2082 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.

Methods, apparatuses and systems directed to facilitating WLAN diagnostics and management using traffic stream metrics. In a data collection stage, according to one implementation of the present invention, localized uplink measurements are taken at a wireless client associated with a wireless access point. During periodic intervals (e.g., every 5 seconds), the wireless client, in one implementation, transmits uplink measurement information to the wireless access point. The wireless access point may also take downlink measurements, which may also include one or more of the following metrics: observed latency, queuing delay, packet loss rate, and packet count information. The wireless access point, in one implementation, may aggregate and report the uplink and downlink metric information to a network management system. In a diagnostic stage, according to one implementation, a wireless network management module performs diagnostics of the WLAN based on the uplink and / or downlink metrics received from the wireless access-point.

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.

A technique for determining when to change a data rate by determining the cause of packet loss. The technique distinguishes between collision mediated packet loss and poor signal mediated packet loss. Rate shifting to a lower rate is performed after determining poor signals are causing packet loss. After switching to a lower rate, the packet loss rate can be compared to the pre-switch packet loss rate. If the packet loss rate has not improved by shifting to a lower data rate, then the data rate can be shifted to a higher rate. The technique can use a combination of channel response, signal strength, packet loss rate and throughput to determine when to switch data rates. A communication unit can maintain separate histories for each unit it is communicating with and employ a different data rate for each unit.

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.

Various embodiments of a network protocol that utilizes a congestion control algorithm that distinguishes between congestion loss and damage loss are described. In response to a packet loss on a network, a delay-based detection algorithm may be performed based on RTT (Round-Trip Time) information to determine whether the network is congested. If the delay-based detection algorithm does not determine that the network is congested then a consistency-based detection algorithm may be performed based on packet loss rate information. If either the delay-based detection algorithm or the consistency-based detection algorithm determine that the network is congested then the rate of data transmission may be reduced, e.g., by reducing a congestion window size.

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.

A method and device for identifying degradation in service quality for a wireless media service. The wireless media service has a known expected packet generation rate, which may be determined by the device based upon the codec and payload size selected on initiating the media service. At a receiving device, the packets actually received by the device from the other termination point are counted over a time period, and this number is compared with the number of packets that were expected based upon the known packet generation rate. From this comparison a packet loss rate may be determined. If the packet loss rate exceeds a threshold level, then the device may issue a user alert, terminate the service, or take other action. The threshold level may be established based upon a packet loss rate corresponding to a significant degree of service quality degradation.

The embodiment of the invention discloses a method and a system for network selection under dual-network coverage, and a dual-mode terminal. The method comprises the following steps that: the dual-mode terminal searches for a wireless local area network (WLAN) signal, and judges whether the intensity of the found WLAN signal reaches a preset intensity threshold value or not; the dual-mode terminal accesses a WLAN in response to that the intensity of the WLAN signal reaches the preset intensity threshold value, and judges whether the intensity of the WLAN signal reaches the preset intensity threshold value or not on a cycle of the first preset time; the dual-mode terminal acquires a weight value based on a packet delay and a packet loss rate in response to that the intensity of the WLAN signal reaches the preset intensity threshold value, and judges whether the weight value is lower than a preset weight value threshold value or not; and the dual-mode terminal is switched from the WLAN to a mobile network in response to that the weight value is not lower than the preset weight value threshold value. By the embodiment of the invention, a wireless network to be accessed by the dual-mode terminal can be rationally selected under the dual-network coverage of the mobile network and the WLAN, the utilization rate of wireless network resources can be increased, and the quality of service of a wireless network service can be improved.

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.

A technique for determining when to change a data rate by determining the cause of packet loss. The technique distinguishes between collision mediated packet loss and poor signal mediated packet loss. Rate shifting to a lower rate is performed after determining poor signals are causing packet loss. After switching to a lower rate, the packet loss rate can be compared to the pre-switch packet loss rate. If the packet loss rate has not improved by shifting to a lower data rate, then the data rate can be shifted to a higher rate. The technique can use a combination of channel response, signal strength, packet loss rate and throughput to determine when to switch data rates. A communication unit can maintain separate histories for each unit it is communicating with and employ a different data rate for each unit.

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 wireless video monitoring system and a method for dynamically regulating a code stream according to network state. The method comprises that: when the wireless video monitoring system performs real-time video data transmission, a mobile video access unit regulates the key frame interval and code rate of the video data adaptively according to the packet loss rate of the video data reported by a mobile terminal before the video data transmitted by the fixed network video monitoring system is forwarded to the mobile terminal. When the system and the method are adopted, a mobile terminal can play real-time video data smoothly in network environments in all states.

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 relates to an internet protocol (IP) over wavelength division multiplexing (WDM) network energy-aware routing method based on multipriority business. The IP over WDM network energy-aware routing method is used for achieving energy saving of an optical network under different QoS constraint conditions of dynamic business. According to the method, the business is divided into different grades. For low-priority business, a Dijkstra algorithm is improved to achieve energy saving by combining an optical bypass and a business volume guidance technology and enabling energy consumption to serve as link weight; and for high-priority business, routing is performed with an optimal path forecasting function serving as a routing standard, a flow partition mechanism and a link weight dynamic adjustment mechanism are integrated while network energy consumption is reduced so as to guarantee the requirements of the business for time delay, packet loss rate and blocking rate. Compared with the prior art, different routing methods can be designed according to priority level of the business, energy-saving efficiency of the network is improved on the premises that a certain service quality is guaranteed, and energy saving of the multipriority business in the IP over WDM network is achieved comprehensively.

The embodiment of the invention discloses a message packet loss detection method and a message packet loss detection system. The message packet loss detection method comprises that: a pseudowire transmitting end transmits a loss measurement message LMM message to a pseudowire receiving end, wherein the pseudowire comprises at least two equivalent paths, and the LMM message comprises first indication information corresponding to each equivalent path; the pseudowire transmitting end receives a corresponding loss measurement reply LMR message transmitted after the pseudowire receiving end sorts the LMM message, wherein the LMR message comprises second indication information corresponding to the first indication information; the pseudowire transmitting end sorts the LMR message according to the second indication information, and acquires the message packet loss rate of the pseudowire according to a value of a counter for the LMR message. Due to the embodiment of the invention, when a plurality of equivalent paths are present in PW, actually transmitted message flux of the PW can be detected. Therefore, the message packet loss rate of the PW is acquired.

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 invention provides a buffer queue management method used in a wireless local area network. In the wireless local area network, when related sites have large quantity and large data amount, network congestion phenomenon can happen. In order to detect the network congestion before the congestion happens or at the beginning period that the congestion happens, the in-queue detection of a data packet is realized by the buffer queue management method of exchanging nodes (such as AP), and a sending end is notified with the congestion by packet loss or setting an explicit congestion notification (ECN) of an IP packet head, so as to relieve the network congestion. The buffer queue management method provided by the invention uses the queue length and flux speed as congestion measurement, so that the invention can evaluate the network congestion degree more exactly, and the invention calculates different packet loss probability for each queue according to the priority difference of buffer queues to realize the random packet loss according to different priorities, so as to achieve the goal of distinguishing packet loss rate.

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.