263 results about "Packet arrival" patented technology

A system and method for synchronizing digital media playback at multiple digital media playback devices interconnected on a network is provided. A digital media playback device comprising a processor, a synchronization component, a timekeeper component and a digital media source performs synchronization processes to arrange for other players to begin playback at a predetermined position and time in the digital media signal. Synchronization is accomplished by processes which approximate the arrival time of a packet containing audio and / or video digital content across the network and instruct the playback devices as to when playback is to begin, and at what point in the streaming media content signal to begin playback. One method uses a time-stamp packet on the network to synchronize all players. Other methods utilize reiterative processes to narrow approximations of packet arrival time at each playback device.

In an improved system for receiving digital voice signals from a data network, a jitter buffer manager monitors packet arrival times, determines a time varying transit delay variation parameter and adaptively controls jitter buffer size in response to the variation parameter. A speed control module responds to a control signal from the jitter buffer manager by modifying the rate of data consumption from the jitter buffer, to compensate for changes in buffer size, preferably in a manner which maintains audio output with acceptable, natural human speech characteristics. Preferably, the manager also calculates average packet delay and controls the speed control module to adaptively align the jitter buffer's center with the average packet delay time.

A system and methods are shown for traffic shaping and congestion avoidance in a computer network such as a data-over-cable network. A headend of the data-over-cable system includes a traffic shaper configured to calculate a packet arrival rate from a cable modem and a traffic conditioner configured to calculate an average queue size on an output interface to an external network. For example, the traffic shaper compares the packet arrival rate to three packet arrival thresholds including a committed rate threshold, a control rate threshold and a peak rate threshold. If the calculated packet arrival rate falls between the committed threshold and control rate threshold, the traffic shaper applies a link layer mechanism, such as a MAP bandwidth allocation mechanism, to lower the transmission rate from the cable modem.

A network monitoring system (10) for monitoring a network along which network traffic flows in a form of packets. The system comprises circuitry (36, 42) for receiving a packet communicated along the network and for determining whether the received packet satisfies a set of conditions. The system further comprises circuitry (36 / 30, 46), responsive to a determination that the received packet satisfies the set, for determining a measure, wherein the measure is determined over a defined time interval and comprises a ratio of packet arrival variance and a mean of packets arriving during the time interval and for comparing the measure to a threshold. Lastly, the system comprises circuitry (36, 52), responsive to the measure exceeding the threshold, for adjusting network resources.

A cable modem communicates the timing of the arrival of certain packets (such as TCP ACK packets) to a cable modem termination system. This timing feature is useful in methods for increasing the throughput in the downstream direction by eliminating TCP ACKs backing up at a cable modem. In particular, a cable modem termination system implementing an unsolicited bandwidth grant service in which unsolicited grants of bandwidth are sent by the cable modem termination system to the cable modem. The grants of bandwidth are timed to arrive at the cable modem simultaneous with or shortly after the arrival of the TCP ACK from an end station connected to the cable modem. Several methods of calculating or predicting when the unsolicited grants should be sent to the cable modem are described. The method is also applicable to other types of packets, such as voice packets from a end station implementing a Voice over Internet Protocol (VoIP) application.

Network testing is conducted using a pair of traffic agents that acts as sender and receiver endpoints. The sending traffic agent generates and transmits a stream of packet batches, each batch being composed of high and low priority packets in a predetermined sending order. The receiving traffic agent analyzes the order of packet arrival in the stream relative to the sending order, and returns results to a traffic control module. Path speed and the priority classification of the traffic are parameters used to configure the tests. The tests are typically run periodically under different load conditions in order to evaluate the effects of other traffic being concurrently transiting the network.

A network monitoring device configured to collect a new packet from one or more observation points of a network and to compare the new packet with a list of a number of received packets based on a packet arrival rate and to identify a duplicate packet. In particular, the number of received packets in the list is equivalent to a number of packets received within a time period, i.e. the packet arrival rate. Stated differently, the network monitoring device is to compare the new packets with received packets stored in a queue of a buffer and wherein the queue has a size based on a packet arrival rate collected at one or more observation points. In addition, the time period is further adjusted according to a threshold value. The threshold value is a variable parameter that can be adjusted to compensate for different network deployment. In one embodiment, the threshold value is a time value that is not more than a transmission time of a TCP retransmitted packet.

An architecture for capture and generation, and a set of methods for characterization, prediction, and classification of traffic in packet networks are disclosed. The architecture consists of a device that stores packet timing information and processes the data so that characterization, prediction, and classification algorithms can perform operations in real-time. A methodology is disclosed for real-time traffic analysis, characterization, prediction, and classification in packet networks. The methodology is based on the simultaneous aggregation of packet arrival times at different times scales. The traffic is represented at the synchronous carrier level by the arrival or non-arrival of a packet. The invention does not require knowledge about the information source, nor needs to decode the information contents of the packets. Only the arrival timing information is required. The invention provides a characterization of the traffic on packet networks suitable for a real-time implementation. The methodology can be applied in real-time traffic classification by training a neural network from calculated second order statistics of the traffic of several known sources. Performance descriptors for the network can also be obtained by calculating the deviation of the traffic distribution from calculated models. Traffic prediction can also be done by training a neural network from a vector of the results of a given processing against a vector of results of the subsequent processing unit; noticing that the latter vector contains information at a larger time scale than the previous. The invention also provides a method of estimating an effective bandwidth measure in real time which can be used for connection admission control and dynamic routing in packet networks. The invention provides appropriate traffic descriptors that can be applied in more efficient traffic control on packet networks.

An aggregation scheduler that determines which frames are to be aggregated in an aggregate at a specific time, and allocates a time and duration for each responder of the aggregate. The AP maintains information for each of its clients, including whether the client is in an awake or doze state and the time stamp arrival (TSA) for each frame received from the client by the AP. When an AP obtains channel access, it only aggregates frames for those clients that are awake. From the TSA and the voice packet arrival interval for the client, the AP can compute how many voice frames are waiting at the client since the AP received the last frame from the client. Knowing the number of voice frames waiting at the voice client and the transmission rate, enables the AP to accurately ascertain the response time and duration for the voice client.

A voice-over-Internet-Protocol (VoIP) application estimates bandwidth and congestion of the reception path to the VoIP application from a sending VoIP application. Packet arrivals are timed and the inter-packet delay is compared to the voice duration of the data contained in the more recent packet. When the inter-packet delay is longer than the voice duration the network is slowing and the bandwidth estimate is reduced. The bandwidth estimate is increased when inter-packet delay is smaller than the voice duration. Packet latencies are the difference in send and receive times and are compared to a moving average latency. When the current packet's latency is longer than the moving average, congestion is detected. When the current packet's latency equals the moving average, the network has recovered from congestion and the congestion estimate is reduced. Congestion and bandwidth estimates are added to packets sent out to provide feedback to the other VoIP application.

A wireless data communication system has a first station or mobile unit is linked to a second station configured as an access unit to support packet communication, voice or data, where the voice packets are transmitted in the Continuously Aware Mode (CAM) mode while other packets are buffered by the access point and held until asked for by the first station when in a Power Saving-Poll (PSP) mode. A monitoring apparatus at the access point monitors all transmitted packets and sorts the packets to the mobile unit according to CAM or PSP mode. Voice packets are sent out immediately to the mobile unit. Other packets are stored at the access point. The packet arrival rate may vary during transmission and due to random packet delays introduced by propagation characteristic and processing apparatus. The packet arrival rate and delays are taken into account by the first station in an algorithm to determine and extend the normal safe period in which the station receiver may be powered off.

Classification of packets into flows is an inherent operation performed by networks that support enhanced services. To support multiple-dimensional packet classification, a packet classification system is provided to select representative bits from a packet to look up a set of rules. The per-flow classification works with a large set of rules, where each rule comprises of multiple fields and also allows fast dynamic variation in the rule set. A lookup process includes a simple and finite set of instructions that can be efficiently implemented as pipelined hardware and support very high packet arrival rates.

A method and apparatus for bandwidth reservation for data flows in interconnection networks. Some embodiments of an apparatus for transmitting a data stream include a transmitter to transmit a data stream to a recipient apparatus, the data stream including a plurality of data packets. The apparatus further includes a receiver to receive a response from the recipient apparatus regarding data packet arrival status, and a network unit to direct the operation of the transmitter, the network unit to direct the transmitter to maintain the data stream with a constant bandwidth.

When path status information is updated, the time at which the update is effective is recorded. A packet arrival time in each path is predicted based on new status information and data transmission history after the effective time. The packet is transmitted to the path that provides a fastest packet arrival time. This reflects the path status available from the reception side on the data transmitted in past, so that an actual data arrival time, an arrival time close to a reception completion time, or a reception completion time can be predicted.

In a duplex radio link digital data information is transmitted to a remote terminal at a constant symbol rate in accordance with a selected data rate mode that is a function of direct sequence spreading gain, error correction code rate, and signal constellation type. The data rate is adapted by selecting a data rate mode that is a function of a data packet arrival rate and a link quality measure fed back from the remote terminal. The data packet arrival rate is controlled as a function of the link quality measure and the current data packet arrival rate. In systems with multiple transmit diversity channels, independent data is sent over each of the transmit diversity channels. In an idealized feedback communication example, a single antenna troposcatter system in a Ku-band application is shown to have 15.5 times the data rate capability of a conventional two-antenna system at S-band.

A network system (10), comprising a plurality of nodes (ERx, CRx). Each node in the plurality of nodes is coupled to communicate with at least one other node in the plurality of nodes. Each node of the plurality of nodes comprises a plurality of queues (32x) and is operable to perform the steps of receiving a plurality of packets and, for each received packet in the plurality of packets, coupling the received packet into a selected queue in the plurality of queues, wherein a respective selected queue is selected in response to the respective received packet satisfying one or more criteria. Each node of the plurality of nodes is also operable to perform the step of assigning a weight (Wx) to each respective queues in the plurality of queues. Each weight assigned to a respective queue in the plurality of queues is responsive to quality requirements for each packet in the respective queue and to a ratio of packet arrival variance in the respective queue and a mean of packets arriving to be stored in the respective queue during a time interval for minimizing the overall network traffic burstiness.

A packet schedule timestamp is provided. A schedule information packet includes a packet schedule timestamp that represents the relative timing of data packets in a particular bitstream of a multiplexed transport stream. The schedule information packet further provides information about the timing of packets within a bitstream or multiple bitstreams before actual packet arrivals. Look-ahead processing techniques are provided to improve the efficiency of compressed bitstream storage and delivery, such as program stream to transport stream conversion and statistical multiplexing. The packet schedule information can be inserted into the bitstream it describes for transmission or storage with it. Other features, such as timing reconstruction, are also provided. The packet schedule information can be used to reconstruct the timing of the transport packets to their original state.

The present invention broadly contemplates addressing QoS concerns in overlay design to account for the last mile problem. In accordance with the present invention, a simple queuing network model for bandwidth usage in the last-mile bottlenecks is used to capture the effects of the asymmetry, the contention for bandwidth on the outgoing link, and to provide characterization of network throughput and latency. Using this characterization computationally inexpensive heuristics are preferably used for organizing end-systems into a multicast overlay which meets specified latency and packet loss bounds, given a specific packet arrival process.

Content is streamed from a providing device to a receiving device, where the content is successively buffered in a receiving buffer, and then successively removed for processing. Periodically, the likelihood of the receiving buffer becoming empty or reaching an unacceptable low threshold is assessed, and the streaming rate is adapted based on the result of the determination. In one embodiment, a probability is computed, modeling the packet arrival into the receiving buffer and removal as Markovian input and output processes respectively, and the receiving buffer as a single channel queue. In one embodiment, the likelihood is determined by the content providing device, based at least in part on feedback information from the content receiving device, and the streaming rate is stochastically adapted accordingly.

Network testing is conducted using a pair of traffic agents that acts as sender and receiver endpoints. The sending traffic agent generates and transmits a stream of packet batches, each batch being composed of high and low priority packets in a predetermined sending order. The receiving traffic agent analyzes the order of packet arrival in the stream relative to the sending order, and returns results to a traffic control module. Path speed and the priority classification of the traffic are parameters used to configure the tests. The tests are typically run periodically under different load conditions in order to evaluate the effects of other traffic being concurrently transiting the network.

A closed-loop voice-over-Internet-Protocol (VoIP) system has a local and a remote VOIP application. Each VOIP application monitors incoming packet arrival times and durations of audio data in the incoming packets to estimate bandwidth. The bandwidth estimates are forwarded to the other VOIP application. The forwarded bandwidth estimates are compared to a sending bandwidth. When the bandwidth estimate is above the sending bandwidth, compression and audio-frame decimation are reduced to improve voice quality. When the bandwidth estimate falls below the sending bandwidth, audio compression and decimation are increased to improve efficiency. Packet size can also be increased. Congestion estimates can also be sent with the audio data, causing packet transmission to pause until congestion ends. Incoming packet latencies are compared to a moving average to determine the congestion estimate, while bandwidth estimates are made by comparing packet audio duration to time between packet arrivals.

A method, system and computer program product for detecting conditions of network instability. An attribute indicative of network instability, e.g., processor and / or co-processor utilization, packet arrival rates, packet peak rates, packet size distribution, packet clustering tendencies, buffer usage patterns, occurrence of peak utilization, out-of-buffer conditions, packet discard rates, may be monitored for a network device, e.g., router. The monitored attribute may be associated with a plurality of labels (variables) where a portion of those labels may be stored in a cache. A hit ratio for the cache storing labels associated with the monitored attribute may be tracked within a period of time based on the number of requested items, i.e., labels, that are currently stored in the cache. A condition of network instability may then be detected based on the hit ratio by determining if the absolute value of the acceleration of the hit ratio exceeds a threshold.

A technique for timing compensation is used in a terrestrial wireless communication system (300) that has a satellite backhaul link (352, 358, 360) to at least one base transceiver station (306, 307, 308). The technique includes establishing a backhaul delay (BHD) of the satellite backhaul link and performing at least one timing compensation function based on the backhaul delay. The technique further includes setting (540, 545) a base controller system time (170). The following timing compensation functions are described: adjustment of packet arrival timing error interval (520), selection of a mobile station power control outer loop path (525), adjustment of at least one protocol timer (530), evaluation of a reverse Markov test call frame based on the BHD and real time (545), adjustment of forward data frame alignment based on the BHD and real time (555), and adjustment of forward Markov test call frame generation based on the BHD and real time (560).

The invention provides a media access control (MAC) method applicable to wireless sensor network, which can adjust the sending process according to the characteristic that the data traffic of the node is time-varying in the specific application of the wireless sensor network. The MAC method comprises the following steps: estimating the current data traffic intensity by measuring the time intervals of data packet arrivals, and adopting a filter to reduce the interference in the measurement; carrying out the self-adaptive control by detecting the queue length of the current data packet of the node, if the queue length of the data packet is larger than 1, batch-sending the data packets in the queue in a burst-transmission manner so as to reduce delay and energy consumption, and if not, judging by the currently measured data traffic; and if the current data traffic value is smaller than the given threshold, enabling the low-traffic self-adaptive mechanism, and if not, removing unnecessary wake-up cycles by the node. The method dispenses with the on-line adjustment of the cycle and the duty cycle of the protocol, avoids the problem of the difficult synchronization between nodes and effectively solves the major problems of the synchronization and the data traffic self-adaption in the existing wireless sensor network.

The invention discloses a method for regulating a discontinuous reception cycle. The method comprises the following steps of: determining a service activity by a node B evolved base station (eNB) through a data packet arrival rate of service data; reporting mobile state information to the eNB by user equipment (UE); and configuring a discontinuous reception (DRX) cycle by the eNB depending on the service activity and the mobile state information. Through the application of the embodiment disclosed by the invention, the DRX cycle is precisely configured, waste of UE power is reduced and switching performance is improved.

The invention provides a semi-persistent scheduling (SPS) resource allocation method and a base station. The SPS resource allocation method is that the base station receives data packets sent during voice communication of UE (user equipment); the base station determines MCS (modulation and coding scheme) level and determines packet information of the data packets according to the received data packets, particularly, the MCS level is determined by the base station according to CQI (channel quality indicator) of a channel which is used for the UE to send the data packets, and the packet information contains packet size; further, the base station determines data volume of each SPS according to the first formula that the data volume of each SPS=min{preset maximum data volume of the SPS, the packet size*[SPS cycle / data packet arrival cycle]}; furthermore, the base station determines quantity of PRB (physical resource blocks) according to the data volume of each SPS and the MCS level so as to allow the UE to transmit the data packets according to the quantity of the PRB. By the SPS allocation method and the base station, time delay of the data packets can be reduced while voice service quality can be improved.