在此篇論文中,我們提出一個基於SIP的VoIP 服務架構,並使得WLAN 和GPRS達到無縫式網路漫遊,並描述 WLAN對WLAN 及WLAN對GPRS中handoff的程序。 在此架構中我們結合了第三層與第四層handoff的機制,其目的是為了減少封包遺失和點對點的延遲時間,並能夠使jitter保持在一定範圍中。 在第三層中,使用基地台所發出訊號的強度來決定handoff的時機且在第四層中我們使用SIP來完成用戶端的移動能力。 因此,我們測量handoff所發費的時間,並使用Session Controller和RTP Proxy作為在handoff 過程裡的媒介通道,並針對於handoff中多媒體串流降低其jitter及packet loss在可容許的範圍內。在mid-call的handoff中,我們減少了SIP UA偵測IP address 的改變,減少總handoff 過程的時間。 而且,Session Controller和RTP Proxy中利用緩衝區減少handoff 的延遲時間。 最後使用SIP UA測量handoff 延遲及RTP packet sequence number。
In this paper, we propose a SIP based architecture that supports roaming for WLAN and GPRS seamless network from the perspective of VoIP services and describes in handoff procedures for WLAN-to-WLAN and WLAN-to-GPRS. This architecture combines layer 3 and 4 for handoff process, in order to decrease packet loss and the end-to-end delay, jitter is kept under control. In Layer 3, Received Signal Strength (RRS) used for handoff detection and Session Initial Protocol (SIP) supports terminal mobility in Layer 4. Therefore, we measure the total handoff time. We used Session Controller and RTP Proxy as media gateway in handoff process. The goal is reducing the handoff latency and packet loss to an acceptable level for media streaming. In our performance study we focus on optimizing the handoff process time and RTP packet number. In SIP mid-call handoff, our propose scheme ignored process of SIP handoff detection and decrease the time of total handoff process. Moreover, Session Controller and RTP Proxy supported the RTP packets buffer for handoff delay time. We have measured the handoff delay with SIP terminal.