An Experimental Study to Analyze SIP Traffic over LAN

VoIP (Voice over Internet Protocol) service has become famous now days due to their affordability and flexibility. VoIP networks use IP (Internet Protocol) phone to communicate over Internet or LAN (Local Area Network). Most IP phone use SIP (Session Initiation Protocol) for communication. The SIP (Session Initiation Protocol) was invented to help RTP (Real-Time Transport Protocol) in order to find destination IP address and port address over Internet. RTP use to transmit voice data between source and destination using their IP addresses and Port numbers. NAT (Network Address Translation) is a technique, which allows users to use multiple IP address internally for multiple devices to share one Internet connection. VoIP packets are routed over public Internet, which is not a secure platform. Sometimes data face delay problem due to network congestion. Many type of security and QoS (Quality of Service) techniques and algorithms are being designed to overcome these problems. In this paper we have experimentally analyzed and examine the bit rate and packet rate during VoIP call process. This paper will show the registration process of SIP and delay in arrival packet due to congestion on network. We have used the WireShark, packet analyzer software to observe the SIP registration and call setup process in LAN (Local Area Network) as well as inter-packet arrival times of RTP frames in the caller to callee direction.