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。

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.

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