無線網路整合環境可以提供無線網路語音服務無所不在且連續不斷的通訊。然而,在這樣的環境下,無線網路語音服務主要面臨的問題就是語音封包遭受不同的傳輸延遲、高封包遺失率(Packet Loss Rate)和換手的問題(Handoff)。 本論文提出AQP(Adaptive QoS Playout) 演算法來提供高品質無線網路語音服務。AQP以語音品質評估模組E-model為調整基準,同時整合播放排程(Playout)控制演算法、重傳(Retransmission)機制和無線換手資訊來達到不中斷的高品質通訊。模擬結果顯AQP可以在高無線訊框錯誤率(Frame Error Rate)和低端點對端點(End-to-End)傳輸的環境下延遲降低封包遺失率並且改善語音通訊品質,並且在發生換手時也可以達到高品質的無線網路語音服務。
Integrated environment of wireless network can provide a ubiquitous and continuous communication for voice over Internet Protocol (VoIP) service. However, the major problems, which VoIP performs over wireless networks, are varying delay of each packet, high packet loss rate and handoff. This thesis proposed an Adaptive QoS Playout (AQP) algorithm to offer a high quality wireless VoIP service. AQP integrates the effect of dejitter buffer control, retransmission, and handoff delay based on the E-model. Simulation results show that AQP can reduce the lateness loss rate and improve speech quality under high frame error rate when the wireless link has small end-to-end network delay and also performs well when a handoff occurs.