packet rate 中文意思是什麼
packet rate
解釋
包速率-
Then, the paper present a protocol misusage ids in application layer based on markov chain, and obtain the following results after test : the system can have good effection on detecting the probe packet, dos attack and novelty attack. the system can overcome part of disadvantages in traditional intrusion detection system, and adapt requirements with different network. the experiment shows that the system can improve the detection accuracy rate, reduce the false negatives probability
通過對檢測系統的測試分析，系統對掃描探測報文、 dos攻擊報文和新的攻擊報文等有較好的檢測率。基於馬爾可夫鏈建立的網路異常檢測系統可以克服傳統入侵檢測系統的部分缺陷，能夠適應不同網路環境的要求，同時也有效提高了系統的檢測能力和檢測效率，降低了誤報率。 -
Thirdly, a novel parameter - varying adaptive algorithm for rtt and rto estimations based on the information theory and the maximum entropy principle ( mep ) is presented. it is used in the implementation of trinomial protocol to detect packet losses and to adjust the sending rate
再次，給出了一種新的基於信息理論和最大熵原理（ mep ）的變參數自適應rtt和rto估計演算法，在三項式協議實現過程中用於探測丟包及速率調整。 -
This thesis elucidates and analyses the key techniques and rules of aolr in detail, and introduces its simulation modeling of the integrated system infrastructure and each function modules. through systematic simulation of aolr in the performance of packet delivery rate, average end - to - end delay, average route set - up delay, routing overhead, and etc, the key techniques and algorithms are testified and can provide reference for practical system design and application
通過對基於蟻群優化的負載均衡路由協議系統的模擬，從數據分組成功遞交率、平均端到端時延、平均路由建立時延、路由開銷等性能的模擬數據和曲線上，分別驗證了該系統中的幾項關鍵技術和演算法。 -
The data transmission rate is effectively increased by using data packet as i / o read and write system call functions " parameter
本文通過將數據包作為i o讀寫系統調用函數的參數有效地提高了數據傳輸率。 -
Presented a differentiated approach of the broadband residential ethernet - based access network. analyzed per hop behavior and per domain behavior of ef service based on packet scale rate guarantee theo
提出了一種寬帶社區網的區分服務方法，基於每跳行為和每域行為確保實時業務服務質量，能夠有效地確保voip 、電視、 vod和高速網際網路接入的服務質量： -
A double time - scale rate control algorithm for multimedia real - time applications in packet switched networks
分組網上多媒體實時業務雙時間尺度速率控制演算法 -
In packet based wireless video communications, when best band distribution, not only source distortion should be considered, but also channel distortion caused by channel errors. the existing macroblock - layer rate control schemes calculates quantization parameters of all macroblocks ( mb ) in a frame in a raster scan order, and then encodes the mbs in the same order. actually, the quantization distortion is heavily dependent upon the coding order of mbs
在基於包的無線視頻傳輸中，最佳帶寬分配時，不僅要考慮信源編碼失真，也要考慮由於通道差錯引起的通道失真；已有的宏塊層碼率控制演算法以矩陣掃描的順序計算一幀中所有宏塊的量化參數，事實上量化失真與宏塊的編碼順序有很大關系，改變宏塊的編碼順序，使復雜的宏塊分配到更多的比特數，顯然能大大提高編碼效率。 -
Rbtcp is rate based, uses only end - to - end mechanism, performs rate control at the receiver, and uses inter - packet delays as the primary metric for rate control
Rbtcp是基於速率的，運用端到端的機制，以分組間的延遲作為主要的監控標準在接受端執行速率控制。 -
Performance analysis of variable - rate packet cdma system
瑞利衰落通道下差分跳頻通信系統性能分析 -
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。 -
In this paper, the mechanism, methods of and countermeasures to denial of services attacks are discussed. after that, several packet marking schemes for traceback are reviewed and some improvements to the basic packet marking scheme are given, which reduce the workload and false positive rate in the attack tree reconstruction. knowing that in existing packet marking schemes, router marks packets with fixed probability, which results in that many packets are required in path reconstruction and that attacker could encumber path reconstruction via spoofed marking information
在本文中，我們首先研究了拒絕服務攻擊的攻擊機制、方法及其對策，並對以包標記的方式追蹤拒絕服務攻擊的來源的各種方法進行了深入的研究，分析了它們各自的優缺點，並對基本包標記方案作了改進，使得其攻擊路徑重構時的運算量和重構的誤報率在原有基礎上大為降低，達到或超過了其他一些方案的水平。 -
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。 -
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。 -
Taking the network life and the packet drop rate as the performance matrix, the simulation performance of imp - ea is analyzed and compared with other three rooting protocols
以網路壽命和丟包率作為評價指標，對改進的能量多路徑路由協議和其它3種路由協議進行了模擬實驗。 -
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。 -
One of the characteristics of gprs service different from real - time voice service is that it requires very low erroneous frame rate. the accuracy of packet data can be guaranteed through the mechanism of retransmission, which, however, leads to longer transmission delay and low payload in spite of busy channels
Gprs業務不同於實時語音的一個特點是它要求有很低的誤幀率，這可以通過重傳機制保證分組數據的正確，但重傳將使傳輸時延大、通道盡管很忙但有效的負荷卻低。 -
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。 -
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。 -
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。 -
Similar to asama protocol, qasama protocol applies ppo and go algorithm, but qasama protocol also considers the requirements of packet delay, packet drop rate, bit error rate ( ber ), traffic capacity and channel utilization, and integrates these requirements into the slot allocation procedure
在qasama協議中，綜合考慮了分組時延，分組丟棄率，誤碼率，業務容量以及通道利用率方面的要求，並將這些要求與多波束智能天線的特點及自適應時隙分配過程有機地結合起來。
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