Investigating latency in two ways communication of VOIP in WLAN

Voice over Internet Protocol (VoIP) has stringent delay requirement. The quality of VoIP does not yet match the quality of a circuit-switched telephone network due to several challenges such as available bandwidth, delay or network latency, packet loss, jitter, echo, security and reliability. International Telecommunication Union – Telecommunications Standards Sector (ITU-T) recommends a certain maximum delay which tolerable to the user of VoIP service. To improve the performance of VoIP during handoff, handoff latency must be kept as low as possible. There are numerous method proposed to improve the performance of real time application such as VoIP during handoff. One of the methods is Mobile IP, which provides mobility management in the network layer of TCP/IP protocol. In this thesis, the operations and handoff procedure for standard Mobile IPv6 (MIPv6) are studied in detail. Fast Handovers for MIPv6 (FMIPv6) which was introduced as an extension of MIPv6 is used to improve the performance of VoIP application. In this project, the implementation of MIPv6 and FMIPv6 are modeled and simulated using Network Simulator 2 (NS2). Three different voice coding schemes namely G.711, G.723.1 and G.729 are studied to determine their effects on the handoff latency and other Quality of Service (QoS) parameters such as packet loss, throughput and delay.