terms of interoperability. The question central to this thesis is the issue of interoperability between H.323 and SIP. A signaling gateway model is proposed and designed to address and solve this issue. The proposed model includes Call Initialization, Call Setup and Control, Capability Exchange and Call Termination, which are the four fundamental features supported by H.323 and SIP for establishing calls. Furthermore, we design the four internal components, which are common to the four main functions mentioned above. These four components are type checking, decomposition, conversion and reformatting. In addition, we illustrate in six different cases the functionality of the proposed signaling gateway for establishing calls between H.323 and SIP. These six cases also demonstrate the conversion capability of the proposed signaling gateway during a call between H.323 and SIP endpoints.