packet loss rate 中文意思是什麼

packet loss rate 解釋
丟包率
  • packet : n 1 包裹；小件行李；（郵件等的）一捆；小批；袋。2 （定期）郵船，班輪。3 〈英俚〉（打賭等中輸贏的...
  • loss : n. 1. 喪失；丟失，遺失。2. 減損，損失，虧損（額）；損耗；減少，下降。3. 失敗；輸掉。4. 錯過；浪費。5. 損毀；【軍事】傷亡；〈pl. 〉 傷亡及被俘人數。
  • rate : n 1 比率，率；速度，進度；程度；（鐘的快慢）差率。2 價格；行市，行情；估價，評價；費，費用，運費...
  1. Subsequently, taking into consideration the characteristics of audio data over internet including delay, jitter, packet loss and etc., we propose a series of methods for solving this above problems, such as pre - storage technology, buffer technology, dynamic adjustment of the voice - coding rate to the state of network and integrated media synchronization playing mechanism, and etc. in the end, simulation on 10 / 100m lan is made using the above methods, and the result of the experiment demonstrates the method has good performance and can improve the quality of the audio data transmission

    其次本文還深入研究了語音數據在非實時的internet數據網上的傳輸特性，這些特性包括延時、延時抖動、數據包丟失等。在本文的設計方案中提出了針對這些問題的解決方法，包括預取機制、設置緩沖區技術、動態速率調節技術以及媒體綜合同步播放機制等。最後採用這些方法在10 100m局域網上做了模擬實驗，實驗結果表明本文提出的方法是有效的，在網路狀況惡劣的情況下能夠改善語音播放質量。
  2. Compared with blue, it can improve link utilization and decrease packet loss rate at the same time

    和blue相比，它在提高鏈路利用率的同時可以降低丟包率。
  3. This paper proposes a new adaptive rate control scheme for h. 264 video, which based on packet loss decision

    現提出了一種新穎的基於丟包率判決來自適應控制h . 264視頻流量的傳輸方法。
  4. Real - time transport with dynamic error correction code utilizes error correction methods that vary with current packet loss rate and packet error rate of network

    該技術是在多媒體數據的實時傳輸中，依據網路丟包率和包錯誤率情況，動態的使用不同糾錯編碼方案的實時傳輸技術。
  5. The algorithm calculated packet loss rate according to the average queue length and waiting time

    這種演算法根據平均隊列長度和等待時間計算數據包的丟棄概率。
  6. Guaranteed qos includes three basic parameters : packet transfer delay, packet delay variation and packet loss rate

    有保證的服務品質包含三種參數：封包傳輸延遲、封包延遲變異，以及封包遺失率。
  7. The sender tracks the packet loss status according to the algorithm - loss rate estimation based on variable frame size and switches two audio sample files in accordance with packet loss status

    傳送端根據以變動框架長度為基礎之遺失率估演算法，來估算目前網路上封包遺失的狀況，並以此作為切換高低音質檔案的依據。
  8. Simulations confirm compared with faa, the mdf can save more lrwcs and fewer packet loss probability, especially in the condition of high load. using mdf algorithm, the number of lrwcs in the condition of high load is more than in the condition of low load when the packet loss rate approaches to a fixed value

    通過模擬實驗驗證了該演算法在低、中和高負載情況下，比faa演算法更節約波長轉換器和更小的丟包率，而且在低負載時mdf演算法的優勢更加明顯。 mdf演算法在高負載下，丟包率到達穩定值所需要的lrwcs數目比低負載多。
  9. Packet loss rate

    分組丟失率
  10. Increasing the converter range of lrwcs could reduce the packet loss rate and meanwhile,

    我們對同時使用波長轉換和fdl競爭解決方式的交換節點模型，
  11. 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

    模擬實驗表明etcc機制可以在保證tcp友好的前提下平滑調節多媒體流的發送速率，同時降低網路的延時和丟包率。
  12. Ahg report on avs1 - p7 performance testing, sept. 2005. 72 stockhammer t, kontopodis d, wiegand t. rate - distortion optimization for h. 26l video coding in packet loss environment. in 12th international packet video workshop pv 2002, pittsburg, py, may 2002

    對rtp打包格式的介紹包括以下內容：什麼是rtp和rtp打包格式，目前國際上都有哪些視頻rtp打包格式標準， avs - m rtp打包格式及其與h . 264 avc rtp打包格式的比較。
  13. While the rate - based dropping on burst level large time scales determines the packet drop aggressiveness and is responsible for low and stable queuing delay, good robustness and responsiveness, the queue - based modulation of the packet drop probability on packet level small time scales will bring low loss and high throughput

    突發行為具有自相似或尺度不變性scale - invariant ，即流量在不同的時間尺度上具有相似的突發特性2局部縮放性。流量過程的局部奇異性使流量在小時間尺度數百ms及以下的突發非常強烈，具有非高斯分佈。
  14. 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

    論文的主要貢獻在於：提出了基於rtp的mpeg - 4視頻傳輸模型並充分利用mpeg - 4的videoobjectplane （ vop ）特性，採用適用於mpeg - 4視頻傳輸的rw載荷格式及組包演算法，同時具有傳輸的高效性和丟包的魯棒性。
  15. This method may effectively improve the quality of real - time transport of multimedia data with a little cost of computation under the situation of limited bandwidth, high packet loss rate and high packet error rate

    這個技術可有效的改善多媒體應用在帶寬資源緊缺的網路及丟包率和包錯誤率較高的網路中的傳輸質量。
  16. Mechanism and difficulty of current real - time transport technology are analyzed in this article, technologies such as multicast, real - time transport, resource reservation and error correction code technologies are studied in this dissertation. a method of improving the quality of real - time transport of multimedia data on network on where packet loss rate and packet error rate are high is provided

    本文分析了現有網路數據實時傳輸的機理及面臨的困難，研究了多播、實時傳輸、資源預留以及糾錯編碼等技術在實時傳輸中的應用，給出了在低帶寬、包錯誤率和丟包率較高的網路中提高多媒體數據實時傳輸質量的方法，提出了帶動態糾錯編碼的實時傳輸技術。
  17. The dynamic bit rate allocation module receives relevant data from the loss rate of multi - media data packet, jitter, delay, and compares these data with predefined values to make one of these 3 decisions that : decrease the video bit rate ( stat = decrease ), or maintain current video bit rate ( stat = hold ), or increase video bit rate ( stat = inccease ). the video bit rate will be adjusted according to the decision and the total bit rate is changed accordingly. in this way the impact of network environmental variation is settled, and consequently the adaptability of videoconference systems in remote education area is improved

    整個模型從rtcp協議獲得多媒體數據包丟失率（ lossrate ） 、抖動（ jitter ） 、延時（ delay ）的相關數據，這些數據與預先設定好的閥值作比較，得出減小視頻位速率（ stat = dectease ） 、維持視頻位速率不變（ stat = hold ）或增大視頻位速率（ stat = increase ）的決定，模塊根據這些決定對視頻位速率進行相應的調整，從而總的位速率也相應變化，這樣可以減小由於網路環境變化而帶來的影響，從而提高視頻會議在遠程教育應用中的實用性。
  18. 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mpeg4視頻流應用的特點，選擇合適的吞吐量模型，進行合理的參數估計，並根據計算出的帶寬進行相應的速率調整來實現擁塞控制，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，增強了控制的實時性，弱化了業務的振蕩性，提高了帶寬預測的準確性；在進行速率調整時，不是簡單地將發送速率調整到與tcp吞吐量模型一致，而是採用類似tcp的aimd策略來調節發送速率，減小了發送速率的振蕩性。
  19. Chapter 5 performs a simulative evaluation of mobile ip via ns - 2 for a scenario comprising of one home agent and two foreign agents based on ieee 802. 11 wireless lan standard. we describe a model which is adaptive to directional motion such as high speed vehicle. we represent the bandwidth and loss packet under constant bit rate ( cbr ) traffic

    本模型的主要特點是考慮了定向運動下諸如列車上的專用網路，移動節點在進行大流量通信下切換時的帶寬和丟包數的性能評估，並分析了丟包數和網路時延/移動速度之間的關系，獲得了相應的結論。
  20. In network study, it ’ s necessary to analyse and compare the network chararistic and technologies. in experiment, the researcher need provide the realistic traffic to simulate the real network environment by the traffic generator. they need to measure network ’ s chararistic about packet loss rate, transfer delay, delay jitter

    在研究階段，需要對這些新技術的各種特性進行深入的分析和比較。為了在實驗室的網路環境中模擬出接近於真實的網路場景，需要根據實驗的要求生成相應的網路流量。
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