• شماره ركورد
    6557
  • شماره راهنما(اين فيلد مربوط به كارشناس ميباشد لطفا آن را خالي بگذاريد)
    6557
  • پديد آورنده

    محمد علي ميرصانع

  • عنوان
    بهبود كارايي پروتكل SIP بر اساس انتخاب مناسب پروتكلهاي شبكه
  • مقطع تحصيلي
    كارشناسي ارشد
  • رشته تحصيلي
    كامپيوتر-گرايش معماري كامپيوتر
  • سال تحصيل
    زمستان86
  • تاريخ دفاع
    زمستان86
  • استاد راهنما
    دكتر احمد اكبري
  • 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.