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.

第一章: 緒論..............................................1

1.1研究背景...............................................1
1.2研究動機...............................................2
1.3研究方法...............................................2
1.4論文架構...............................................3

第二章:文獻探討...........................................4

2.1音頻基本原理...........................................4
2.2VoIP相關協定...........................................6
2.3VoIP在異質網路的運作..................................16

第三章:研究系統架構......................................18

3.1測試主架構(PESQ&JITTER)...............................18
3.2回音測試子架構........................................19
3.3測試考量因素..........................................20

第四章: 模擬與分析.......................................22

4.1語音品質感知評估與抖動測試參數設定....................22
4.2回音測試參數設定......................................34
4.3結果分析..............................................38

第五章: 研究成果與展望...................................39

參考文獻.................................................40

圖目錄
圖2.1:H.323 Protocol 方塊圖...............................6
圖2.2:H.248 Protocol相關元件示意圖........................8
圖2.3:SIP 呼叫流程........................................9
圖2.4:SDP內容............................................10
圖2.5:G.131..............................................11
圖2.6:RTP標頭............................................13
圖2.7:NGN異質網路示意圖..................................17
圖3.1:測試主架構.........................................18
圖3.2:不同位置Return Loss................................19
圖3.3:測試子架構.........................................19
圖4.1:Voice封包範例......................................22
圖4.2:Call behavior 設定示意圖...........................23
圖4.3:雜訊能量頻譜圖.....................................27
圖4.4:Traffic Priority 設定圖............................28
圖4.5:AG-1 &PSTN PESQ圖..................................29
圖4.6:AG-1&AG-2 PESQ圖...................................30
圖4.7:AG-2&PSTN PESQ圖...................................30
圖4.8:典型FIR............................................32
圖4.9:回音tail長度示意圖.................................32
圖4.10:AG-1與PSTN PESQ圖................................33 
圖4.11:AG-1與AG-2 PESQ圖.................................34
圖4.12:話機交叉測試示意圖................................34
圖4.13:FM002 round trip delay圖..........................36
圖4.14:TA-205 round trip delay圖.........................36
圖4.15:TA-205 Return Loss圖..............................37

表目錄
表4.1:Codec 頻寬使用表...................................23
表4.2:通話行為設定(final)................................24
表4.3:通話行為設定(干擾設定).............................24
表4.4:通話行為設定(PESQ設定).............................25
表4.5:通話行為設定(Dail設定).............................25
表4.6:通話行為設定(Ring設定).............................26
表4.7:通話行為設定(通話間隔設定).........................26
表4.8:AG-1與PSTN實驗數據結果表...........................29
表4.9:AG-1與AG-2實驗數據結果表...........................29
表4.10:AG-2與PSTN實驗數據結果表..........................30
表4.11:AG-1與PSTN實驗數據結果表..........................33
表4.12:AG-1與AG-2實驗數據結果表..........................34
表4.13:各型話機實驗數據結果表-1..........................35
表4.14:各型話機實驗數據結果表-2..........................35

[1] S. Ganguly and S. Bhatnagar, “VoIP: Wireless, P2P and New Enterprise Voice over IP”, Wiley, 2010.
[2] B. Douskalis, “Putting VoIP to Work: Softswitch Network Design and Testing”, Prentice Hall, 2008.
[3] 香港環境保護署, http://www.epd.gov.hk/epd/noise_education/web/
CHI_EPD_HTML/m1/intro_5.html 
[4] 維基百科http://zh.wikipedia.org/zh-tw/會話發起協定
[5] 黃松柏, ”Provisioning of VOIP Service Based on Multiplexing and De-Multiplexing Scheme in Mobile Ad Hoc Access Networks, ”國立台灣科技大學碩士論文, 2007
[6] Cisco網站, http://www.cisco.com/web/TW/about/news_20041122_2.html 
[7] http://dslab.ee.ncku.edu.tw/~lily/learning/learning_ch2-2.html, 成功大學
分散系統實驗室
[8] 王小川, ”語音訊號處理”, 全華圖書公司, 2009.
[9] http://andrew.csie.ncyu.edu.tw/pdf3/050519-2.pdf, 嘉義大學
[10] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, ”RTP: A
Transport Protocol for Real-Time Applications”, RFC 1889, 1996.
[11] ITU-T Recommendation P.862 Perceptual Estimation of Speech Quality (PESQ)
[12] 胡竹生, “DSP晶片原理與應用”, 全華圖書公司, 2002.
[13] 劉云, “數位通訊系統”, 新文京開發出版(股)公司, 2006.
[14] 鍾隆宇, “數位通訊原理與應用”, 全華圖書公司, 2007.
[15] F. Ohrtman, “Softswitch : Architecture for VoIP”, McGraw-Hill , 2009.
[16]http://vaplab.ee.ncu.edu.tw/chinese/pcchang/course2005b/comsp/audio/lgorithm.htm
[17]http://www.adaptivedigital.com/product/echo_cancel/g168_echo_canceller.htm 
[18] 維基百科 http://en.wikipedia.org/wiki/H.323
[19] 周勝鄰, “無線與頻寬整合之多媒體通訊技術,”工研院, Nov. 2003
[20] W.-E. Chen, H.-N. Hung and Y.-B. Lin, " Modeling VoIP Call Holding Times for Telecommunications," IEEE Network, November/December. 2007, pp. 22-28.
[21] S. Ganguly, V. Navda, K. Kim, A. Kashyap, D. Niculescu, R. Izmailov, S. Hong, and S. R. Das, ”Performance Optimizations for Deploying VoIP Services in Mesh Networks,” IEEE Journal on Selected Areas in Communications, vol. 24, no. 11, Nov. 2006, pp. 2147-2158.
[22] C.-Y. Lee, H.-K. Kim, K.-H. Ko, J.-W. Kim, H.-C. Jeong, “A Study on Structure for Monitoring and Detecting VoIP Abnormal Traffic,” Second International Conference on Future Generation Communication and Networking, 2008, pp. 178-182.
[23] H. M. Chong and H. S. Matthews, “Comparative Analysis of Traditional Telephone and Voice-over-Internet Protocol (VoIP) Systems,” International Symposium on Electronics and the Environment, 2004, pp. 106-111.
[24] G. Zhang, M. Hillenbrand, and P. Müller, "Facilitating the Interoperability among Different VoIP Protocols with VoIP Web Services, " Proceedings of the First International Conference on Distributed Frameworks for Multimedia Applications, 2005