政大機構典藏-National Chengchi University Institutional Repository(NCCUR):Item 140.119/61200

政大機構典藏-National Chengchi University Institutional Repository(NCCUR):Item 140.119/61200

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政大機構典藏 > 資訊學院 > 資訊科學系 > 學位論文 > Item 140.119/61200

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題名:	利用多層編碼配合DCCP形成與TCP友善的網路電話 TCP-friendly VoIP By Scalable Codec over DCCP
作者:	林耿誠
貢獻者:	連耀南林耿誠
關鍵詞:	壅塞控制機制網路電話語音編解碼器 DCCP
日期:	2012
上傳時間:	2013-10-01 13:47:09 (UTC+8)
摘要:	壅塞控制是網路管理的重大問題。目前的網路應用程式多半使用TCP或UDP這兩個傳輸協議來傳遞資料。TCP具有壅塞控制機制可以隨著網路狀況調整資料傳送速率，但其重送機制所導致的時間延遲不利於時效性的網路服務。而UDP的傳輸速率多半是在傳輸之前先行設定，在資料的傳輸過程中不再改變，對於網路壅塞並無任何調節作用，不利於網路之和諧共用。因此DCCP 這種具有壅塞控制機制的不可靠傳輸協議被提出，期望取代UDP 成為不可靠傳輸的主流協議。網路電話大部分使用UDP作為傳輸層協定，UDP不具壅塞控制機制，有礙於網路之和諧共用，而且網路的品質也會因為網路壅塞而遺失封包導致品質受損，要達成網路電話在網路壅塞時有壅塞控制能力，且能有好的通話品質，必須要根據網路的狀況調整封包傳送速度或是封包大小。本研究利用多層編碼配合DCCP形成與TCP友善的網路電話。壅塞控制必須針對網路中不同程度的壅塞作出適當程度的反應，因此我們改進語音編解碼器，設計出多層語音編碼並設計出可以與之配合的DCCP，讓DCCP依照網路的狀況送出不同層級的語音封包。本研究透過實際網路的實驗環境中評估以CBR over UDP、Flexible Bit-Rate以及Scalable Codec三種方式傳輸網路電話封包的效能。並也評估Scalable Codec VoIP與TCP同時存在於頻寬不足的網路中，對於頻寬競爭能力的表現。實驗結果顯示在網路壅塞最嚴重的情況下可以和一般的CBR配合UDP的方法比較封包遺失率達到40%左右的改善，語音品質評估指標MOS達到1.5分的改善，與Flexible Bit-Rate方法比較封包遺失率也達到8%以上的改善，MOS達到1分的改善，讓整體網路狀況更為穩定，並因為封包遺失的減少讓語音品質提升。而在和TCP頻寬競爭實驗中可看出，在頻寬不足的情況下，此研究提出的方法可以和TCP友善且公平的競爭頻寬。 Congestion control is one of the major problems of network management. Most current network applications use either TCP or UDP to transport data. TCP is equipped with a congestion control mechanism but is not suitable for real-time multimedia applications due to its instability of delay time. On the other hand, UDP fixes its data rate and doesn`t change it during the period of transmission even when the network is congested. Under this circumstance, DCCP, which is an unreliable transport protocol but has a congestion control mechanism, is proposed to replace UDP to support real-time network applications such as VoIP. Our previous study showed that a flexible bit rate CODEC to support VoIP over DCCP can effectively control network congestions while maintaining a good voice quality. However, it has an implementation issue yet to be addressed: it requires a bidirectional interaction between DCCP and CODEC. This thesis proposes to use a scalable CODEC approach to support flexible bit rate VoIP over DCCP. The CODEC sends the entire spectrum of input voice stream to DCCP. DCCP then selects the appropriate voice activation level to compose output stream according to the measured network status, which is feedbacked from the receiver side. The interaction between DCCP and CODEC is avoid. The proposed scheme was evaluated in a real local area network against two other protocols under various VoIP environments, CBR over UDP and Flexible Bit-Rate. Experimental results show that the proposed scheme can outperform CBR over VoIP in the most serious network congestion (under our lab configuration) by 40% in packet loss rate and 1.5 in MOS. It can outperform Flexible Bit-rate over VoIP by 8% in packet loss rate and 1.0 in MOS. Finally,the fairness test shows that our scheme can coexist with TCP with a fairness index higher than 95% even when network is congested.
參考文獻:	[1] C. Albuquerque, B.J. Vickers, and T. Suda, "Network border patrol: Preventing congestion collapse and promoting fairness in the Internet," IEEE-ACM Transactions on Networking, vol. 12, no. 4, Dec. 2004, pp. 173-186. [2] S. Andersen and A. Duric, “Internet Low Bit Rate Codec (iLBC), “ IETF RFC 3951, Dec. 2004. [3] L. S. Brakmo, S. W. O’Malley, and Larry L. Peterson. "TCP Vegas: New Techniques for Congestion Detection and Avoidance," ACM SIGCOMM, Aug. 1994, pp. 24-35. [4] Ming Cao, "EVCCM: An Efficient VOIP Congestion Control Mechanism," IEEE GLOBECOM Workshops, 2008 [5] D. Chiu and R. Jain, "Analysis of the Increase and Decrease Algorithms for Congestion Avoidance in Computer Networks," Computer Networks and ISDN Systems, vol.1, 1989, pp. 1-14. [6] A. Falk, D. Katabi, Y. Pryadkin," Specification for the Explicit Control Protocol (XCP)," draft-falk-xcp-03.txt (work in progress), Jul. 2007. [7] K. Fall and S. Floyd, "Simulation-based Comparisons of Tahoe, Reno, and SACK TCP," ACM Computer Communication Review, vol. 26, no.3, 1996, pp. 5-21. [8] S. Floyd and T. Henderson, "The NewReno Modification to TCP`s Fast Recovery Algorithm," IETF RFC 2582, 1999. [9] S. Floyd and E. Kohler, "Profile for DCCP Congestion Control ID 2: TCP-like Congestion Control," IETF draft-ietf-dccp-ccid2-08, http://www.ietf.org/internet-drafts/draft-ietfdccp-ccid2-10.txt, Mar. 2005. [10] S. Floyd, E. Kohler, and J. Padhye, "Profile for DCCP Congestion Control ID 3: TFRC Congestion Control," IETF draft-ietf-dccp-ccid3-11, http://www.ietf.org/internetdrafts/draft-ietf-dccp-ccid3-11.txt, Mar. 2005. [11] Z. Fu, P. Zerfos, H. Luo, S. Lu, L. Zhang, and M. Gerla, "The impact of multihop wireless channel on TCP throughput and loss," IEEE INFOCOM, San Francisco, CA, Mar. 2003. [12] M. Handley, "TCP Friendly Rate Control (TFRC):Protocol Specification," IETF RFC 3448, Jan. 2003. [13] ITU-T Rec. P.800, "Methods for Subjective Determination of Transmission Quality," Aug. 1996. [14] ITU-T Rec. G.107, "The E-Model, A Computational Model for Use in Transmission Planning, " May 2000. [15] ITU-T Rec. G.723.1, "Dual Rate Speech Coder for Multimedia Communications Transmitting at 5.3 and 6.3 kbit /s," May 2006. [16] ITU-T Rec. G.729, ”Coding of speech at 8 kbit/s using Conjugate-structure Algebraic-code Excited Linear-prediction (CS-ACELP),” Jun. 2012. [17] V. Jacobson, "Congestion Avoidance and Control," ACM SIGCOMM, Aug. 1988, pp. 314-329. [18] E. Kohler, M. Handley and S. Floyd, "Datagram Congestion Control Protocol (DCCP)," IETF RFC 4340, Mar. 2006. [19] Eddie Kohler, Mark Handley, and Sally Floyd, "Designing DCCP: Congestion Control Without Reliability," SIGCOMM 06, Sep. 2006, Pisa, Italy, pp. 27-38. [20] Y.N. Lien, Y.C. Ding, "Congestion Control Enabled VoIP by Flexible Bit-rate" M.A. thesis, University of Chengchi, Taiwan, 2010 [21] De Cicco, L. and S. Mascolo, "A Mathematical Model of the Skype VoIP Congestion Control Algorithm," IEEE Transactions on Automatic Control, Mar. 2010 [22] J.Nagle, "Congestion Control in TCP/IP," RFC896, Jan. 1984. [23] K. Nahm, A. Helmy, and C.-C J. Kuo, "TCP over Multihop 802.11 Networks: Issues and Performance Enhancement," ACM MobiHoc 05, Urbana-Champaign, Illinois, USA, May 2005. [24] F. Sabrina and J.-M. Valin,"Adaptive Rate Control for Aggregated VoIP Traffic," GLOBECOM 2008, pp. 1405-1410. [25] W. Stevens, "TCP Slow Start, Congestion Avoidance, Fast Retransmit, and Fast Recovery Algorithms," IETF RFC 2001, 1997. [26] Q.C. Wu, J.Y Wang, "Porting VoIP applications to DCCP", in Proc. of the International Conference on Mobile Technology, Applications, and Systems Article no. 8,2008 [27] Xiph.org Foundation, "The Speex Codec Manual Version 1.2 Beta 3", 2007.
描述:	碩士國立政治大學資訊科學學系 100753011 101
資料來源:	http://thesis.lib.nccu.edu.tw/record/#G0100753011
資料類型:	thesis
顯示於類別:	[資訊科學系] 學位論文

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