在即時多媒體系統中以loss-jitter做為
速率調整之應用

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姓名

姜怡佳(Yi-Chia Chiang) 查詢紙本館藏

畢業系所

資訊工程學系

論文名稱

在即時多媒體系統中以loss-jitter做為速率調整之應用
(The loss-jitter Based Adjustment for Multimedia Applications )

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摘要(中)

在網路上傳播即時多媒體是重要的應用，多媒體應用在網路上有三項需求是要注意的，分別是頻寬、網路延遲及封包遺失，然而這些需求在現有的網路上是無法主動提供我們服務品質保證(QoS)。為了有效率的使用頻寬，端點對端點的控制是必要的，在相關的議題中，頻寬調整及TCP-friendly是我們應要探討的。
我們提出一個適合即時多媒體傳輸的架構，我們將它細分成傳輸層及應用層。在傳輸層中，傳送端會根據接收端回報的資訊做為調整速率的依據。在先前的研究，都是以封包遺失率當作調整速率的準則，然而我們認為，劇跳(jitter)也可以當作速率調整的一個指標。
我們要提出的方法為，第一，傳送端會將遺失率(loss)及劇跳(jitter)這兩項資訊當作速率調整的依據，因此傳送端可以算出連續且公平分享的頻寬。但是應用層的編碼/解碼器能使用的速率可能是離散的，第二，我們要將傳輸層得到的連續頻寬對應到離散頻寬。在無服務品質保証的網路上發生封包遺失是無可避免的，最後，我們採用重送機制來提升接收品質。模擬結果顯示我們提出的架構不僅提升網路使用率而且能改善接收品質。

摘要(英)

Delivering real-time streaming over the Internet is one of the critical applications nowadays. Novel real-time Internet applications have bandwidth, delay, and loss requirements. However, the current Internet offers best effort services without sufficient Quality of Service assurance. To provide an efficient bandwidth allocation for multimedia applications, end-to-end control is necessary. Specifically, the issues of bandwidth adjustment and TCP-friendly should be properly addressed.
This thesis presents a QoS adjustment architecture for real-time applications, which includes transport layer and application layer. In transport layer, the video sender adjusts the transmission rate by the feedback information (RTCP) from the receiver. In addition to the traditional loss rate for rate-adjustment decision, the thesis compounds loss rate and jitter as feedback information and adjust the transmission rate according loss-jitter adjustment parameters (LJA). From the LJA scheme, the continuous scales of bandwidth will be maintained for a RTP connection; however, the codec scales for the multimedia applications could be discrete. To support both of scales, a mechanism is applied to map the continuous bandwidth to the codec bit-rate. To address the unavoidable packet loss, an error recovery mechanism by retransmission is adopted to enhance the reception quality in application layer. The results show that the proposed architecture does improve the bandwidth utilization and reception quality for real-time applications.

關鍵字(中)

★ 即時速率控制

關鍵字(英)

★ Real-time streaming
★ tcp-friendly
★ RTP

論文目次

CHAPTER　1 INTRODUCTION 1
1.1 BACKGROUND AND MOTIVATION 1
1.2 MULTIMEDIA APPLICATION SYSTEM 5
1.2.1 Transport Layer 5
1.2.2 Application Layer 6
1.3 CONTRIBUTIONS 6
CHAPTER　2 RELATED WORKS 8
2.1 TRANSPORT LAYER CONGESTION CONTROL 8
2.2 PROBING-BASED APPROACH 10
2.2.1 Dynamic QoS on RTP 10
2.2.2 Rate Adaptation Protocol (RAP) 11
2.2.3 The Loss-Delay Based Adjustment scheme (LDA) 12
2.2.4 The Direct Adjustment Algorithm (DAA) 12
2.3 MODEL-BASED APPROACH 13
2.3.1 Simple steady-state model 13
2.3.2 TCP-Friendly Rate Control (TFRC) 14
2.4 ERROR RECOVERY 14
2.5 STUDY OF NETWORK JITTER 15
CHAPTER　3 REAL-TIME MULTIMEDIA SYSTEM 19
3.1 LOSS-JITTER ADJUSTMENT SCHEME (LJA) 21
3.2 ERROR RECOVERY 25
3.2.1 Single Retransmission 25
3.2.2 Multiple Retransmissions 27
3.2.3 Retransmission with QoS support 28
3.3 BUFFER DESIGN 31
3.4 CODEC BIT-RATE ESTIMATION 33
3.4.1 Probing-based bit-rate estimation 33
3.4.2 Sender-driven bit-rate estimation 36
CHAPTER　4 SIMULATION AND PERFORMANCE EVALUATION 38
4.1 SIMULATION TOPOLOGY AND PARAMETERS SETUP 38
4.2 BANDWIDTH COMPETITION AND RESPONSIVENESS 41
4.3 ERROR RECOVERY 45
4.3.1 Error Recovery without QoS support 45
4.3.2 Error Recovery with QoS support 48
4.4 CODEC RATE ESTIMATION 50
4.4.1 Probing-based codec bit-rate estimation 50
4.4.2 Sender-driven codec bit-rate estimation 51
4.5 COMPARISON OF TCP AND RTP 53
4.6 PERFORMANCE EVALUATION 55
4.6.1 Inter-protocol fairness 55
4.6.2 Intra-protocol fairness 56
CHAPTER　5 CONCLUSION 58
CHAPTER　6 REFERENCE 60

參考文獻

REFERENCE
[1] R.Braden, L.Zhang, S. Berson, S. Herzog, and S. Jamin, “Resource reServation protocol (RSVP) version 1 functional specification,” RFC 2205, Sept. 1997
[2] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang, AND W. Weiss, “An architecture for differentiated services,” RFC2475,IETF, Dec. 1998
[3] Postel, J., "Transmission Control Protocol – DARPA Internet Program Protocol Specification", RFC 793, DARPA, September 1981
[4] Postel, J., "User Datagram Protocol", RFC 768, USC/Information Sciences Institute, September 1981
[5] Jacobson, Van, “Congestion Avoidance and Control", SigComm `88, Symp., ACM, Aug. 1988, pp. 314-329
[6] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson. “RTP：a transport protocol for real-time applications”, Technical Report RFC 1889, Internet Engineering Task Force, Jan.1996
[7] S. Floyd and K. Fall, "Promoting the Use of End-to-End Congestion Control in the Internet," Submitted to IEEE/ACM Transactions on Networking, 1998
[8] I. Busse, B. Deffner, and H. Schulzrinne, “Dynamic QoS control of multimedia applications based on RTP”, Computer Communications, Jan. 1996
[9] D. Chiu and R. Jain, “Analysis of the Increase/Decrease Algorithms for Congestion Avoidance in Computer Networks”, Journal of Computer Networks and ISDN, Vol. 17, No. 1, June 1989, pp. 1-14
[10] Dorgham Sisalem, “End-To-End Quality of Service Control Using Adaptive Applications”, IFIP Fifth International Workshop on Quality of Service IWQOS ’’97,New York, USA, IFIP WG 6.1,May 1997
[11] Dorgham Sisalem, Frank Emanuel, Henning Schulzrinne, “The Direct Adjustment Algorithm: A TCP-Friendly Adaptation Scheme”, Quality of Future Internet Services Workshop, Berlin, Germany, September 25-27, 2000
[12] Dapeng Wu, Hou, Y.T., Wenwu Zhu, Hung-Ju Lee, Tihao Chiang, Ya-Qin Zhang, Chao, H.J., ”On end-to-end architecture for transporting MPEG-4 video over the Internet”, Circuits and Systems for Video Technology, IEEE Transactions on, Volume: 10 Issue: 6, Sept. 2000, Page(s): 923 –941
[13] D. Ferrari, “Delay jitter control scheme for packet-switching internetworks,” Computer Communications, Vol. 15 N ffi 6 (July/August 1992)
[14] Rejaie, R.; Handley, M.; Estrin, D., “RAP: An end-to-end rate-based congestion control mechanism for realtime streams in the Internet”, INFOCOM ’’99. Eighteenth Annual Joint Conference of the IEEE Computer and Communications Societies. Proceedings. IEEE , Volume: 3 , 1999 ,Page(s): 1337 -1345 vol.3
[15] D. Sisalem and H. Schulzrinne, “The loss-delay based adjustment algorithm: A TCP-friendly adaptation scheme”, Workshop on Network and Operating System Support for Digital Audio and Video, 1998
[16] J. C. Bolot, “Characterizing end-to-end packet delay and loss in the internet”, Journal of High Speed Networks, 2(3):289-- 298, September 1993
[17] Sisalem, D.; Wolisz, A, “LDA+: a TCP-friendly adaptation scheme for multimedia communication”, Multimedia and Expo, 2000. ICME 2000. 2000 IEEE International Conference on , Volume: 3 , 2000 ,Page(s): 1619 -1622 vol.3
[18] J. Padhye, V. Firoiu, D. Towsley, and J. Kurose, “Modeling TCP throughput: a simple model and its empirical validation”, ACMSIGCOMM, September 1998
[19] S. Floyd and F. Kevin, “Router mechanisms to support end-to-end congestion control”, Technical report, Feb. 1997
[20] Sally Floyd, Mark Handley, Jitendra Padhye, and Joerg Widmer, “Equation-Based Congestion Control for Unicast Applications", ACM SIGCOMM 2000, Stockholm, Aug 2000
[21] M. Podolsky, K. Yano, S. McCanne, “A RTCP-based Retransmission Protocol for Unicast RTP Streaming Multimedia”, IETF, draft-podolsky-avt-rtprx-00.txt, October, 1999
[22] D. Leon, V. Varsa, Nokia, “RTP retransmission framework”, IETF, draft-leon-rtp-retransmission-02.txt, March 2002
[23] C. Papadopoulos, M. Parulkar, “Retransmission-Based Error Control for Continuous Media Applications”, NOSSDAV 1996
[24] V. Jacobson, “Congestion avoidance and control” In ACM SIGCOMM ’’88, 1988
[25] Sze, H.P.; Liew, S.C.; Lee, Y.B, “A packet-loss-recovery scheme for continuous-media streaming over the Internet”, IEEE Communications Letters , Volume: 5 Issue: 3 , March 2001,Page(s): 116 -118
[26] D. Loguinov, and H. Radha, “On Retransmission Schemes for Real-time Streaming in the Internet”, IEEE INFOCOM, 2001
[27] S. Floyd, V. Jacobson, ”Link-sharing and resource management models for packet networks,” Networking, IEEE/ACM Transactions on, Volume: 3 Issue: 4, Aug. 1995, Page(s): 365 –386
[28] Network simulator (ver. 2), LB. URL: http://www.isi.edu/nsnam/ns/
[29] Jörg Widmer, “Equation-Based Congestion Control”, Master’’s thesis, University of Mannheim / ATT Center for Internet Research at ICSI (ACIRI), 2000
[30] R. Jain, D. Chiu, and W. Hawe, "A Quantitative Measure Of Fairness And Discrimination For Resource Allocation In Shared Computer Systems", DEC Research Report TR-301, September 1984
[31] Hassan Shojania, Baochun Li, ” Experiences with MPEG-4 Multimedia Streaming”, ACM Multimedia, p492-p494, 2001
[32] Seung-Gu Na and Jong-Suk Ahn, “TCP-like Flow Control Algorithm for Real-Time Application”, Networks, 2000. (ICON 2000). Proceedings. IEEE International Conference on , 2000 ,Page(s): 99 –104
[33] Brakmo, L.S.; Peterson, L.L., “TCP Vegas: end to end congestion avoidance on a global Internet”, Selected Areas in Communications, IEEE Journal on, Volume: 13 Issue: 8, Oct. 1995, Page(s): 1465 -1480

指導教授

吳曉光(Hsiao-Kuang Wu)

審核日期

2002-7-4

推文