近來關於無線區域網路的研究，逐漸將其觸角延伸至網路電話在網路封包易受損之無線網路的應用議題上。在本研究當中，我們利用網路模擬方法，分別針對有線對無線及無線對無線等兩種網路架構，當無線網路中網路電話及資料傳輸同時存在的前提之下，評估網路電話之語音品質受到資料傳輸之影響。在語音模型部分，將採用發話受話間隔皆是遵從指數分配之模型。在資料模型當中，主要是用TCP傳送FTP封包。此外，無線傳輸包括封包損失狀況，封包損失模型則是採用兩狀態(好／壞)間轉換的模式。

　　在本研究當中，我們假設網路電話使用不同的語音解編碼技術,如G.711、G.723.1、G.729A以及iLBC，而不同語音解編碼技術通常會對應至各自不同傳輸速率及封包大小，也因此對服務品質產生不同程度的影響。在語音衡量的指標上，我們採用由國際電信組織ITU-T所制定與建議的平均意見分數(MOS)法與E-Model。前者是透過多位受話測試者聆聽些許聲音片段下，給予主觀的評分，以用來鑑別語音品質的好壞。後者則是透過計算在語音傳輸過程中若干因素對聲音品質的負面影響綜合參數，來評估該通話之品質水準。在標準802.11b無線網路的環境之下，我們將經由上述兩種語音評估指標，來求知不同語音解編碼技術之最大通話數。根據模擬結果，我們發現特別是在下鏈傳輸中，過長的時間延遲以及大量封包的碰撞，會導致語音品質與最大通話數大幅地下滑，此種情況在上鏈就較不明顯。

　Recent studies in the wireless LANs deployment have generated a lot of interest in making Voice over IP (VoIP) available over error-prone wireless networks. In this thesis, we evaluate the quality of speech over not only a wired-cum-wireless scenario but also a wireless-cum-wireless scenario in the presence of both voice and data flows. Voice traffic is modeled by a silence period and a spurt period both with exponential distribution. Data model is generated by FTP based on TCP, and error model depends on the two-state (Good/Bad) error model.

　In this thesis, we assume that VoIP users use some kinds of codecs, such as G.711/G.723.1/G.729A/iLBC, in signal sampling process. Each codec has specific data rate and packet size. In order to measure voice quality, we select mean opinion score (MOS) and E-model recommended by International Telecommunication Union Telecommunication Standardization Sector (ITU-T). In the MOS model, the users who have listened to some sample voice segments give subjective scores for speech quality. Then, the scores are averaged to get the value of MOS. In the E-model, the evaluation of call quality is calculated from some key parameters including several negative factors, such as delay and packet loss. Based on the two evaluation methods, we study the influence of data traffic on the quality of VoIP over wireless LANs. From our simulation results, we find that especially for downlink transmission, longer delay and plenty of collisions will highly degrade voice quality and drastically decrease the maximum allowable number of calls.

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