| 研究生: |
張嘉祜 Chia-Hu Chang |
|---|---|
| 論文名稱: |
視訊串流於盡力傳送式網路上之調適性平順化研究 Adaptive Smoothing for Streaming Videos over Best-Effort Network |
| 指導教授: |
張寶基
Pao-Chi Chang |
| 口試委員: | |
| 學位類別: |
碩士 Master |
| 系所名稱: |
資訊電機學院 - 通訊工程學系 Department of Communication Engineering |
| 畢業學年度: | 93 |
| 語文別: | 中文 |
| 論文頁數: | 120 |
| 中文關鍵詞: | 盡力傳送式網路 、視訊串流 、調適性平順化 |
| 外文關鍵詞: | best-effort network, video streaming, adaptive smoothing |
| 相關次數: | 點閱:8 下載:0 |
| 分享至: |
| 查詢本校圖書館目錄 查詢臺灣博碩士論文知識加值系統 勘誤回報 |
隨著寬頻網路的普及與多媒體壓縮技術的成熟,多媒體資料透過網路傳輸的相關應用日趨豐富。而不用等待多媒體資料的完整下載便可以即時播放的串流技術是目前視訊傳輸的發展趨勢。但由於目前的網際網路大都是盡力傳送式的環境,其變動的可用頻寬和網路的延遲變異等問題,均對具有叢集特性的視訊訊務其接收品質造成嚴重的影響。此外,在多媒體網路應用的蓬勃發展下,非TCP的資料流在網路中將佔有極大的比例;然而非TCP資料流大都不具有壅塞控制的特性,勢必造成TCP資料流的可用頻寬枯竭,進而癱瘓整個網路。
在本論文中,根據上述問題提出一套在盡力傳送式網路下,具有調適性平順化傳輸機制的端點對端點視訊串流傳輸系統。本系統分為兩部分:視訊串流傳輸平順化機制與對TCP友善的傳輸率控制機制。本論文針對此兩個機制進行整合,首先根據初始的網路狀況、可用的緩衝區資源以及視訊訊務的特性,規劃視訊流量平順化的傳輸排程,並在一定的視訊接收品質保障下,利用對TCP友善的傳輸率控制機制,適時地反應網路壅塞情況予視訊傳輸平順化機制,進一步動態地調整平順化的傳輸排程以符合目前的網路現況,提高視訊接收品質。模擬結果顯示,在盡力傳送式的網路環境下,當平均可用頻寬約等於視訊所需的平均傳輸率時,可將原本僅使用TFRC的線上視訊串流其5.67%的封包丟棄率降至0.91%。
With the technology advances in multimedia compression and Internet, multimedia applications deliveried over Internet are dramatically boosted. Curently, real-time streaming without waiting for complete downloads is the trend of video delivery. However, it is very difficult to allocate resources effectively due to the bursty nature of video traffic. The impact of variable available bandwidth and network delay, which results from a best-effort network such as today’s Internet, may degrade the received video quality drastically. Besides, these multimedia streaming applications generally utilize the UDP protocol that does not provide the congestion control. This may lead to congestion collapse and starvation of TCP traffic in the Internet.
Therefore, this thesis proposes an end-to-end adaptive video streaming system by integrating the TCP-Friendly Rate Control (TFRC) with the video traffic smoothing algorithm. The proposed framework can adaptively adjust the smoothing transmission schedule to ensure the smooth delivery quality for different real-time and pre-stored video streams with low quality degradation. The adaptation is based on the current network condition, the available resources and the characteristics of video traffic. Simulation results show that, when the average available bandwidth is close to the video target encoding rate, the proposed system can effectively reduce the total packet loss rate of online video streaming from 5.67% to 0.91% compared with traditional TFRC solution.
[1] S. Sen, J. L. Rexford, J. K. Dey, J. F. Kurose and D. F. Towsley, “Online Smoothing of Variable-Bit-Rate Streaming Video,” IEEE Transactions on Multimedia, vol. 2, no. 1, pp. 37–48, Mar. 2000.
[2] D. Loguinov and H. Radha, “Effects of Channel Delays on Underflow Events of Compressed Video Over the Internet,” IEEE ICIP, Sep. 2002.
[3] J. Widmer, R. Denda, M. Mauve, “A survey on TCP-friendly congestion control,” Network, IEEE , Vol. 15 , no. 3 , pp. 28–37, May-June 2001.
[4] R. Rejaie, M. Handley, and D. Estrin, “Rap: An End-to-End Rate-Based Congestion Control Mechanism for Realtime Streams in the Internet,” Proc. IEEE INFOCOM, Mar. 1999.
[5] J. Padhye, D. Kurose, and R. Towsley, “A model based TCP-friendly ratecontrol protocol,” Proc. Int’l. Wksp. Network and Op. Sys. Support for Digital Audio and Video, Jun. 1999.
[6] S. Floyd et al., “Equation-based Congestion Control for Unicast Applications,”Proc. ACM SIGCOMM, Stockholm, Sweden, Aug. 2000, pp. 43–56
[7] S. Floyd, M. Handley, J. Padhye and J. Widmer, “Equation-Based Congestion Control for Unicast Applications: the Extended Version”, ICSI tech report TR-00-03, Mar. 2000.
[8] J. Vieron, C. Guillemot, “Real-time constrained TCP-compatible rate control for video over the Internet,” IEEE Trans. Multimedia, Vol. 6, no. 4, pp. 634–646, Aug. 2004.
[9] J. Feng, K.T. Lo, “A simple hierarchical traffic model for VBR MPEG video,” Performance, Computing and Communications, IPCCC ''98., IEEE International, no. 16-18, pp.147–153, Feb. 1998.
[10] K. Chandra, A.R. Reibman, “Modeling one- and two-layer variable bit rate video,” IEEE/ACM Trans. Networking, Vol. 7, no. 3, pp.398–413, Jun. 1999.
[11] W.-C. Feng, J. Rexford, “Performance evaluation of smoothing algorithms for transmitting prerecorded variable-bit-rate video,” IEEE Trans. Multimedia, Vol. 1, no. 3, pp.302–312, Sep. 1999.
[12] W. Feng, F. Jahanian, and S. Sechrest, “Optimal buffering for the delivery of compressed prerecorded video,” ACM Multimedia Syst. J., pp. 297–309, Sep. 1997.
[13] J. D. Salehi, Z.-L. Zhang, J. F. Kurose, and D. Towsley, “Supporting stored video: Reducing rate variability and end-to-end resource requirements through optimal smoothing,” IEEE/ACM Trans. Networking, vol. 6, pp. 397–410, Aug. 1998.
[14] ISO/IEC JTC1/SC29/WG11, “MPEG-4 Video Verification Model version 18.0,” N3908, Jan. 2001.
[15] ISO/IEC JTC1/SC29/WG11, “Text of ISO/IEC 14496-2: 2001/COR2,” N5158, Oct. 2002.
[16] T. Wiegand, G. J. Sullivan, G. Bjontegaard, and A. Luthra, “Overview of the H.264/AVC video coding standard,” IEEE Trans. Circuits Syst. Video Technol., vol. 13, Jul. 2003.
[17] J. L. Mitchell, W. B. Pennebaker, C. E. Fogg, and D. J. LeGall, MPEG Video Compression Standard, Chapman & Hall, 1997.
[18] R. Talluri, “Error-Resilient Video Coding in the ISO MPEG-4 Standard,” IEEE Commun. Mag., vol. 36, no. 6, pp. 112–119, Jul. 1998.
[19] D. Wu, Y.T. Hou, W. Zhu, T.H. Chiang, Y.Q. Zhang and H.J. Chao, “On end-to-end architecture for transporting MPEG-4 video over the Internet,” Circuits and Systems for Video Technology, IEEE Transactions, vol. 10, pp. 923–941, Sep. 2000.
[20] F.L. Leannec and G.M. Guillemot, “Error Resilient Video Transmission Over the Internet,” in SPIE Proceeding Visual Communications and Image Processing (VCIP’99), Jan. 1999.
[21] T. Turletti and C. Huitema, “RTP payload format for H.261 video streams,” RFC 2032, Oct. 1996.
[22] C. Zhu, “RTP payload format for H.263 video streams,” RFC 2190, Sep. 1997.
[23] Y. Kikuchi, T. Nomura, S. Fukunaga, Y. Matsui, H. Kimata, “RTP Payload Format for MPEG-4 Audio/Visual Streams,” RFC 3016, Nov. 2000.
[24] ISO/IEC 14496-8 Information technology – Coding of audio-visual objects – Part 8: Carriage of ISO/IEC 14496 contents over IP networks, 2001.
[25] RTP: A Transport Protocol for Real-Time Applications, RFC 1889, Jan. 1996.
[26] RTSP: Real Time Streaming Protocol, RFC 2326, IETF 1998.
[27] SDP: Session Description Protocol, RFC 2327, IETF 1998.
[28] S. Floyd and K. Fall, “Promoting the Use of End-to-end Congestion Control in the Internet,” IEEE/ACM Trans. Net., vol. 7, no. 4, pp. 458–472, Aug. 1999.
[29] J. Widmer, “Equation-based Congestion Control,” Feb. 2000, Diploma Thesis.
[30] J. Padhye et al., “Modeling TCP Reno Performance: A Simple Model and Its Empirical Validation,” IEEE/ACM Trans. Net., vol. 8, no. 2, pp. 133–145, Apr. 2000.
[31] Paxson V. and M. Allman, “Computing TCP’s Retransmission Timer,” RFC 2988, Nov. 2000.
[32] J. Rexford, S. Sen, J. Dey, W. Feng, J. Kurose, J. Stankovic, and D. Towsely, “Online smoothing of live, variable-bit-rate video,” in Proc. Workshop on Network and Operating System Support for Digital Audio and Video, pp. 235–243, May 1997.
[33] UCB/LBNL/VIANT. (1998) Network Simulator – NS (version 2).