隨著網路技術的快速發展及網路電話整合技術的進步，相對的應用也日益增多。近年來Voice over Internet protocol (VoIP)技術漸趨成熟，在未來Internet protocol (IP)網路上的整合，開始朝向語音/視訊/資料/傳真一體化的趨勢，並且與傳統公眾電話網路電話服務結合；利用閘道器來將傳統電話部分轉送到IP網路上，以結合下世代的電信服務。這種傳輸機制，除了能服務即時語音的傳輸，更可避免傳統電話因佔線所造成的閒置浪費。然而目前網路電話常會面臨的一個重要課題是通話品質的監測與評估，由於網路頻寬是共用分享的，而在語音封包與一般網路流量互相競爭使用固定頻寬的情況下，整個網路電話通話路徑是否能達到通話品質的要求便很難去瞭解及評估。因此，本論文基於VoIP技術基礎，利用語音感知評量標準衡量通話品質並針對接取閘道器進行以下幾點改善:第一透過調整接取閘道器系統的訊務封包順序、第二使用獨立語音通道的虛擬區域網標籤、第三實測市售各類型話機以獲得回音的參考依據、第四增加有限脈衝響應階數，因而改善VoIP的網路效能。 The rapid development of technology has facilitated the integration of computers and telephones in recent years. The related voice over Internet protocol (VoIP) service will be imperative to a greater extent in the Internet market. The new trend is to replace the conventional telecommunication exchange network by the convenient VoIP network, which integrates the services of voice, video, data, Fax, local area network, wireless area network, and answering machine with the gateway. The VoIP service can economically finish the voice communications in time. Moreover, it can efficiently utilize the network bandwidth and, prevent the unnecessary idle of the bandwidth in the conventional communication networks. The maintenance of the communication qualities in the VoIP environment is a challenging task since the bandwidths are shared by many Internet users. It is difficult to evaluate that the current routes used by VoIP service can achieve the required quality or not. In this thesis, we propose a method to evaluate the communication quality for VoIP networks by adjusting the related parameters.
