66 results about "Latency jitter" 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.

A high accuracy clock synchronization mechanism between a sender and a receiver in a communication network achieves time synchronization using broadcast beacons, directly at the PHY/MAC layer of the sender and the receiver, to minimize synchronization delay jitter. This provides a more efficient synchronization method than either NTP or SNTP, because multiple handshaking information exchange is avoided. Further, using beacons avoids the overhead of introducing additional synchronization packets in higher layer synchronization.

A method is described for measuring system latency and latency jitter on a near real-time basis regarding packets of information moving from sources to destination. The method for measuring system latency and latency jitter includes time-stamping packets of information as they pass through a plurality of locations between the source and destination of these packets, with the time-stamping being performed by real-time clocks set to a universal standard. The time-stamped packets are analyzed to determine latency and latency jitter. The measurements of system latency and latency jitter can be performed based upon packet types so as to obtain accurate information concerning different types of business bandwidths associated with an observed network.

This invention describes a method for transmitting and forwarding packets over a switching network using time information. The network switches maintain a common time reference, which is obtained either from an external source (such as GPS—Global Positioning System) or is generated and distributed internally. The time intervals are arranged in simple periodicity and complex periodicity (like seconds and minutes of a clock). A data packet that arrives to an input port is switched to an output port based on its order or time position in the time interval in which it arrives at the switch. The time interval duration can be longer than the time duration required for transmitting a data packet, in which case the exact position of a data packet in its forwarding time interval is predetermined.

This invention provides congestion-free data packet switching for data packets for which capacity in their corresponding forwarding links and time intervals is reserved in advance. Furthermore, such data packets reach their destination, which can be one or more (i.e., multicast) in predefined time intervals, which guarantees that the delay jitter is smaller than or equal to one time interval.

A method for estimating end-to-end quality of service (QOS), such as packet loss, delay, and delay jitter, in a packet-switched communications network includes steps for calculating packet loss and packet delay each router output link in a network path and using the packet loss and packet delay calculations for each individual router output link to estimate end-to-end QOS. The method includes calculating the loss probability for all possible numbers of active sources to take into account changes in the number of active sources as active sources connect to and disconnect from the network. As a result, the inventive method allows the number of active sources to vary in its estimation of end-to-end packet loss, delay, and delay jitter for enhanced Internet Protocol network planning.

A pull-based parallel video server system and method of implementation useful in an on-demand video system includes a plurality of slave admission schedulers operating in parallel with a master admission scheduler to back up the master admission scheduler that controls access to an array of pull-based video servers according to a protocol that accounts for jitter and loss of packets, as well as network delays. A transport protocol is provided that improves video data throughput under such conditions. To determine the architecture and functional requirements of the redundant admission schedulers, an analytical tool in the form of a performance model has been developed that incorporates network delays, delay jitters and packet losses on the communication links between the clients, schedulers and servers.

Embodiments of the present invention provide a method, system and device for clock transmission between a sender and a receiver. The sender generates clock information of a clock to be sent relative to a system clock and sends a data packet containing the clock information to the receiver over a Packet-Switched Network (PSN). The receiver obtains the clock information in the data packet received and obtains the clock sent by the sender according to the clock information and the system clock. According to the embodiments of the present invention, after the clock of the sender is transmitted over the PSN, the receiver may obtain the clock of the sender without being affected by such damage as a network delay jitter and a packet loss.

A delay unit 103 adds holding time that has been set by a holding time setting unit 104 to a received data. The holding time is computed based on delay time of received data and the minimum delay time of data received up to a certain point for the purpose of reducing a total delay time. The delay time is estimated in a delay time estimating unit 106 from the difference between a reception time of a packet counted based on an internal clock generator 107 and a time designated by a time stamp in the received packet.

Described are a device and method used for controlling a jitter buffer. In one execution mode, the device used for controlling the jitter buffer includes an inter-voice-segment delay jitter estimator which is used for estimating an offset value of a delay of a first frame in a current voice segment relative to a delay of a last anchor frame in a prior voice segment; and a jitter butter controller which is used for adjusting the length of the jitter buffer on the basis of a long-term length of each frame of the jitter buffer and the offset value.

The invention discloses a clock synchronization method and device, a computer storage medium and electronic equipment, and the method comprises the steps: achieving the determination of time deviationthrough the receiving and transmitting of a time stamp of a conventional message, and judging whether to carry out the time synchronization or not according to the time deviation through network delay. Furthermore, the excessive delay jitter is judged through the network delay, so that the influence of the excessive delay jitter on the time correction of the time service end is avoided.

A method is described for measuring system latency and latency jitter on a near real-time basis regarding packets of information moving from sources to destination. The method for measuring system latency and latency jitter includes time-stamping packets of information as they pass through a plurality of locations between the source and destination of these packets, with the time-stamping being performed by real-time clocks set to a universal standard. The time-stamped packets are analyzed to determine latency and latency jitter. The measurements of system latency and latency jitter can be performed based upon packet types so as to obtain accurate information concerning different types of business bandwidths associated with an observed network.

A system and method for guaranteeing a delay jitter bound when scheduling bandwidth grants for voice calls via a communication medium is provided. The method includes the steps of: determining the delay jitter bound; based on the determined delay jitter bound, dividing a packetization frame period into phases; assigning a voice call to one of the phases; and scheduling a bandwidth grant to the voice call during the assigned phase, thereby guaranteeing the delay jitter bound. The system includes a scheduler, where the scheduler determines the delay jitter bound, divides a packetization frame period into phases based on the determined delay jitter bound, assigns a voice call to one of the phases, and schedules a bandwidth grant to the voice call during the assigned phase, thereby guaranteeing the delay jitter bound. A dejitter buffer implements a way to provide zero jitter service, even though the packet transmission on the cable network has jitter, by delaying the packet and thus converting jitter into delay.

The invention relates to a sampling value synchronization interpolation algorithm with data delay jitter or loss compatible. The sampling value synchronization interpolation algorithm is applied to a protecting, measuring and controlling system which comprises a sampling unit, a merging unit and a function unit. The sampling value synchronization interpolation algorithm can be applied to two scenes: (1) when the merging unit receives a sampling value sent from the sampling unit and the sampling value is initially integrated, the merging unit conducts primary interpolation replenishing and primary interpolation calculating on the sampling value, and the sampling value after being initially integrated is established; (2) when the function unit receives the sampling value sent from the merging unit after initial integration and the sampling value is integrated once again, the function unit conducts secondary interpolation replenishing and secondary interpolation calculating on the sampling value, and the sampling value after being integrated once again is established. According to the sampling value synchronization interpolation algorithm, the fact that the sampling unit and the merging unit, and the merging unit and the function unit choose sampling points according to needs in the protecting, measuring and controlling system can be achieved, when data delay jitter or loss occurs, supplement and calibration can be used, and high efficiency and flexibility are achieved.

The embodiment of the invention provides a method and apparatus for delay jitter elimination during a message transmission process. The method comprises the following steps that: before stamp taking is carried out on a target message, the bit of the target message is increased and the length of the target message is adjusted , so that the length of the target message reaches a preset one; stamp taking is carried out on the target message at a stamp taking point so as to obtain a first timestamp of the target message; whether an indication code of the target message is received is detected at the stamp taking point; if no indication code is detected, a current counting value of a first counter is obtained and is used as a target bit number, wherein a bit number is subtracted from the first counter after one work cycle each time and is also one going out of a stack outgoing point in the work cycle; and according to the target bit number and the first timestamp, a second timestamp at the stack outgoing point at current moment is obtained. According to the invention, the length of the target message is adjusted before stamp taking and the to-be-transmitted bit number in a protocol stack is measured by the counter, thereby realizing delay jitter correction and eliminating the delay jitter during the transmission.

In a multiple access system, transmit authorizations are transmitted at a high rate in order to fulfil the requirements relating to the delay jitter. However, the transmit authorizations are transmitted only as long as is necessary. Following the reception of a data packet of a terminal, the transmission of further transmit authorizations for the same terminal is interrupted. Only after a specific time period, i.e. shortly before the expected transmission of a data packet of the same terminal, are transmit authorizations again sent to this terminal at a high data rate. In this way short time periods, in which transmit authorizations are sent to a terminal at a high rate, alternate with long time periods in which no transmit authorizations are sent to this terminal. In the long time periods, for example, transmit authorizations can be sent consecutively, in each case in short time periods, to one, two or more further terminals. In this way the bandwidth utilization is optimised.

The invention relates to a multipath congestion control method and device based on time delay and window jitter. The multi-path congestion control method based on time delay and window jitter comprises the following steps: obtaining a characteristic value R (k) of current network delay jitter according to a product of a difference value between data round-trip time delay and a data round-trip timedelay average value on a current transmission path and a first weight value; obtaining the average value of the size of the congestion window on the current transmission path and the sizes of the previous n congestion windows, and obtaining the characteristic value w (k) of the jitter of the current congestion window according to the product of the difference value between the congestion window value and the average value of the previous n congestion windows and the second weight value; determining the current packet loss type on the data transmission path by using the result of the sum of R(k) and w (k); and adjusting a congestion window corresponding to the data sending path according to the packet loss type. According to the multi-path congestion control method based on the time delayand the window jitter, the effective throughput of the MPTCP of the mobile terminal is improved.