2070 results about "Packet loss rate" patented technology

A method for providing network access to a shared access communications medium for a plurality of users includes the steps of conducting predictive admission control by arbitrating user requests for access to the shared medium based on predicted aggregate demands, conducting lookahead scheduling for use in making user channel assignments by forecasting schedule transmission opportunities one or more channels of the shared medium, and balancing load by making channel assignments such that a plurality users are each assigned a respective channel of the shared medium based upon a predicted need. Congestion parameters can predicted for each channel of the shared medium and mapped to a congestion measure using a mathematical function that takes into account packet loss rate, packet delay, packet delay jitter, and available capacity.

In a system including a load balancer to select a virtual server to which a request is to be transferred, the load balancer includes a function to monitor resource use states of physical and virtual servers and a function to predict a packet loss occurring in a virtual switch. The request count of requests processible by each virtual server is calculated based on the resource amount available for the virtual server and a packet loss rate of the virtual switch, to thereby select a virtual server capable of processing a larger number of requests.

A method and apparatus for determining what level of performance must be maintained by a second communication service so as to be perceived as being of equal quality to a first communication service, especially where the second communication service is subject to additional impairments not experienced by the first communication service. In one example embodiment, the present invention is applied to determining what combinations of packet loss rate and packet delay are tolerable where a packet switched telephony service is desired to be perceived to be substantially equivalent to a traditional toll-quality non-packet-switched telephone service. Empirically derived models are used to relate user perception to objectively measurable characteristics, such as packet loss rate.

The present invention relates to the field of software defined network, and particularly relates to an SDN load balancing method with highest network utility. The method comprises the steps of determining that multiple paths between a source node and a destination node are available; assigning different weights to impact factors meeting service quality according to different data flow service attributes; then collecting information of a switch to acquire values of impact factors of service quality in the network, wherein the information comprises link use ratio, time delay and packet loss rate; calculating a utility value and a load of each path in a path set; and finally, selecting an optimal path for the data flow according to overall consideration. Through adoption of the SDN load balancing method with highest network utility, network dynamic load is balanced, further, the service quality demands are met to the greatest extent.

A client device presents streaming media and includes a stream manager, a request accelerator, and a source component coupled to the stream manager and the request accelerator for determining which requests to make. A rate selection process can make rate decisions so that the buffer is filled when it is low, avoiding erratically changing rates and can choose the correct steady rate quickly. Multimedia download strategies can be used for HTTP that allow for accurate rate estimations, achieving link capacity even if network delays and packet loss rates are high, achieving timely delivery of the stream, and achieving relatively steady download rates with little short term variability. A receiver might use multiple HTTP connections, decompose media requests into smaller chunk requests, synchronize the connections using TCP flow control mechanisms, and request data in bursts. In addition, the receiver might use an HTTP pipelining process to keep the connections busy.

A method and apparatus for estimating packet loss rate are described including calculating a real packet loss rate in a time slot at the end of the time slot, estimating average packet loss rate for a subsequent time slot, estimating variance of packet loss rate for the subsequent time slot and estimating the packet loss rate for the subsequent time slot. A method and apparatus and also described for dynamically allocating available bandwidth for video multicast including selecting an intra-frame rate, determining a packet loss rate threshold, receiving user topology information, receiving channel conditions for each user, determining an optimal operation point for encoding and transmitting video frames in a subsequent time slot, adapting dynamically quantization parameters and a forward error correction code rate, encoding the video frames using the quantization parameters and applying forward error correction code with the forward error correction code rate to data packets of the video frames to generate forward error correction packets.

Embodiments of the present invention provide a video data transmission processing method, a video data sending processing method, an apparatus, and a network system. The data transmission processing method includes: receiving a source stream sent from a source transmission network to a target transmission network; performing, according to respective packet loss rates of the source transmission network and the target transmission network as well as error tolerance aid information corresponding to the source stream, error tolerance coding processing on the source stream to obtain an error tolerance stream; and sending the obtained error tolerance stream to the target transmission network.

A VoIP call signaling method including a codec selection mechanism that measures the effective data transmission rate and other network characteristics. In some embodiments, a proxy performs a network test between itself and both end terminals to quantify the quality of the link before notifying both of the endpoints the optimal codec they should use for a peer-to-peer call. In alternative embodiments, the proxy informs the endpoints of the network characteristics and allows the endpoints to negotiate the codec selection. The network test preferably identifies delay, bandwidth, jitter and packet loss rates.

The invention relates to a wireless video streaming service self-adaption rate control method based on the QoE. The method includes the steps of setting up a QoE evaluation model at a receiving end to be used for calculating the quality of user experience, periodically feeding the packet loss rate, the end-to-end one-way time delay and user experience quality information which are obtained at the receiving end back to a sending end through a real-time transmission control protocol, conducting subdivision on network states through the sending end according to the user experience quality and by combining the packet loss rate with the increase or decrease trend of the end-to-end one-way time delay, and judging the network congestion degree. When the user experience quality decreases to a threshold value, the sending end of the wireless video streaming service starts a coding bit rate adjustment unit and adjusts the coding bit rate in a self-adaption mode by taking corresponding strategies according to the monitored network congestion degree, and therefore the user experience quality is improved. According to the method, the user experience quality of the video steaming service can be accurately evaluated in real time, network congestion is reduced through the self-adaption rate control method, and the user experience quality is effectively improved.

The invention discloses a method, system and device for detecting packet loss rate. The method comprises that a head node transmits performance detection packet through a detected LSP, wherein a first statistic value of data packet transmitted to the LSP from a transmitting time is carried; a tail node receives performance detecting packet, and then obtains a second statistic value of data packet received from the LSP; the tail node returns the performance detecting response packet carrying the first statistic value and the second statistic value, the head node calculates packet loss rate in current monitoring time range according to received at least two performance detecting response packets; or the tail node calculates packet loss rate in current monitoring time range according to the first statistic value and respectively corresponding second statistic value included in the received at least two performance detecting response packets, and transmits to the head node. By using the method and system, packet loss rate of each LSP in an MPLS network can be conveniently detected, healthy condition of the LSP can be real-time monitored, so as to know changes in time and provide basis for more accurate switching adjustment.

A coding method to code original data and transmit the coded data as a plurality of data packets to a destination, the coding method comprising: receiving packet loss information associated with the transmission of at least some of the plurality of data packets, the packet loss information including a lost packet restoration rate, a continuous packet loss rate and a packet loss rate of the packets transmitted to the decoding device; and generating an error-correcting redundant packet and inserting the error-correcting redundant packet in subsequent ones of the plurality of data packets for transmission to the destination based on the received packet loss information.

The invention relates to the field of network communication safety of the internet of vehicles, in particular to a malicious node detection method based on clustering trust evaluation in the internet of vehicles. The malicious node detection method comprises the following steps of clustering and cluster head selection; cluster head node trust evaluation; trust evaluation on member nodes in clusters; cluster trust evaluation and node trust update. A network is divided into a plurality of clusters, intra-cluster nodes are communicated, and different clusters are communicated through cluster heads. The method can adapt to a large VANET network and is good in extendibility, and the communication cost of different intra-cluster member nodes is reduced. In addition, a recommendation chain on the aspect of recommendation trust computation is simplified, and the computation is performed by directly using packet loss rate.

The embodiment of the application discloses a service switching method, a context migration method and related equipment for ensuring the continuity of services. The method disclosed by the embodiment of the invention comprises the following steps: a target distributed control unit (DU) receives a switching request message sent by a centralized control unit (CU), and the switching request message carries at least one of the following information: the switching of a URLLC service for indicating the target DU to perform near-end deployment, a context migration protocol supported by a URLLC service server for indicating a source DU to perform near-end deployment, the information for indicating delay requirements corresponding to the URLLC service, and the information for indicating packet loss rate requirements corresponding to the URLLC service; the target DU sends a switching request response message to the CU, and the switching request response message is used for the CU to indicate UE to initiate an access to the target DU; and the switching request response message carries at least one of the following information: an IP address of a target URLLC service server, an IP address of a target local gateway, instructions for supporting the switching and a negotiated context migration protocol of the URLLC server.

The invention discloses a congestion control method and system based on deep reinforcement learning. The congestion control method includes: firstly, initializing the environment and model parametersof a network; and training a congestion control model by utilizing the collected current window, the throughput, the time delay, the data transmission rate and the like in the network, selecting the congestion control model with the minimum model loss function value and the maximum reward function value according to a training result, and deploying the model into the network to perform congestioncontrol. According to the method, the size of the congestion window is dynamically adjusted according to the current network throughput, round-trip time delay and data packet loss rate, so that the data transmission rate is controlled, the network throughput is improved, the data transmission delay and the data packet loss rate are reduced, the network congestion is reduced, and the aim of optimizing the network performance is fulfilled.

The invention provides a control method and system for adaptive adjustment of video data transmission. According to the control method and system, a receiving end forecasts the delay time for transmission of the next data packet through Kalman filtering, and provides a suggested sending code rate for a sending end; and meanwhile, the sending end combines sending code rates which are calculated according to a packet loss rate, the suggested sending code rate and a TFRC algorithm, so as to calculate an adjusted code rate for data packet sending. By adoption of the control method and system, the sending code rate of the next data packet is forecast through Kalman filtering, and the sending end dynamically adjusts the call quality under various adaptive network conditions according to the data receiving condition of the receiving end, that is, the image fluency can be improved under a good network condition and can be reduced under a poor network condition. Therefore, the purpose of real-time congestion control is fulfilled, and the probabilities of intermittence and other abnormal conditions are reduced.

The packet loss in a wired section is reported to a receiving terminal (61) using a loss notification packet from a gateway (62) located at the boundary between a wired section and a wireless section, and thus the causes for packet loss are separated. Also, using this loss notification packet, the packet loss rate in the wired section and the wireless section are separately calculated and the transmission rate and robustness of the data packets are determined with a sending terminal (60) in accordance with the respective loss rate of the sections.

A system for testing a segment of a data-packet-network has a first probe connected substantially at one end of the segment; a second probe connected substantially at the opposite end of the segment from the location of the first probe; and a process application distributed to each probe. The first and second probes collect data from and time stamp data packets as they pass forming first and second records of the individual packets whereupon the second-formed records of each packet are compared with the fist records of each packet for record matching, time-stamp comparison and test result processing.

The invention provides a method and a system for regulating a jitter buffer. The method for regulating the jitter buffer comprises the following steps: acquiring the packet loss rate and the time delay of a data transmission receiving end; regulating the size of the jitter buffer of the receiving end by a preset period according to the packet loss rate and the time delay. Jitter buffer self-adaptation can be carried out according to network conditions, and data transmission quality can be ensured when a network delay jitter is solved. The method and the system are particularly suitable for the transmission of network video data.

A sending probe sends a quality measurement packet to a receiving probe, and increments a sent packet counter every time the quality measurement packet is sent. A receiving probe receives the quality measurement packet from the sending probe, and attaches a received packet counter to the quality measurement packet and returns the quality measurement packet to the sending probe. The sending probe receives a returned quality measurement packet, extracts a received packet count from the quality measurement packet, and calculates a packet loss rate by using a difference between the sent packet counter and the received packet counter and a difference between the received packet counter and the returned packed counter.

The invention discloses a wireless access point and a radio-frequency emission power regulation method thereof. In the method, the wireless access point is used for continuously monitoring packet loss rate of a workstation which is connected with the wireless access point; and when the packet loss rate is less than a first threshold value, the radio-frequency emission power is reduced by the wireless access point step by step. By using the invention, on the premise that data pass between the wireless access point and the workstation is not affected, radio-frequency emission power can be reduced to the largest extent and the whole energy consumption of the wireless access point is reduced effectively.