現今Internet的發展規模是藉著互連異質網路，而不斷地延伸著。由於不同區域通訊系統之頻寬及容量等傳輸資源不同，加上路由器對IP資料包的存轉處理通常只有選徑功能，因此從來源到目的端對端壅塞控制是必須的。在IP之上的傳輸協定即提供此類的機制，如TCP被設計來維持壅塞避免的可靠性傳遞。
然而，對於多媒體串流（multimedia streaming）的應用，TCP的可靠要求卻需要付出很不利於此類應用的代價，有如重送對時間延遲的增加，以及滑動視窗（sliding window）基礎的爆衝（burst）。我們需要有調整速率基礎的傳輸協定，並能支援媒體展現的格式，RTP/RTCP已為此成形，但是多媒體的應用仍有重要研究議題。
兩個顯著的因素包括：較長週期控制的壅塞避免效果？速率控制是否為TCP-friendly？在這篇論文裡，提出根據類似RTP/RTCP的接收端對封包的loss-ratio及jitter-ratio等採樣與使用時間基礎的TCP模型以進行TCP-friendly速率估計。所謂時間基礎模型是指以時間為變數的速率方程式；而現有的封包基礎模型則是以計數封包為變數的方程式。估計的意思是對於Internet上某個固定速率的連線，其接收端能很接近地估計與其競爭的其他TCP連線的平均速率。若有良好的速率估計，這就能成為速率調整與壅塞避免等機制的基礎。
這個研究包括了推導時間基礎的TCP傳輸率模型，在接收端的取樣與分析機制，以及讓UDP固定速率連線在理想及競爭環境下的ns2模擬實驗，結果顯示我們的方式有良好的估計效果。此做法和其他的相關研究的不同點，在於雖然它們有些可以達到TCP-friendly，速率卻不平順，或者不能配合RTCP較長週期控制。另外的不同點則是我們採用創新的模型和參考jitter資訊。

By interconnecting heterogeneous networks, the deployment scale of Internet is continuously expanding. Since distinct local communication systems may have different resources like capacities and bandwidths, and the IP datagrams store-and-forward processing of routers is usually in routing semantics, the end-to-end congestion control on the source and destination is necessary. The transport protocol over IP provides the mechanism. For example, TCP is designed for reliable data delivery with congestion avoidance.
However, for the multimedia streaming applications, the reliability requirement of TCP has some costs which are harmful to this kind of applications. These costs are, for example, the un-necessary delay to retransmit data, and the packets burst of sliding window-based scheme. Thus, it is necessary to have an alternative transport protocol using rate-based scheme and supporting the multimedia presentation profiles. RTP/RTCP suite is designed for these purposes, but there still are the critical research issues for multimedia streaming applications.
Two significant factors are the performance of longer periodic control, and the friendliness to TCP on the rate adjustment. To satisfy these two criteria, this dissertation proposes a rate estimation scheme based on the packets loss ratio and jitter ratio sampling at the RTP/RTCP-like receiver and using the time-based TCP model. The ‘time-based’ model is a rate equation of time variables, while the existing model, says ‘packet-based’, is another equation of the variable of packets count. The meaning of the rate estimation is that for one connection of a particular sending rate, the receiver can closely estimate the average transmission rate of other TCP traffic flows in competition. If the rate estimation performs well, it can be the basis of rate adjustment and congestion avoidance.
This research contains the inferring of the time-based TCP model, the functionalities for sampling and analysis at receiver, and ns-2 simulation of fixed sending rate UDP under ideal and competition environments. The results show that this approach conducts good estimation. The time-based proposal addresses the drawbacks of existing proposals that even with TCP friendliness, their sending rates are not smooth, or they are not suitable for RTCP long periodic control. Other differences are the innovative model for TCP and the consideration of jitter information.

AbstractII
摘要IV
ContentsVI
List Of FiguresVIII
List Of TablesXI
1. Introduction1
1.1 Internet: Best Effort in General1
1.2 Transport Protocol: End-to-End Control2
1.3 Media Steaming and Flow Control Issues3
1.4 Requirement on Multimedia Transport Design5
1.5 Organization of the Dissertation7
2. Related Work8
2.1 Researches on Control Schemes8
2.2 Practical Issues on These Researches13
2.3 Motivation on Time-Based TCP Model21
3. Time-Based TCP Model24
3.1 Simple Model24
3.2 Complex Model with Retransmission Time Out27
4. TCP-Friendly Rate Estimation36
4.1 Architecture for Rate Estimation36
4.2 Loss Ratio and Jitter Ratio Sampling37
4.3 Round Trip Time Sampling41
4.4 Congestion Event Period Calculation43
4.5 Weighted Average on Samples46
5. Simulation Results49
5.1 Modifications and Configurations49
5.2 Phase 1: On Specified Drop Pattern51
5.3 Phase 2: On Phase Effect from TCP58
5.4 Phase 3: On Resource Competition with TCP66
6. Conclusion and Future Work74
References76

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