127results about How to "Packet loss" patented technology

An alarm system (100), including ease of programming of a family or group of interoperating alarm devices (102) via a learn mode that detects new devices and provides reliable accounting of the group via state dumps to an external system (104). Reliable communications with the external system (104) are provided via a set of protocols. Disabling of the alarm system is prevented, by transmitting a pre-alarm signal prior to expiration of an entry delay, and by verifying communications with an external device, prior to an alarm-triggering event. Multi-priority message code assignment, including error tolerance, employs n-bit codes with maximized error tolerance. Message transmissions include multiple levels of error protection. The group of monitored alarm devices (102) can be easily set up, purchased and activated by a consumer, and do not become permanent fixtures.

A method is provided for seamless roaming over multiple dissimilar wireless networks leveraging a set of one or more SDRs, and possibly some traditional radio communication devices. The method includes determining whether alternate networks are available regardless of whether the SDR has only a single transmitter and receiver or multiple transmitters and receivers. In addition, if the SDR provides multiple transmitters and receivers, or if multiple distinct SDRs are available to the mobile computing device, the method will be able to better optimize the seamless roaming experience for the user while still utilizing the set of SDRs, in part, to determine whether alternate networks are available. The method further includes managing the transitions of an SDR between various modulation algorithms according to a variety of states and settings of the SDR or of the set of SDRs as a whole.

An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.

Improved data transport and management within a network communication system may be achieved by utilizing a transmit timer incorporated within the sender device and exploiting host-level statistics for a plurality of connections between a sender and receiver. The period of the transmit timer may be periodically adjusted based on a ratio of the smoothed round-trip time and the smoothed congestion window, thereby reducing or eliminating bursty data transmission commonly associated with conventional TCP architectures. For applications having a plurality of connections between a sender and a receiver that share a common channel, such as web applications, the congestion window and smoothed round trip time estimates for all active connections may be used to initialize new connections and allocate bandwidth among existing connections. This aspect of the present invention may reduce the destructive interference that may occur as different connections compete with one another to maximize the bandwidth of each connection without regard to other connections serving the same application. Error recovery may also be improved by incorporating a short timer and a long timer that are configured to reduce the size of the congestion window and the corresponding transmission rate in response to a second packet loss with a predefined time period in order to increase resilience to random packet loss.

System and method for load-balancing a plurality of transmission media. A first plurality of packets of a first stream may be transmitted to a second device on a first transmission medium. It may be determined that current medium utilization of the first transmission medium exceeds a first threshold. The first stream may be selected for transmission on both of the first transmission medium and a second transmission medium based on said determining that current medium utilization of the first transmission medium exceeds the first threshold. A second plurality of packets of the first stream may then be transmitted to the second device using both the first transmission medium and the second transmission medium. A first portion of the second plurality of packets may be transmitted on the first transmission medium and a second portion of the second plurality of packets may be transmitted on the second transmission medium.

A method supports scalable and reliable multicast in a wireless network with a large bandwidth-delay product. In this method, acknowledgement packets from different receivers experiencing the same number of data packets lost are assigned the same time slots. This method can be combined with other loss recovery techniques, such as forward error correction (FEC) recovery, proactive protection, feedback suppression and collision detection. Scalability is achieved as bandwidth usage relates only to the number of packets transmitted, rather than the number of receivers.

<heading lvl="0">Abstract of Disclosure< / heading> An embodiment of the invention includes an efficient error-control system and method for recovering packet losses, especially losses in distributing multicast video over broadband residential networks. Preferably, unlike most existing error-control algorithms designed for Internet multicast, the system and method does not employ substantial feedback suppression. Preferably, the system and method does not employ substantial multicasted retransmission. Preferably, the system and method does not employ substantial parity retransmission. Preferably, the system and method does not employ substantial local loss recovery. The system and method integrates two existing classes of error-control algorithms: Automatic Repeat Request (ARQ) and Forward Error Correction (FEC), to reduce traffic overhead and achieve scalability.

Provided are a sensor node device and method for supporting mobility of a mobile node in a sensor network. The sensor node device includes: a network interface for communicating with a sink node, at least one neighboring sensor node, and the mobile node; a mobility supervisor for detecting a location of the mobile node, and deciding a sensor node for a service of a data packet received through the network interface on the basis of the detected location; a transmission decision maker for deciding a transmission mode for the data packet received through the network interface according to the location of the mobile node; and a state reporter for reporting a transmission state of the data packet through the sensor node device. When the mobile node moves to a place where radio wave propagation areas of sensor nodes overlap each other, the radio wave intensity of each sensor node is detected to perform a handoff to the sensor node of the area having the strongest radio wave intensity. Thus, a ping-pong problem caused by unnecessary repeated handoff, and a delay problem caused by too slow handoff, can be prevented.

A method of adaptively routing voice packets in an IP network receiving a request for a new call at a voice gateway, determining a destination gateway for the new call, determining an availability of at least one route for routing the new call across the IP network to the destination gateway, and deciding whether to admit the new call into the IP network based on whether a route is available over which voice packets for the new call can be routed. A call admission decision is made based on whether a direct, single-path route or a two-path route is available, where each path is an LSP tunnel whose availability is determined using a token bucket technique. If no route is available, the voice gateway does not admit the call into the IP network.

Switching process at the occurrence of a path trouble is performed more quickly to reduce the number of packet discards during a traffic transition from a currently used system path to a standby system path within a section to be protected. An OLT (210-W) refers to the DBA information of a PON section and, if receiving no CCM frame at a timing at which the same should be received, then determines that some trouble has occurred on the path (S801) and transmits, to an OAM-compliant NE (200-Z), an application-for-switching frame (1501) to notify the OAM-compliant NE (200-Z) of the abnormal condition. The OAM-compliant NE (200-Z) monitors the occurrence of the trouble within the PON section nearly in real time, starts a switching process (S802) and generates and transmits a standby-system delivery request (321) (S302, S803). An OAM-compliant NE (200-A) switches the communication path to the standby-system passing through an OLT (210-P).

Disclosed is a method for high speed rerouting in a multi protocol label switching (MPLS) network which can minimize a packet loss and enable a fast rerouting of traffic so as to protect and recover a multi point to point LSP occupying most LSPs in the MPLS network. The method for high speed rerouting in a multi protocol label switching (MPLS) network, the method comprising the steps of controlling a traffic stream to flow in a reverse direction in a point where a node or link failure occurs by using a backup Label Switched Path (LSP) comprising an Explicitly Routed (ER) LSP having a reverse tree of a protected multi point to point LSP and an ingress LSR through an egress LSR.

A method and apparatus for rerouting traffic in a computer network select a routing element, apply a subnet mask to an interface address of the routing element to generate subnet addresses, determine the address of neighboring elements from the subnet addresses, and obtain configuration information from the neighboring elements. Network topology is determined from the configuration information, a link in the network topology is selected, and a cost associated with the link is modified. The modified cost renders the link less desirable for routing. A method and apparatus for determining the address of neighboring routing elements in a computer network, combine a subnet mask and interface address associated in a bitwise AND operation to generate subnet addresses, disregard the highest and lowest subnet addresses and the address of the selected routing element from the subnet addresses, and identify the remaining subnet address as associated with the neighboring routing element.

In accordance with a movement detection method, a mobile terminal in moving among access router devices all set at the same frequency, stores the link layer address of the currently reachable access router device. When receiving a Layer 2 frame from a new access router device, the mobile terminal issues a trigger for requesting to implement a part of IP handover processing. By so doing, it allows IP handover processing to be initiated at higher speed than in the case of the conventional IP layer, thus enabling to reduce the total time of handover processing.

The present invention relates to a receiver system in a communication system supporting packet-based communicatior (e.g., an IP-network), including a receiver, speech decoder (40) and a jitter buffer (20) for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller (50) is provided for keeping information about the functional size of the jitter buffer (20) and for providing the speech decoder (40) with control information, such that the speech decoder (40), based on that information. provides a dynamic adaptation of the size of the jitter buffer (20) using the received encoded, packetized speech signal. The invention also relates to a method of adapting the functional size of the jitter buffer of a receiver system.

A method of streaming an audio signal from an audio transmission device to first and second audio receiver devices via a Bluetooth link using a protocol that requires an audio data packet to be retransmitted if a positive packet receipt is not received. The first audio receiver device is synchronized to the audio transmission device to enable the first audio receiver device to receive an audio signal stream from the audio transmission device, the second audio receiver device is synchronized to the audio transmission device to enable the second audio receiver device to eavesdrop the audio signal stream to the first audio receiver device. When a positive packet receipt acknowledgement is not transmitted from the first audio receiver device to the audio transmission device, the audio transmission device repeats transmission of that audio data packet irrespective of whether the packet has been correctly received by the first audio receiver device.

Described herein is a method and system for hierarchical wireless video with network coding which limits encryption operations to a critical set of network coding coefficients in combination with multi-resolution video coding. Such a method and system achieves hierarchical fidelity levels, robustness against wireless packet loss and efficient security by exploiting the algebraic structure of network coding.

System and method for selecting a transmission medium on which to transmit a first stream. At least one of the plurality of transmission media may be substantially dynamic in nature. Path characteristics of each of a plurality of transmission media may be determined. A first transmission medium may be selected from the plurality of transmission media for the first stream based on the determined path characteristics. A first plurality of packets of the first stream may be transmitted on the first transmission medium.

Communication time period measuring frames are simultaneously sent from an OAM adaptive device on a transmission side, to both a working path and a protection path. In an OAM adaptive device on a reception side, reception times of the frames having arrived from both the paths are checked so as to measure a time period difference between both the paths. The time period difference is fed back to a logic distance adjustment function of each PON section to determine required communication time periods of the PON sections respectively included in a working system and a protection system. Communication time periods of the working system and the protection system in a packet communication network are arbitrated in order to decrease a packet loss at line switching in a packet relaying network.

Various techniques are disclosed for improving packet loss concealment to reduce artifacts by using audio character measures of the audio signal. These techniques include attenuation to a noise fill instead of attenuation to silence, varying how long to wait before attenuating the extrapolation, varying the rate of attenuation of the extrapolation, attenuating periodic extrapolation at a different rate than non-periodic extrapolation, and performing period extrapolation on successively longer fill data based on the audio character measures, adjusting weighting between periodic and non-periodic extrapolation based on the audio character measures, and adjusting weighting between periodic extrapolation and non-periodic extrapolation non-linearly.

The present invention relates to a receiver system in a communication system supporting packet-based communication (e.g., an IP-network), including a receiver, speech decoder and a jitter buffer for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller is provided for keeping information about the functional size of the jitter buffer and for providing the speech decoder with control information, such that the speech decoder, based on that information, provides a dynamic adaptation of the size of the jitter buffer using the received encoded, packetized speech signal. The invention also relates to a method of adapting the function size of the jitter buffer of a receiver system.

Generally speaking, there are provided systematic techniques for increasing and decreasing jitter buffer delay. The disclosed techniques typically utilize various combinations of: evaluating received data over a specified interval, increasing a recommended buffer delay if the interval delay exceeds a first threshold and decreasing the recommended buffer delay if the interval delay is less than a second threshold, causing the recommended buffer delay to decrease over time until an underflow condition is identified, and / or increasing the recommended buffer delay in response to identifying the underflow condition.

A vertical handover decision to perform a seamless vertical handover of a wireless apparatus from a current access network to a target access network is established. When deciding to perform the vertical handover from the current access network to the target access network it can be considered whether an intermediate access network could be used. It can be decided that first a seamless vertical handover is performed from the current access network to an intermediate access network, and subsequently a seamless vertical handover from the intermediate access network to the target access network is performed.

A system for transmitting data packets representing a source signal across a packet data network is provided. The encoder comprises a first encoder (110) and a redundancy encoder (120). The redundancy encoding is generated with a bit rate continuously scalable, the bit rate being provided by a bit rate controller (142) that uses input from the network (130) and packet-loss rate information. At the decoder, recovery is performed by a parameter estimator based in part on information transmitted from the first encoder using information from previous and / or future blocks and in addition on redundant information. The method may be added to existing lossy source coding systems or may be used to enhance the quality of the reconstructed source signal even in scenarios without packet loss.

A method for encoding video signals subjects the signals to unbalanced multiple description coding. The unbalanced multiple description coding codes a video signal in a first high resolution packet and a second low resolution packet and represents, respectively a first high resolution description and a second low resolution description. The unbalanced multiple description coding step includes using different intra refresh periods for the first and second high resolution descriptions, with an intra refresh period for the second low resolution description shorter than the intra refresh period of the first high resolution description.

Described herein is a method and system for hierarchical wireless video with network coding which limits encryption operations to a critical set of network coding coefficients in combination with multi-resolution video coding. Such a method and system achieves hierarchical fidelity levels, robustness against wireless packet loss and efficient security by exploiting the algebraic structure of network coding.

System and method for switching a stream to a new transmission medium. A first stream may be received. A first plurality of packets of the first stream may be transmitted to a second device on a first transmission medium. The first plurality of packets may include one or more index marker packets. It may be determined that at least a portion of the first plurality of packets may not have been received by the second device. A second plurality of packets of the first stream may be transmitted to the second device on a second transmission medium. The second plurality of packets may include at least a subset of the first plurality of packets and at least a subset of the one or more index marker packets. The one or more index marker packets may be configured for use by the second device to detect and discard duplicate packets.

An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.