隨著網路的蓬勃發展，多媒體資料更加豐富且隨手可得，尤其串流技術更是使得多媒體播放更具有即時性，舉凡：隨選視訊系統（VOD），網路教學系統，視訊系統……等等應用更是因應而生。但是目前的網路仍是屬於best-effort(盡力而為)的網路，也就是說它不提供任何的品質保證，仍然無法提供最低頻寬的保證來傳輸多媒體資料。
　　在串流視訊系統中，因為多媒體傳輸乃是依據RTP協定傳送，而RTP乃是使用UDP作為傳輸協定，是一種非連接導向的傳輸。因此在這樣的網路環境下，使用者遇到擁塞的狀況時，我們提供一種應用層QoS機制，此種機制是由伺服器端來負責處理擁塞控制，不論是接收端因為網路頻寬造成擁塞，或是伺服器因為負荷過重導致無法處理過多的使用者時，伺服器都會採取應用層QoS機制來調適傳輸位元率，降低資料遺失，並且可以避免能因為延遲過大而接收端收到的資料皆是過時資料，播放器會將之丟棄，導致接收端播放不順暢甚至畫面停格。因此應用層QoS機制提供因為擁塞原因，適當調適傳輸資料，使之能夠即時傳送到接收端，並使得丟棄的資料達到最小。
此論文中之應用層QoS機制是採用畫面丟棄(frame-dropping)方式，因此也提出一個MOD-based畫面丟棄演算法來避免連續的畫面遺失，使得畫面停頓不會過長。

　　With rapid growth of the internet, we can reach rich multimedia data by the internet easily. For example, VOD (Video on Demand)、E-Learning、Video Conference …, etc. Specially, the streaming technology makes the real time media playback possible. However, the internet is a best-effort network. In other words, the QoS cannot be always satisfied under the limited bandwidth in the internet.
　　In the video streaming system, the multimedia data are transported by the RTP protocol which is based on UDP. The UDP is a non-connection protocol, the data-transporting may subject to the congestion under such environment. Therefore, we develop a mechanism in the application-layer QoS for the congestion. This mechanism can handle the congestion problem on the server, whether the congestion occurs on the client-side due to the insufficient bandwidth, or on the server-side due to the overload of user connections. The mechanism can adapt the transmission rate to reduce packet loss, and also can avoid the frame-dropping or pause due to out-of-date frame arisen from delay.
　　In the thesis, we also propose a MOD-based selected-frame-discard algorithm to meet the application-layer QoS. The algorithm can avoid the continuous frame loss so that the playback can be smoother. Experimental results also confirm the feasibility of the proposed algorithm.

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