چكيده
Abstract
Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol
for creating, modifying, and terminating sessions with one or more participants. These
sessions include Internet telephone calls, multimedia distribution, instant messaging
and so on. Before a user can start a VoIP session, the end-user terminal has to
establish the session using signaling protocols such as H.323 and session initiation
protocol (SIP) in order to negotiate media parameters. A lot of researches have been
done on VOIP quality but since SIP is a new protocol, its performance is not wellstudied
or understood. The time interval to perform the session setup is called the
session setup time (SST). It is one of the most important SIP end to end metrics and
can be affected by the Round Trip Delay (RTT) and packet Loss ratio. There are a lot
of approach to reduce SST since reducing it will improve protocol performance .One
way to do so is by choosing the appropriate retransmission timer and the underlying
protocols. In this project, has been decided to improve SIP performance based on
appropriate selection of network transport protocol. The SIP performance has been
evaluated in the simulated environment (by means of OPNET modeler) and real
environment (by means of SIPp test tool). The SIP session setup performances and
also call drop ratio evaluated with various underlying protocols (transport control
protocol (TCP), user datagram protocol (UDP)) as a function of the RTT and packet
loss. Also the effect of adjusting the retransmission timer T1 on SST and the number
of retransmitted messages (as a trade off) has been shown. At the end, a novel
adaptive retransmission timer T1 has been proposed to optimize SST and improve
protocol performance .By using SIPp test tool, the function of the proposed adaptive
retransmission timer and its effect on reducing SST has been shown.
