Abstract: A TCP connection controller (10) for controlling 3-way handshakes establishing multiple TCP connections between an initiating network node and another network node is described. It includes a TCP pressure limiter (12) configured to limit TCP pressure during network congestion by reducing the rate of a TCP message type (SYN, SYN-ACK, ACK) participating in the 3-way handshakes.

Abstract: A method of controlling a radio access network entity of a data unit communication network is described, said radio access network entity being involved in providing a transport service to a radio terminal of said data unit communication network, said transport service being characterized by a set of values associated with respective service quality parameters, said method comprising: monitoring (S30) a communication quality at the radio access level of a communication between said terminal and an access point of said radio access network, and setting (S31) a congestion indication marking in data units of said transport service if said communication quality fulfills a radio communication degradation condition.

Abstract: A method is proposed for determining the round-trip delay for sending multimedia data, such as voice or video data, between a first network node and a second network node over a communications network, where data is encapsulated in data units comprising at least one frame of encoded data. The method includes the steps of: said first network node sending a request for reconfiguration of the data unit structure to the second node; the first node detecting a reconfiguration response from the second node, where the reconfiguration response comprises at least one data unit of data having a structure that corresponds to the request for reconfiguration. The first node then measures the time elapsed between sending the reconfiguration request and detecting the reconfiguration response and determines a value for roundtrip delay using said measured time. In accordance with a first embodiment, the request for reconfiguration is a request to change the encoding mode of the multimedia data.

Abstract: A wireless communications network for communicating with wireless communications devices, the wireless communications network comprising a processor arranged to: measure congestion for each wireless communication link between a plurality of the wireless communications devices and the wireless communications network; and prioritize scheduling of communications with the plurality of wireless communications devices dependent upon the measured congestion.

Abstract: A fluid powered percussion tool 1 has a housing 2 with a supply channel for pressurized fluid 3, a percussion mechanism 4, a swing joint 5 arranged to carry the percussion mechanism 4 relative to the housing 2 at a point situated between the forward end A and the rear end B of the percussion mechanism 4, at least one elastically resilient element 6 arranged between the housing 2 and the percussion mechanism 4 at a distance from the swinging joint 5 and arranged to load the percussion mechanism 4 against a neutral position in the housing 2 and to absorb the vibrational movements of the percussion mechanism 4, and a flexible connection for pressurized fluid 7 for distribution of pressurized fluid from the supply channel for pressurized fluid 3 to the percussion mechanism 4. The elastically resilient element 6 includes a conical spring 8. The flexible connection for pressurized fluid 7 includes a hose 11.

Abstract: A network node arranged to monitor (531) the number of consecutive blocked scheduling attempts made by a terminal (532), and to identify a congestion event for the terminal (534) when the number of consecutive blocked scheduling attempts exceeds a threshold value (533).

Abstract: The present invention relates to speech coding in wireless and wireline communication systems. The present invention provides a method of saving bandwidth by a controlled dropping of speech frames at an encoder in a sending communication device. The dropping is controlled in a manner to minimize the effects on the speech quality after the decoding in the receiving communication device, by assuring that the state mismatch between the encoder and the decoder is removed or at least significantly reduced. This is achieved by letting the encoder run an ECU algorithm with a similar behavior as the one running in the decoder in the receiving communication device.

Abstract: The present invention is to select an adaptation scheme for the transmission of the encoded media that results in a satisfactory performance of the transmitted encoded media. A difference from the prior art is that each adaptation scheme defines a set of different transmission formats, wherein each transmission formats is a combination of at least two of the parameters the source codec bit rate, the packet rate, the number of frames of each packet (referred to as frame aggregation), and the level of redundancy. By using the different transmission formats, the transmission can be adapted to different operating scenarios and the performance is hence improved.

Abstract: In a method of improved media frame transmission in a communication network. Initially a plurality of “original” or regular media frames are provided for transmission. According to the invention, robust representations of the provided regular media frames are generated and stored locally. Subsequently, one or more of the regular media frames is/are transmitted. The invention detects an indication of a loss of a transmitted media frame, and the idea is to transmit, in response to a detected frame loss, a stored robust representation of the lost media frame and/or a stored robust representation of a subsequent, not yet transmitted, media frame to increase the media quality.

Abstract: A base station detects the transition of a mobile station from a DTX mode to a continuous transmission mode. The base station schedules an uplink transmission by the mobile station during an uplink transmission period while the mobile station is in DTX mode and receives an uplink LLC PDU from said mobile station during the scheduled uplink transmission periods. The base station determines the operating mode of the mobile station based on the size of the received uplink LLC PDU.

Abstract: A network processing node (e.g., MGW, MRFP) and method are described herein that can: (1) receive packets on a first heterogeneous link (e.g., wireless link); (2) manipulate the received packets based on known characteristics about a second heterogeneous link (e.g., “Internet” link); and (3) send the manipulated packets on the second heterogeneous link (e.g., “Internet” link). For example, the network processing node can manipulate the received packets by adding redundancy, removing redundancy, frame aggregating (re-packetizing), recovering lost packets and/or re-transmitting packets.

Abstract: A method is proposed for determining the round-trip delay for sending multimedia data, such as voice or video data, between a first network node and a second network node over a communications network, where data is encapsulated in data units comprising at least one frame of encoded data. The method includes the steps of: said first network node sending a request for reconfiguration of the data unit structure to the second node; the first node detecting a reconfiguration response from the second node, where the reconfiguration response comprises at least one data unit of data having a structure that corresponds to the request for reconfiguration. The first node then measures the time elapsed between sending the reconfiguration request and detecting the reconfiguration response and determines a value for roundtrip delay using said measured time. In accordance with a first embodiment, the request for reconfiguration is a request to change the encoding mode of the multimedia data.

Abstract: A method for localized congestion exposure within a local loop in a cellular network that is performed by a localized congestion exposure receiver node of the local loop. The method includes receiving downlink packets destined for a downstream user device. The downlink packets have headers that indicate a level of congestion experienced by the downlink packets. The headers also indicate a level of expected downstream congestion declared by an upstream node. The method also includes forwarding the downlink packets to the downstream user device through a wireless connection. The method further includes sending packets upstream that have feedback indicative of the level of congestion experienced by the downlink packets and any congestion experienced within the localized congestion exposure receiver node.

Abstract: A network processing node (e.g., MGW, MRFP) and method are described herein that can: (1) receive packets on a first heterogeneous link (e.g., wireless link); (2) manipulate the received packets based on known characteristics about a second heterogeneous link (e.g., “Internet” link); and (3) send the manipulated packets on the second heterogeneous link (e.g., “Internet” link). For example, the network processing node can manipulate the received packets by adding redundancy, removing redundancy, frame aggregating (re-packetizing), recovering lost packets and/or re-transmitting packets.

Abstract: In an improved method in a control node managing media sessions between nodes in a telecommunication network, exchanging S10 at least one media session description message between two nodes, and determining S20 if the one message comprises at least one option tag, said at least one option tag indicating potential configurations for said media session. Subsequently, comparing S30 the at least one option tag to a set of network supported option tags, said network supported option tags indicating supported configurations for said network, and modifying S40 the media session initiation message based on the comparison.

Abstract: A method of controlling a radio access network entity of a data unit communication network is described, said radio access network entity being involved in providing a transport service to a radio terminal of said data unit communication network, said transport service being characterized by a set of values associated with respective service quality parameters, said method comprising: monitoring (S30) a communication quality at the radio access level of a communication between said terminal and an access point of said radio access network, and setting (S31) a congestion indication marking in data units of said transport service if said communication quality fulfils a radio communication degradation condition.

Abstract: A method and a communication station for determining a packet format of a data packet based on at least one compressed header information field. A header packet format of a data packet is determined based on the determined packet format of a partially or completely compressed header information part. The station may also determine a codec rate based on the result of a previously performed compression of at least one header information field. In one embodiment, internal cross-layer signaling from a header generator module to a payload generator module in the communication station is used for signaling information associated with determining the header packet format of the data packet. The method reduces bandwidth fluctuations associated with services such as text messaging audio, or audiovisual services; lowers the delay variations and the erasure rates for application media streams; and provides for faster set-up of sessions.

Abstract: A control logic means preferably for a receiver comprising a jitter buffer means adapted to receive and buffer incoming frames or packets and to extract data frames from the received packets, a decoder connected to the jitter buffer means adapted to decode the extracted data frames, and a time scaling means connected to the decoder adapted to play out decoded speech frames adaptively. The control logic means comprises knowledge of whether a state recovery function is available and is adapted to retrieve at least one parameter from at least one of the jitter buffer means, the time scaling means, and the decoder, to adaptively control at least one of an initial buffering time of said jitter buffer means, the knowledge of the availability of the state recovery function, and a time scaling amount of said time scaling means from the time scaling means or the decoder.

Abstract: This patent defines a device for automatic calibration of optical sights for firearms. Only one shot has to be fired to achieve this. The device can be integrated with an optical sight or fitted as a separate unit mounted on the sight. The device consists of a digital camera (e.g. CCD), a beam splitter, a microprocessor including a memory for camera images and computer software, servo motors including a gear mechanism, an electrical switch and light emitting diodes and a shot detecting sensor. In the digital sight application, the device also includes a display unit while servo motors, gear mechanism, light emitting diodes and beam splitter are excluded. The calibration procedure involves firing a round at a target consisting of a rectangular white surface on a dark background at the chosen range. The camera saves the last image immediately before the firing moment and compares the cross-hairs position with that of a following image from which the projectile point of impact can be found.

Abstract: Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal. A number of encoding schemes for the side signal are provided. Each encoding scheme is characterized by a set of sub-frames of different lengths. The total length of the sub-frames corresponds to the length of the encoding frame of the encoding scheme. The encoding scheme to be used on the side signal is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimize the side residual signal. The optimized side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.