在盡力式的網路(Best-effort Networks)裡，使用壅塞控制在多媒體串流服務裡來調整傳輸速率是一件非常重要的事。然而，傳統的TCP擁塞控制並不適用於即時性的多媒體串流服務裡，最主要的因素在於TCP爆炸式的傳輸以及陡峭速率波動影響了影音播放的品質。也因此有許多的研究針對了即時性多媒體串流服務的特性提出了友善式傳輸層協定(TCP-Friendly)的壅塞控制。
　　在針對即時性多媒體串流服務的友善式傳輸層協定裡，他們主要把焦點放在提供一個比較平穏的傳送速率以及如何與網路上的其它傳輸服務平等的共享網路的資源。例如，友善式傳輸層協定壅塞控制(TCP-Friendly Rate Control, TFRC)與接收端傳輸層協定仿真(TCP Emulation At Receivers, TEAR)把同一個封包來回時間(RTT)內所有網路封包遺失視作同一個事件，藉由此種做法來改善TCP陡峭的速率波動，而他們的方法也都被證明能與TCP公平的共享網路資源。然而，當傳送端以友善式壅塞控制來傳送聲音封包時，傳統的VoIP適應式talkspurt調整機制會造成大量的封包在接收端被丟棄，並且會產生較大的播放延遲。
　　因此，為了解決這個問題，這篇論文提出了一個系統架構。在這個系統架構裡，我們使用現存的友善傳輸層協定(TFRC)來協助傳輸聲音封包，而為了解決傳統VoIP適應式talkspurt調整機制所造成的問題，我們提出了二個控制器，一個以傳送端為基礎(sender-based)，另一個則以接送端為基礎(receiver-based)的控制器來改善聲音的品質，它們分別為編碼器位元率控制(encoder bit-rate controller, EBC)以及適應式撥放控制(adaptive playout controller, APC)，分別用來調整伺服傳送端壓縮編碼的位元率以及調整接收端封包的撥放時間長度。最後，我們使用NS-2模擬器來模擬APC以及EBC的效能，以及使用Bernoulli遺失模型來分析我們此系統的效能。模擬結果顯示，我們的系統在擁塞發生時可以有效的降低封包在接收端被丟棄的機率以及可以有效的降低平均的播放延遲，數學分析的結果也證明所提出方法的穩定性以及適用性。

　　In the best-effort networks, it is important for multimedia streaming applications to adjust their sending rate for congestion avoidance. However, TCP is ill-suited to real-time multimedia streaming applications due to its bursty transmission and abrupt rate fluctuations. Hence, many researchers are investigating TCP-Friendly congestion control for real-time applications.
　　In TCP-Friendly congestion control mechanisms for real-time applications, they focus on providing a smooth sending rate and compete with TCP flows fairly. For example, TCP-Friendly rate control (TFRC) and TCP Emulation at Receivers (TEAR) regard packets losses within a single RTT as a loss event to reduce the rate fluctuation and they all have fair performance compared with TCP flows. However, when a sender transmits voice packets under TFRC and the network is busy, traditional adaptive talkspurt adjustment mechanisms always cause serious packet drops and long playout delay since they do not take the network congestion problem into account. Therefore, under TFRC these mechanisms fail to maintain voice quality at the receiver.
　　In this thesis, we propose a novel systematic model for effective voice packets transmission in the best-effort network under TFRC control to solve the problems of VoIP talkspurt adjustment mechanisms. The proposed model considers both server and client sides’ improvements by implementing the encoder bit-rate controller (EBC) and the adaptive playout controller (APC) respectively. Finally, we use NS-2 simulator to simulate the performance of APC and EBC controllers and use the Bernoulli loss model in NS-2 simulator to analyze the characteristics of our system model. The simulation results present that our methods avoid most of unnecessary packet drops and reduce the average playout delay and work well under most of network states.

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