High Load Diminution by Regulating Timers in SIP Servers Ahmadreza Montazerolghaem 1 , Seyed-Amin Hosseini-Seno 2 , Mohammad Hossein Yaghmaee 3 and Rahmat Budiarto 4 1,2,3 Department of Computer Engineering, Ferdowsi University of Mashhad, Mashhad, Iran, 4 Dept. Of Computer Information System, Al Baha University, Kingdom of Saudi Arabia Ahmadreza.montazerolghaem@stu-mail.um.ac.ir, hosseini@um.ac.ir, h.yaghmaee@um.ac.ir, rahmat.budiarto@surya.ac.id Abstract—To start voice, image, instant messaging, and generally multimedia communication, session communication must begin between two participants. SIP (session initiation protocol) that is an application layer control induces management and terminates this kind of sessions. As far as the independence of SIP from transport layer protocols is concerned, SIP messages can be transferred on a variety of transport layer protocols including TCP or UDP. Mechanism of Retransmission that is embedded in SIP could compensate for the missing packet loss, in case of need. This mechanism is applied when SIP messages are transmitted on an unreliable transmission layer protocol like UDP. Also, while facing SIP proxy with overload, it could cause excessive filling of proxy queue, postpone increase of other contacts, and add to the amount of the proxy overload. In the present work, while using UDP as transport layer protocol, invite retransmission timer (T 1 ) was appropriately regulated and SIP functionality was improved. Therefore, by proposing an adaptive timer of invite message retransmission, attempts were made to improve the time of session initiation and consequently improve the performance. Performance of the proposed SIP was implemented and evaluated by SIP P software in a real network environment and its accuracy and performance were demonstrated. Keywords—Load Diminution, TCP, SIP, VoIP, Regulating Timers. I. INTRODUCTION In recent years, Internet-based networks can be seen everywhere. This issue has become a factor for popularizing dial-up calls via IP network. Some cases among these factors that have more welcomed such communication can be named as follows: the first case is about economic problems; Internet phone calls, especially for international calls, are much cheaper than typical phone calls. The next case is development of IP communication in a variety of applied equipment. Personal computers are easily connected to the Internet using different modes. Existence of IP in cell phones is another factor that could cause further development of the Internet than other technologies. Another factor is further use of packet switched architecture, instead of circuit switched one that can itself make optimum use of the resources possible. In simultaneous and two-way communication such as audio and video ones, file transfer, exchanging instant message, and generally multimedia sessions, in which communication is online, first, a session must initiate among the participants. The most important point about these types of communication, especially the Internet phone call contact, is signaling, which is responsible for the task of initiating and managing each session. One of the very suitable protocols in this field is SIP, the task of which is making, modifying, and terminating the session. One of the key components in delay-sensitive applicable programs like voice and image transmission by the Internet is the time required for the session start-up, which is highly effective in protocol efficiency. Reduction of this period of time leads to increase in the SIP server transmission and consequently improves its efficiency. On the other hand, it will cause more user acceptability [15]. Due to the importance of this issue, some researchers have focused on reducing the necessary time for creating the session in progress, some examples of which are mentioned below. In this article, to improve the session initiation time in UDP protocol using the appropriate regulation of timer and sending invite message retransmission, session initiation time was improved the number of missed calls was reduced. II. RELATED WORKS As certain servers which are specified for SIP can be used for a session, increasing their number, load distribution among them, or their processing capability can help reduce the time for session initiate [10]. It is worth knowing that these changes are costly and sometimes difficult. Another way is to apply stateless mode instead of stateful in servers [11]. However, it has a major problem; a complete history of communication will not be available [2]. Another method which can be used for reducing the time for session initiation is to remove users authentication confirmation, which has the highest positive impact on reducing the time of the session initiation; but, due to the nature of the Internet network and the necessity of using authentication confirmation in some cases, this method has some considerable problems as well. On the other hand, another important component, which must be considered, is the percentage of missed sessions, the reduction of which evidently influences the protocol. Generally, with regard to compatibility in interoperability, UDP is considered the standard transmission protocol for SIP; however, it should be considered that other protocols such as TCP and SCTP are also reliable and applicable [9] (in some cases, use of connection-oriented protocols are required: for example, in the circumstances in which the length of SIP message transmission is more than that if MTU [4]). In contrast, according to the standard, all the implementations of SIP must back-up TCP and UDP [10]. Accordingly, by appropriately selecting transmission protocol and UDP as transmission layer protocol and by regulating SIP Proceeding of International Conference on Electrical Engineering, Computer Science and Informatics (EECSI 2014), Yogyakarta, Indonesia, 20-21 August 2014 46 brought to you by CORE View metadata, citation and similar papers at core.ac.uk provided by Proceeding of the Electrical Engineering Computer Science and Informatics