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研究生:曾國坤
研究生(外文):Kuo-Kun Tseng
論文名稱:網路電話之智慧型漸進式撥放緩衝區演算法
論文名稱(外文):Codec and Duplex Aware Adaptive Playout Algorithms for VoIP Systems
指導教授:林盈達林盈達引用關係
指導教授(外文):Ying-Dar Lin
學位類別:碩士
校院名稱:國立交通大學
系所名稱:電資學院學程碩士班
學門:工程學門
學類:電資工程學類
論文種類:學術論文
論文出版年:2002
畢業學年度:90
語文別:中文
論文頁數:33
中文關鍵詞:撥放緩充區演算法網路延遲差異網路電話聲音品質衡量
外文關鍵詞:Playout AlgorithmNetwork JitterVoice over IPSpeech Quality Measurement
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網路的語音傳輸會受到網路的延遲差異(Jitter)的影響,在接收端控制撥放延遲(Playout Delay)可以解決Jitter 及封包過多所造成的遺失(Overflow Packet Loss)的問題。 傳統的漸進式緩充區演算法(Conventional Adaptive Playout Algorithms) 只是用過去的網路延遲差異去預測撥放延遲。 他們並沒有利用語音壓解器(Codec),及單雙工工作模式(Duplex Mode)的特性去最佳化撥放延遲。 所以我們提出兩種方法: 智慧型漸進知壓解器 (Codec Aware Adaptive Playout, CAAP)方法及智慧型漸進式單雙工模式(Duplex Aware Playout Adaptation, DAPA)方法。CAAP又可以細分為:簡單智慧型漸進式壓解器 (Simple CAAP, SCAAP) 方法和高級智慧型漸進式壓解器 (Deluxe CAAP, DCAAP) 方法。 兩者的設計觀念是,假設每一種壓解器對封包遺失(Packet Loss)有不同的容忍程度,低壓縮比的壓解器可以增加更多的撥放延遲及減少封包遺失。 SCAAP是用單一預測撥放延遲,但DCAAP用多預測撥放延遲去設計,所以DCAAP有較好的效果。而DAPA的設計觀念為,在單工模式(Half Duplex Mode)下可以增加撥放延遲去減少封包遺失的問題。 因為用不同的觀念,CAAP及DAPA可以單獨或一起工作,且它們的改善也是累加的。在我們語音效能衡量中,CAAP及DAPA比起平均延遲及差異(Mean Delay and Variance, MDV) ,可以有10%到16%的改善。 對於同步的雙方通話(Two-way Communication)因為目前沒有客觀的語音品質衡量 (Objective Speech Quality Measurement) 方法,所以我們同時提出一套模型化的衡量 (Model-based Measurement) 方法。

Internet voice transmission suffers from variance of network delay, called — jitter. In VoIP receiver, the playout algorithm can control the playout delay to eliminate the jitter and minimize the overflow packet loss. Conventional adaptive playout algorithms estimate playout delay from previous network jitters only. They are not aware of voice codec type, and communication duplex mode. Thus, we present two approaches - Codec Aware Adaptive Playout (CAAP) and Duplex Aware Playout Adaptation (DAPA). The CAAP has two sub-algorithms, Simple CAAP (SCAAP) and Deluxe CAAP (DCAAP). Both design concepts are that the different codec has different tolerance to packet loss, the playout delay using a low compression ratio codec can be increased more to reduce packet loss. SCAAP uses single estimated playout delay and DCAAP uses multiple estimated playout delays, consequently DCAAP has better performance. As regards the DAPA design concept, it increases more playout delay to reduce the packet loss in half duplex mode. Because of different design concepts, CAAP and DAPA can work either alone or together; and their improvements are accumulative. In our performance evaluation of speech quality, CAAP and DAPA outperform the Mean Delay and Variance (MDV) algorithm by 10% to 16%. No objective mechanisms for measuring the speech quality of two-way communication exist; a model-based measurement mechanism is also proposed.

摘要 I
ABSTRACT II
CONTENTS III
LIST OF FIGURES IV
LIST OF TABLES V
1. INTRODUCTION 1
2. ARCHITECTURE AND ALGORITHMS 4
2.1 NOTATION 4
2.2 CAAP 6
2.2.1 Simple CAAP (SCAAP) 7
2.2.2 Deluxe CAAP (DCAAP) 8
2.3 DAPA 11
3. EVALUATION METHODOLOGY 12
3.1 THE COMPARED ALGORITHMS 12
3.2 MODEL-BASED MEASUREMENT 13
3.2.1 Continue MOS Modeling 13
3.2.2 Discrete MOS Modeling 14
3.3 EMULATION ARCHITECTURE 15
3.4 CAAP CONFIGURATION 16
3.5 DAPA CONFIGURATION 16
4. RESULT AND ANALYSIS 17
4.1 CAAP RESULTS AND ANALYSIS 17
4.1.1 CAAP Simulation Result and Analysis 17
4.1.2 CAAP Emulation Result and Analysis 20
4.2 DAPA RESULTS AND ANALYSIS 21
4.2.1 Loss Rate and Playout Delay Analysis 21
4.2.2 DAPA Results 22
5. CONCLUSION 24
6. REFERENCES 26

[1] R. Ramjee, J. Kurose, D. Towsley, and H. Schulzrinne, “Adaptive Playout Mechanisms for Packetized Audio Applications in Wide-Area Networks”, Proceedings of IEEE INFOCOM, Toronto, Canada, pp. 680 - 686, June 1994.
[2] Marco Roccetti, Vittorio Ghini, Giovanni Pau, Paola Salomoni and Maria Elena Bonfigli, “Design and Experimental Evaluation of an Adaptive Playout Delay Control Mechanism for Packetized Audio for use over the Internet”, November 1998.
[3] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", RFC 1889, January 1996.
[4] Jesus Pinto and Kenneth J. Christensen, “An Algorithm for Playout of Packet Voice based on Adaptive Adjustment of Talkspurt Silence Periods”, 1999, http://citeseer.nj.nec.com/pinto99algorithm.html.
[5] Phillip DeLeon and Cormac J. Sreenan, “An Adaptive Predictor for Media Playout buffering”, Stanford, March, 2001, http://citeseer.nj.nec.com/deleon99adaptive.html.
[6] ITU-T P.861 (1996): "Objective quality measurement of telephone-band (300 - 3 400 Hz) speech codecs“,02.1998.
[7] ETSI EG 201 377-1 V1.1.1, “Specification and measurement of speech transmission quality”, 02.2001.
[8] ITU-T G.862, “Perceptual evaluation of speech quality (PESQ), an objective method for end-to-end speech quality assessment of narrow band telephone networks and speech codecs”, 02.2001.
[9] ETSI, TR101 329 v1.2.5, “General aspects of Quality of Service(QoS)”, 10.1998.
[10] Les Cottrell, Stanford Linear Accelerator Center (SLAC), “QoS on Best-Effort Networks”, Presented at April 25-27,2001, http://www.slac.stanford.edu/grp/scs/net/talk/qos-itu-apr01/.
[11] ITU-T G.107 (05/00): “The E Model, A Computational Model for use in Transmission Planning”.
[12] ITU-T G.113, “Transmission Impairments”, 02.1996.
[13] ITU-T G.114, “One-way transmission time”, 05.2000.
[14] ETSI TR41.4-01-02-005, “Passive Monitoring for Voice over IP Gateways”, Costa Mesa, CA, February 2001.
[15] Rosenberg, J.; Lili Qiu; Schulzrinne, H. , “Integrating packet FEC into adaptive voice playout buffer algorithms on the Internet”, INFOCOM 2000. Nineteenth Annual Joint Conference of the IEEE Computer and Communications Societies. Proceedings. IEEE , Volume: 3 , 2000 Page(s): 1705 -1714 vol.3.
[16] Padhye, C.; Christensen, K.J.; Moreno, W. , “A new adaptive FEC loss control algorithm for voice over IP applications” Performance, Computing, and Communications Conference, 2000. IPCCC '00. Conference Proceeding of the IEEE International, 2000 Page(s): 307 —313.
[17] Sanneck, H., “Concealment of lost speech packets using adaptive packetization “, Multimedia Computing and Systems, 1998. Proceedings. IEEE International Conference on , 1998, Page(s): 140 -149.
[18] Abreu-Sernandez, V.; Garcia-Mateo, C. , “Adaptive multi-rate speech coder for VoIP transmission”, Electronics Letters , Volume: 36 Issue: 23 , 9 Nov. 2000 Page(s): 1978 —1980.

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