摘要：To initiate voice, image, instant messaging and general multimedia communication, the Session communication must initiate between two participants. SIP (Session initiation protocol) is an application layer control, which task is creating management, and terminating this kind of Sessions. With regard to the independence of SIP from the Transport layer protocols, the SIP messages can be transferred on a variety of Transport layer protocols such as TCP or UDP. The mechanism of Retransmission, which has been embedded in SIP, is able to compensate the missing Packet loss, if needed. The application of this mechanism is when SIP messages are transmitted on an unreliable transmission layer protocol such as UDP. This mechanism, while facing with SIP proxy with overload, causes excessive filling of proxy queue, delays the increase of other contacts and adds the amount of the proxy overload. We in this article, while using UDP, as the Transport layer protocol, by regulating the Invite Retransmission Timer appropriately (T1), have improved the SIP functionality. Therefore, by proposing an Adaptive Timer of Invite message retransmission, we have tried to improve the time of Session initiation and as a result, improving the performance. The performance of the proposed SIP, by the SIPP software in a real network environment has been implemented and evaluated and its accuracy and performance has been demonstrated.
