We introduce two thresholds of the packet loss ratio to divide the network load condition into three status : congest , normal , idle 通過兩個門限值c 、 _ ,將網(wǎng)絡(luò)的負載狀況分為擁塞、適中、空閑三種狀態(tài),進一步減小發(fā)送速率的振蕩頻率。
Sending rate could be adjusted according to the rate scheme in the end host . simulations showed that etcc can adjust sending rate smoothly , decrease delay and packet loss ratio while maintaining good tcp - friendliness 仿真實驗表明etcc機制可以在保證tcp友好的前提下平滑調(diào)節(jié)多媒體流的發(fā)送速率,同時降低網(wǎng)絡(luò)的延時和丟包率。
The main contributions of the thesis are : ( 1 ) we present an end - to - end transport architecture using the rtp / udp / ip protocol stack and employ an efficient and robust packetization algorithm for mpeg - 4 video bit - streams at the sync layer for internet transport . ( 2 ) we study the congestion control mechanism based on aimd algorithm , and make improvement in order to reduce the oscillation of transimition rate due to tremendous contrast of packet loss ratio caused by dynamical change of the network load 論文的主要貢獻在于:提出了基于rtp的mpeg - 4視頻傳輸模型并充分利用mpeg - 4的videoobjectplane ( vop )特性,采用適用于mpeg - 4視頻傳輸?shù)膔w載荷格式及組包算法,同時具有傳輸?shù)母咝院蛠G包的魯棒性。
We chose suitable tcp throughput model to estimate the available bandwidth correctly , using the estimated round trip time and packet loss ratio for the next time interval as parameters of the model to achive the accuracy of estimated network bandwidth . as the observed losses and round trip time vary very dynamically , adjust the sending rate equivalent to the amount of tcp throughput may result in a rather fluctuant sending rate . so we present a rate adjustment like tcp congestion control based on aimd , which increases its sending rate by an additive inereease rate 根據(jù)mpeg4視頻流應(yīng)用的特點,選擇合適的吞吐量模型,進行合理的參數(shù)估計,并根據(jù)計算出的帶寬進行相應(yīng)的速率調(diào)整來實現(xiàn)擁塞控制,我們使用未來rtt的估計值和分組丟失率的估計值作為吞吐量模型的參數(shù),增強了控制的實時性,弱化了業(yè)務(wù)的振蕩性,提高了帶寬預(yù)測的準確性;在進行速率調(diào)整時,不是簡單地將發(fā)送速率調(diào)整到與tcp吞吐量模型一致,而是采用類似tcp的aimd策略來調(diào)節(jié)發(fā)送速率,減小了發(fā)送速率的振蕩性。