The present invention relates to a method of managing congestion in a multi-path network, the network having a plurality of network elements arranged in a plurality of switch stages and a plurality of network links interconnecting the network elements, the method comprising the steps of detecting congestion on a network link, the congested network link interconnecting the output port of a first network element with a first input port of a second network element in a subsequent switch stage; identifying an uncongested network link connected to a second input port of said second network element; and directing future data packets on a route across the multi-path network which includes the identified uncongested network link. Also provided is a multi-path network and an Ethernet bridge or router incorporating such a multi-path network.

A method for transmitting a Voice over Internet Protocol (VoIP) by a wireless LAN Access Point (AP) in a communication system includes when a VoIP frame is input from a terminal, determining whether a buffer for eliminating a jitter generated in a wireless network is used, and storing the VoIP frame in a queue, determining whether the buffer is used, and when the buffer is used, determining a time interval for transmitting the VoIP frame stored in the queue and a service start time, and transmitting the VoIP frame stored in the queue in a wired fashion based on the determined time interval and service start time.

A communication device includes: a plurality of output ports; a plurality of queues in which packets are stored so as to be sorted into groups of packets that are output from an identical output port in an identical time period, from among the plurality of output ports; a plurality of first selectors that respectively corresponds to the plurality of output ports, and each of which switches a queue from which packets that are output from the output port are read, between the plurality of queues each time the time period elapses; and a second selector that switches a first selector from which packets are output, between the plurality of first selectors, at time intervals in accordance with output rates of packets of the plurality of output ports.

A base station, system, method, and computer program product are disclosed for transmitting data packets received by the base station. According to one aspect, the base station includes a first buffer configured to store data packets received by the base station, a second buffer configured to store data packets that are scheduled for transmission by the base station, and a computer processor, operatively coupled to the first buffer and the second buffer. The computer processor is configured to inspect a data packet to determine an application type of the data packet, determine a delay tolerance associated with the data packet based on the determined application type, determine a channel condition of a communication session with the user device, move the data packet from the first buffer to the second buffer based on the determined delay tolerance and the determined channel condition, and transmit data packets stored in the second buffer.

Circuits and methods are disclosed for low-skew bonding of a plurality of data channels into a multi-lane data channel. In one embodiment, phase-measuring first-in first-out buffer circuits buffer pre-buffer parallel data signals and generate phase-measurement signals. A channel-bonding control circuit receives the phase-measurement signals and generates bit-slip control signals. Transmission bit-slip circuits slip integer numbers of bits based on the bit-slip control signals. Bypass registers may be used when the integer number of bits is greater or equal to the parallel width of a lane. In another embodiment, the channel-bonding control circuit receives the phase-measurement signals from the phase-measuring FIFO buffer circuits and generates clock-slip control signals. Clock slip circuits controllably slip parallel clock signals by integer numbers of unit intervals of a serial clock signal. Various other aspects, features, and embodiments are also disclosed.

Systems and methods for multiplexing audio/video data and generating transport streams for WiFi network with reduced latency for real time playback at a remote device. A virtual presentation clock reference (PCR) representing a scheduled transmission time of a transport stream packet at a transport stream multiplexer is calculated based on the network transmission rate and generation of the data packets. The virtual PCR is compared with the corresponding system PCR to derive a time difference. Based on the time difference, the transport stream multiplexer is configured to adaptively drop packets or throttle packet generation so as to synchronize the playback of audio/video data on a sink device with the generation of interleaved audio/video packets.

Methods, devices and systems are disclosed for dynamically adjusting the load priority of a backup nameserver in a computer network based on the health and responsiveness of primary and backup nameservers.

Disclosed is a method and apparatus operative to process packets of media received from a network including a receiver unit operative, a jitter buffer data structure and a playback head defining a point in the jitter buffer data structure from which the ordered queue of packets are to be played back, and at least one prototype head. Each prototype head having a predetermined latency assigned thereto and defining a point in the jitter buffer data structure from which the ordered queue of packets is being played back containing said latency a processor operable to determine a measure of conversational quality associated with the ordered queue of packets being played back by each prototype head. Also described is a head selector operable to compare the measures of conversational quality associated with the ordered queue of packets being played back by each prototype head to select the prototype head with the highest measure of conversational quality and a playback unit coupled to the playback head.

Failure to identify an accurate rising position of a queueing delay alone has led to the inability to ensure estimation precision. An available bandwidth estimation method comprises: transmitting a packet train constituted by packet pairs arranged at equal intervals, each of the packet pairs comprising two packets of the same size arranged at a zero transmission interval, with the packet size of the packet train increasing at a constant rate; receiving the packet train; calculating an observation delay of the packets included in the received packet train; calculating a serialization delay on the basis of the reception interval between the two packets constituting each packet pair included in the received packet train; removing the calculated serialization delay from the calculated observation delay, thereby calculating a queueing delay; and estimating an available bandwidth on the basis of the calculated queueing delay.

The dynamic proportioning of a maximum queue size of a data transport device queue based on throughput parameters may decrease routing latency of a data transport device. A maximum queue size parameter for a data queue may be calculated based on at least a plurality of throughput parameters during routing of data traffic from a data source device to a data recipient device. Subsequently, a maximum queue size of the data queue may be decreased according to the maximum queue size parameter to prevent enqueuing of incoming service frames into the data queue. The lack of enqueueing of the incoming service frames may cause the data source device to retransmit the one or more incoming service frames to the data transport device, instead of allowing the one or more incoming service frames to be enqueued and trapped in the data queue by additional incoming service frames.