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研究生:吳晉愷
研究生(外文):Chin-Kai Wu
論文名稱:串流語音之延遲變異減緩與封包遺失減少演算法
論文名稱(外文):itter Smoothing and Packet Loss Reduction Algorithms for Streaming Voice
指導教授:王家祥
指導教授(外文):Jia-Shung Wang
學位類別:碩士
校院名稱:國立清華大學
系所名稱:資訊工程學系
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:41
中文關鍵詞:延遲變異封包遺失減少串流語音
外文關鍵詞:JitterPacket LossReductionStreamingVoice
相關次數:
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網際網路快速的成長與多媒體相關技術已經為我們生活的世界帶來戲劇性且基本的變化。近年來,高品質的即時串流影音與網際網路上影像的多媒體服務漸趨熱門;網際網路電話以及隨選視訊服務即為兩種典型。因此,隨著網路頻寬快速的成長,網際網路電話傳輸大量之壓縮語音封包儼然晉身熱門服務之一。網際網路電話成為大眾之所好不僅因為其架構之建設較為便宜;更因為其具有延伸性,未來可提供多種加值服務。
對於特定的串流服務,由於真實網路在頻寬與可靠性上的變異,提供有保證的服務品質來滿足即時性與緊迫的延遲需求是重要的。有保證的服務品質包含三種參數:封包傳輸延遲、封包延遲變異,以及封包遺失率。一旦語音封包從來源端送出,網路的變異將使得語音封包的狀況變的不可預測。因此有許多研究投注心力在提供有保證的服務品質方法上。
早期的研究傾向提供有保證的服務品質於來源端,或者是在配合互動式回饋機制之目的端。近期的研究則嘗試在傳送路徑之中間點(路由器、交換器)上達成目標。
本論文中提出兩種線上演算法,目的在中間點上作調整以提供有保證的服務品質。首先,延遲變異控制機制目標在低延遲代價條件下,減緩延遲變異。藉此使配置在目的地之緩衝區得以減少。其二,封包遺失減少演算法乃為減少連續封包遺失之發生頻率而設計。這將使得語音串流對於網路流量激增與壅塞的情況更具韌性,而可在客戶端有更高品質的表現。
The vast progresses in Internet and multimedia related technologies have brought dramatic and fundamental changes to the world that we live. Recently, the emerging high quality streaming audio, voice, and video for multimedia services on the Internet is progressively popular; voice over IP (VoIP) and video on demand (VOD) services are two typical examples. Thus, owing to the rapidly growing of bandwidth on IP networks, Internet-telephony that delivers a large amount of compressed voice IP packets to optimize the bandwidth utilization becomes one of the most favorite services. Internet-telephony is admired not only its cheaper to build the infrastructure, moreover, it can provide various value-added services in the future.
On certain streaming services, it is also significant to provide guaranteed QoS (quality of services) to satisfy the real-time and stringent delay requirement because the real networks are heterogeneous in bandwidth and reliability. Guaranteed QoS includes three basic parameters: packet transfer delay, packet delay variation and packet loss rate. Once the voice packets sent from the source end, the variation of networks makes the condition of voice packets unpredictable. As a result, there are many researches concerning providing guaranteed QoS schemes.
Early related works tend to provide QoS at the source end or the destination end with interactive feedback mechanism. Recent works, such as jitter earliest due date queueing (JEDD) and multilayer-gated frame queueing (MGFQ) tries to achieve the goal in the intermediate nodes.
In this thesis, two on-line algorithms for providing QoS tuning at the intermediate nodes are proposed. First, the jitter control algorithm aims at smoothing the jitter variance with low queueing delay latency. So that the buffer allocated at destination becomes smaller. Second, the burst packet loss reduction algorithm is designated to decrease the frequency of continuous packet loss. This makes the voice stream more robust to go well with burst or congestion conditions, thus, can be performed with higher quality at the client site.
中文摘要 ii
感謝函 iii
Abstract iv
Chapter 1. Introduction 1
1.1 Motivation 1
1.2 QoS of streaming voice on network 1
1.2.1 End-to-end delay 1
1.2.2 Delay jitter 1
1.2.3 Loss rate 2
1.3 Related works 2
1.3.1 Work-conserving services 2
1.3.2 Nonwork-conserving services 3
Chapter 2. The service model of multi-channel streaming voices 7
2.1 Sender 7
2.2 Switch 8
2.3 Receiver 8
Chapter 3. The proposed QoS algorithms for streaming voice 10
3.1 Off-line algorithm 10
3.2 On-line algorithm 11
3.2.1 The jitter smoothing algorithm 12
3.2.2 The burst packet loss reduction algorithm 15
Chapter 4. Simulation 20
4.1 Trade-off between jitter variance and delay latency 20
4.2 Results of the jitter smoothing algorithm 21
4.2.1 Sources with exponential distribution 22
4.2.2 Sources with hybrid distribution 25
4.3 Results of burst packet loss reduction algorithm 26
Chapter 5. Conclusions and future works 30
References 32
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[6] F. M. Tsou, H. B. Chiou, and Z. Tsai, “Design and simulation of an efficient real-time traffic scheduler with jitter delay and guarantees,” IEEE Tran. Multimedia, vol. 2, no. 4, pp. 255-266, Dec. 2000.
[7] Y. Mansour, and B. Patt-Shamir, “Jitter control in QoS networks,” IEEE/ACM Tran. Networking, vol. 9, no. 4, pp. 492-502, Aug. 2001.
[8] J. C. Bolot, F. P. Sacha, and D. Towsley, “Adaptive FEC-based error control for internet-telephony,” 1999 IEEE, pp. 1453-1460.
[9] O. J. Wasem, D. J. Goodman, C. A. Dvorak, and H. G. Page, “The Effect of Waveform Substitution on the Quality of PCM Packet Communications,” IEEE Transaction on Acoustics, Speech and Signal Processing, vol. 36, no. 3, pp. 342-348, Mar. 1998.
[10] J. F. Wang, J. C. Wang, J. F. Yang, and J. J. Wang, “A voicing-driven packet loss recovery algorithm for analysis-by-synthesis predictive speech coders over internet,” IEEE Transaction on Multimedia, vol. 3, no. 1, pp. 98-107, Mar. 2001.
[11] L. Zheng, L. Zhang, and D. Xu, “Characteristics of Network Delay and Delay Jitter and its Effect on Voice over IP (VoIP),” IEEE International Conference on Communications 2001, vol. 1, pp. 122-126.
[12] ITU-T Recommendation G.723.1, “Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s”.
[13] ITU-T Recommendation G.729, “Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)”.
[14] A. Tanenbaum, Computer Networks, 3rd ed. Englewood Cliffs, NJ: Prentice-Hall, 1996.
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