Abstract: A base station (103) assigns a set of mobile stations (101) to a group wherein the group will share a set of radio resources (770). A control field (1103) may be sent with a payload field (1105) wherein the control field (1103) and payload field (1105) are sent using a single Orthogonal Variable Spreading Factor or a single Walsh Code (1101) wherein various modulation and coding schemes may be applied to the control field (1103) and payload field (1105) such that different modulation and coding schemes may be used within the single channel. HARQ is handled by sending a single retransmission if a NACK message is received or no ACK/NACK message is received at all.

Abstract: A plurality of wireless addresses to be transmitted using a shared message and at least one item of informational content are provided (101 and 102). At least some of the plurality of wireless address are then arranged (103) into an order that represents the at least one item of informational content. Upon transmitting (104) a shared message that contains this content, a receiving platform can, in turn, decode (302) (and hence recover) the at least one item of informational content as a function, at least in part, of that aforementioned particular order.

Abstract: An access network unit of a third-generation mobile communication system controls a wireless LAN system by regarding the wireless LAN system as a packet communication channel. The access network unit includes a radio network controller connected to a base transceiver station that is capable of performing a wireless communication with a mobile terminal in the wireless LAN system, and electrically connected to access point stations of the wireless LAN system. Therefore, setup and control of a wireless LAN channel lead by the third-generation mobile communication system can be performed between both systems.

Abstract: A mapping table holds correspondence relation between quality of service information of wired sections and quality of service information of wireless sections. A quality of service information converter, upon receiving encapsulated packets directed from a mobile communication network to a wireless local area network terminal, acquires quality of service information of wired sections that has been conferred to the packets, and finds quality of service information of wireless sections that corresponds to the quality of service information of wired sections in the mapping table. The quality of service information converter then uses this quality of service information of wireless sections to wirelessly transmit the encapsulated packets to a wireless local area network terminal.

Abstract: A geographic-based communications service system has a mobile unit for transmitting/receiving information, and access points connected to a network. The access points are arranged in a known geographic locations and transmit and receive information from the mobile unit. When one of the access points detects the presence of the mobile unit, it sends a signal to the network indicating the location of the mobile unit and the information requested by the mobile unit. Based on the signal received from the access point, the network communicates with information providers connected to the network and provides data to the mobile unit through the access point corresponding to the location of the mobile unit.

Abstract: A method and a radio communication system transmit user information as a service to several user stations in a radio communication system comprising at least one base station and user stations which are connected to each other via a radio communication interface. The base station signals to the user stations using the at least one service via a group control channel, radio resource information on at least one traffic channel via which the user information of the at least one service is transmitted to the user stations.

Abstract: The present invention is related to a backward outer-loop power control method in a wideband CDMA mobile communication system for improving backward call quality and capacity of an IMT-2000 asynchronous system. In accordance with the present invention, the power control is optimized by examining the field transmitted through the backward Dedicated Physical Control Channel (DPCCH), as well as the CRC examination of the uplink data frame received, thus setting the optimum target SIR.

Abstract: In an H-ARQ system, when the AN is receiving packet data traffic on the RL from an AT and is generating ACKs and NAKs according to the ability of the AN to properly decode such data, the AN gates-off a DRCLock bit within in a sub-packet duration in which an ACK is transmitted on the FL MAC channel. When it receives an ACK, the AT ignores the non-transmitted DRCLock bit in a current sub-packet duration, and assumes that the DRC channel is “good”. When the AN sends a NAK to the AT, it also sends the DRCLock bit. When the AT receives a NAK in a sub-packet duration, it reads and processes whatever DRCLock bit is received during that sub-packet duration. When no data traffic is transmitted on the RL traffic channel, corresponding DRCLock bits are not gated-off by the AN and are transmitted to the AT. The AT then processes the received DRCLock bits. In an alternative embodiment, transmission of DRCLock is totally eliminated.

Abstract: In some embodiments of the invention, an unassociated node in a low-duty-cycle wireless sensor network may determine when the next network operational period begins, so that the node may go into a power-saving sleep mode until the start of that operational period.

Abstract: A system and method for calculating a bandwidth requirement between two elements in a communications network are described. The method determines data stream information comprising a total number of data streams available for transmission within at least a portion of a communications network. Endpoint information is determined regarding a number of endpoints in the communications network associated with transmission stream information. Distribution information is determined wherein the distribution information is associated with transmission of the data streams to their end points. The method calculates the bandwidth between two elements in the communications network associated with a data stream demand using the data stream information, the endpoint information, and the distribution information.

Abstract: The disclosure is directed to an access terminal, and method for selecting an access point in a communications system. The access terminal may include memory and a processor. The processor may be configured to maintain in the memory a database containing a list of access points that have previously served the access terminal with at least a minimum quality of service, the processor being further configured to use the list to search for an access point to associate with, or search for an access point to handoff the access terminal to during operation. In addition to, or alternatively, the processor may also be configured to maintain in the database a second list of access points. The database may be used by the processor to search for an access point that is not on the second list to associate with, or handoff the access terminal to.

Abstract: Various embodiments are disclosed relating to a multicast-only data transmission mode in a wireless network. According to an example embodiment, an apparatus for wireless communications may include a controller. The apparatus, which may include an AP or a virtual AP may be configured to operate in a multicast-only data transmission mode (e.g., transmitting only multicast and/or broadcast data frames). The AP or virtual AP may operate in a multicast-only data transmission mode to handle multicast transmissions for one or more multicast streams for one or more wireless networks, for example.

Abstract: This specification describes technologies relating to a phone-based system for identifying broadcast audio streams, and methods of providing such a system. In one aspect, a method includes receiving a plurality of broadcast streams, each from a corresponding broadcast source and generating a first broadcast audio identifier based on a first broadcast stream of the plurality of broadcast streams. The method also includes storing for a selected temporary period of time the first broadcast audio identifier. The method further includes receiving a user-initiated telephone connection; and generating a user audio identifier. Other implementations of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A method for implementing the media gateway function is disclosed by the embodiment of the present invention, including: receiving a first signalling message; determining to perform a processing of a base station application layer or a processing of a media gateway control application layer according to a Simple Control Transmission Protocol (SCTP) link, wherein the SCTP link is indicated by the first signalling message; if determine to perform the processing of the media gateway control application layer, routing traffic data to a destination device. A radio access control device and a radio access system according to the above mentioned method are further disclosed. According to the embodiment of the present invention, the exchange number of an IP packet is reduced, and the bearer is separated from the control.

Abstract: In a communication system wherein a CDMA segment at each access point consists of multiple sub-segments a three frame transmission time interval (TTI) with eight retransmissions is utilized for data transmission. The access point not only specifies the interlaces to be utilized for data transmission it also assigns packet start interlaces for particular access terminals. An auxiliary pilot channel R-AuxPICH is transmitted by an access terminal along with CDMA data on reverse link. The ratio of R-AuxPICH to R-PICH is varied based on ACK/NACK feedback. A reverse link activity bit (RAB) which can be used as an emergency load-control mechanism for non-QoS flows is also disclosed.

Abstract: A wireless communication technology, and more particularly, a transmission packet for wireless transmission in a high frequency band, and a method and apparatus for transmitting and receiving using the same, are described. The structure of a transmission packet may include an MPDU composed of a plurality of transmission data units, a MAC header unit added to the MPDU, and a PHY header unit added to the MAC header unit, wherein the MAC header unit includes a MAC header generated based on the information used in a MAC layer, a first HCS field which determines if an error occurred in the MAC header or the PHY header, a MAC header extension field which exists depending on the setting of an indicator field in the MAC header, and a second HCS field which determines if an error occurred in the MAC header extension field.

Abstract: The present disclosure relates generally to systems and methods for random access in a wireless communication system. In one example, the method includes allocating first and second random access orthogonal frequency subbands to first and second subdivisions, respectively, in a wireless communication network. The first subband is assigned to a first position in a first subframe for use in the first subdivision during a first transmission slot, and the second subband is assigned to a second position in the first subframe for use in the second subdivision during the first transmission slot. The first subband is then assigned to a third position in a second subframe for use in the first subdivision during a second transmission slot following the first transmission slot, and the second subband is assigned to a fourth position in the second subframe for use in the second subdivision during the second transmission slot.

Abstract: A multiple-input multiple-output (MIMO) scheme uses precoding and feedback in a wireless communication system including a transmitter and a receiver. The system may use either a single codeword (SCW) or a double codeword (DCW). The precoding scheme is based on transmit beamforming (TxBF). Combined differential and non-differential feedback with periodic resetting is considered.

Abstract: When a wireless transmit receive unit (WTRU) is using a reduced dedicated physical control channel (DPCCH) frame, the WTRU may periodically send a normal DPCCH frame. The WTRU may periodically increase the transmit power or send a normal DPCCH frame if there is a transmit power control (TPC) error or a downlink quality is below a threshold. When the WTRU implements DPCCH transmission gating, the WTRU may set a gating period based on the number of received TPC commands. The WTRU or Node-B may restore to a normal mode, if the link quality is below a threshold. The WTRU may periodically increase a transmit power. When a reduced signal-to-interference ratio (SIR) target is used for TPC, the WTRU may increase a transmit power, if the downlink power is not responsive. The WTRU or the Node-B may restore a normal SIR target if the link quality is below a threshold.

Abstract: A system and method for effectively and reliably terminating a voice call in any burst within a multi-burst superframe. A transmitting unit generates a termination burst upon detecting a dekey event. The termination burst includes a data synchronization pattern, a slot type field indicating an end of a call, and an information field surrounding the data synchronization pattern and the slot type field. The information field is encoded from a predetermined voice encoder frame unique to the termination burst. Once all buffered voice information is transmitted, the termination burst is transmitted prior to the end of the multi-burst superframe. A base station or other receiving unit monitors the incoming signal. Upon detecting the data synchronization pattern, the receiving unit decodes the slot type field and the information field.

Abstract: A mobile station measures channel variation in a time direction and channel variation in a frequency direction, and selects a CQI compression method to be actually applied, on the basis of a CQI updating cycle, a CQI channel size, the channel variation in the time direction and the channel variation in the frequency direction transmitted from a base station. The mobile station notifies the base station of CQI format information associated with the selected CQI compression method together with a measured CQI value. On the basis of the CQI format information, the base station analyzes in what compression method the CQI value is compressed, and executes scheduling in accordance with a result of the analysis.

Abstract: A Common Wireless Communications Protocol (CWCP) below the TCP/IP layer is a part of OSI layer 3a (SNDCF-Subnetwork Dependent Convergence Functions). CWCP serves as a common Application Program Interface (API) to the upper application layers. CWCP supports different call processing standards (GSM, WCDMA, CDMA2000) and provides seamless Web or e-mail interfaces using Wireless Application Protocol (WAP) or Transport Layer Interface (TLI) by the CWCP interfacing with the different physical layer (3G) standards. CWCP provides interoperability between the different physical standards and provides a plug-and-play environment for different wireless devices and applications. CWCP provides interoperability at the systems level via interoperable software through the IP software layer.

Abstract: The present invention discloses a technique for a user to control a manner in which a data store of a mobile device synchronizes with a remotely located data store. More specifically, synchronization actions can be automatically executed by comparing combinations of user modifiable device settings against device-specific conditions. Device conditions can include device resources, timing parameters, and user-defined criteria, such as data priority, device location, and application-specific conditions. In one embodiment, the invention provides a means to override an application's existing synchronization rules. This permits synchronization for a device that includes multiple applications to be centrally and consistently handled, even when the applications normally lack synchronization control capabilities.

Abstract: The present invention provides a unique manner of identifying the Frame boundaries in multiple identical/non-identical Synchronization Channels (SCHs) in different sub-frames via a new sub frame position difference method. The method implements the differences between the different sub-frames carried by the SCH. The sub frame identity, and hence the frame boundary, can be identified by calculating the difference between the positions, based on time or number of slots/sub-frames or any data packets, of the two subsequent slots/sub-frames or any identical data packets.

Abstract: A method, circuit, and system are disclosed. In one embodiment, the method comprises dynamically adjusting the output voltage of the output drivers of one side of a bi-directional serial link down to the lowest voltage level that is able to maintain compliance with the error rate allowance threshold of the serial link, and operating the one side of the serial link at that voltage level.

Abstract: The present invention advantageously provides a method, system and apparatus for aggregating multiple site-specific routes, by determining a first aggregate-aware route that includes a prefix of a site-specific address and a prefix length of an aggregate route address of a first service provider. The method and system can be implemented as an enhancement to existing IP protocols such as BGP and other inter-domain routing protocols. The method and apparatus may further include applying a routing protocol policy in which an aggregate route may serve as a proxy for an aggregate-aware route when the address of the aggregate route matches the address of the aggregate-aware route. The method and system may yet further include determining a second aggregate-aware route that includes the prefix of the aggregate route address of the first service provider and a prefix length of a second aggregate route address of a second service provider.

Abstract: A method for transmitting data from a source to a destination. The method includes: transmitting the data from the source to both the destination and a relay station within a first transmission; decoding the data in the first transmitting at both the relay station and the destination wherein the destination acknowledges success of the decoding by generating a ACK message or lack of success by generating either a NACK message or by failing to generate the ACK signal; retransmitting from the source information related to the first transmission in response to either the NACK message or the failure of the destination to generate the ACK message; and wherein when the relay station decodes the data sent by the source within the first transmission correctly, the relay station transmits a second transmission; or, on the other hand, if the relay station is unable to decode the data correctly from the first transmission sent by the source, the relay station remains silent. The destination may be another relay station.

Abstract: A routing multiplexer system provides p outputs based on a selected permutation of p inputs. Each of a plurality of modules has two inputs, two outputs and a control input and is arranged to supply signals at the two inputs to the two outputs in a direct or transposed order based on a value of a bit at the control input. A first p/2 group of the modules are coupled to the n inputs and a second p/2 group of the modules provide the n outputs. A plurality of control bit tables each contains a plurality of bits in an arrangement based on a respective permutation. The memory is responsive to a selected permutation to supply bits to the respective modules based on respective bit values of a respective control bit table, thereby establishing a selected and programmable permutation of the inputs to the outputs.

Abstract: A system and method for delivering data via an internet protocol television (IPTV) network are disclosed. In a method disclosed, a first data packet of a first data stream is received. The method determines whether a corresponding second data packet of a second data stream has been forwarded. The corresponding second data packet includes a second payload substantially the same as a first payload of the first data packet. When the corresponding second data packet has not been forwarded, the method forwards the first data packet.

Abstract: A 3-in-1 automatic call forward system integrates the domestic phone, the mobile phone and the internet phone for switching the communication method automatically and choosing an appropriate method to transmit or receive a phone call. The users also can choose the same telecom service provider to perform communication with each other for reducing the telephone fees.

Abstract: An indirect communication system and a method of indirect communication include a mobile phone as either a calling device or receiving device. The mobile phone calls another device to set up a prospective communication, and then uses Voice over Internet Protocol (VoIP) to communicate with the other device over the Internet. The receiving device receives a generated signal notifying the receiving device of a proposed communication with the calling device. A server sets up a meeting point channel after the calling device has connected to the server. The server receives outgoing VoIP packets from the calling device and redirects the outgoing VoIP packets to the receiving device.

Abstract: A switch having a first communications interface to receive a stream of data units, a buffer to store the payload portion of the data unit, a replicator to duplicate at least the payload portion of the data unit, and a second communications interface to enable access to the stream of data units by two or more terminals.

Abstract: The object is to realize a multi-line service with proprietary messages/procedures only between telephone apparatuses such as IP telephones, and to use a connection management server complying with a standard connection protocol without using proprietary messages/procedures between the connection management server such as the SIP server and the telephone apparatus. A telephone apparatus for performing a telephone call using a predetermined communication protocol and a connection protocol, includes: a unit configured to mutually report a line status using a proprietary interface different from the connection protocol between the telephone apparatus and another telephone apparatus forming a multi-line group.

Abstract: The described method, system and terminals may allow the sending capability information from a first terminal to a second terminal in a capability exchange performed without, or prior to, conducting a call between the first and second terminals. The capability information may include at least one of multimedia call related parameters such as a codec for example, or network capability related information. A session initiation protocol may be used for sending the multimedia call related parameters. A service change may thus rapidly be performed.

Abstract: A voice communication system, which is connected to a LAN to which communication terminals are connected and to a public network to which telephones are connected, is provided with a communication server between the LAN and public network having different protocols from each other. The communication server enables a voice communication between a telephone on the public network and a communication terminal connected to the LAN by performing processing similar to that for a voice communication between two communication terminals connected to the LAN. The communication server determines whether an address of the other party inputted by a user is a communication terminal address or a telephone number, and transmits a voice communication request to a communication terminal of the other party when the address is a communication terminal address.

Abstract: A voice communication system, which is connected to a LAN to which communication terminals are connected and to a public network to which telephones are connected, is provided with a communication server between the LAN and public network having different protocols from each other. The communication server enables a voice communication between a telephone on the public network and a communication terminal connected to the LAN by performing processing similar to that for a voice communication between two communication terminals connected to the LAN. The communication server determines whether an address of the other party inputted by a user is a communication terminal address or a telephone number, and transmits a voice communication request to a communication terminal of the other party when the address is a communication terminal address.

Abstract: A voice communication system, which is connected to a LAN to which communication terminals are connected and to a public network to which telephones are connected, is provided with a communication server between the LAN and public network having different protocols from each other. The communication server enables a voice communication between a telephone on the public network and a communication terminal connected to the LAN by performing processing similar to that for a voice communication between two communication terminals connected to the LAN. The communication server determines whether an address of the other party inputted by a user is a communication terminal address or a telephone number, and transmits a voice communication request to a communication terminal of the other party when the address is a communication terminal address.

Abstract: A voice communication system, which is connected to a LAN to which communication terminals are connected and to a public network to which telephones are connected, is provided with a communication server between the LAN and public network having different protocols from each other. The communication server enables a voice communication between a telephone on the public network and a communication terminal connected to the LAN by performing processing similar to that for a voice communication between two communication terminals connected to the LAN. The communication server determines whether an address of the other party inputted by a user is a communication terminal address or a telephone number, and transmits a voice communication request to a communication terminal of the other party when the address is a communication terminal address.

Abstract: A voice communication system, which is connected to a LAN to which communication terminals are connected and to a public network to which telephones are connected, is provided with a communication server between the LAN and public network having different protocols from each other. The communication server enables a voice communication between a telephone on the public network and a communication terminal connected to the LAN by performing processing similar to that for a voice communication between two communication terminals connected to the LAN. The communication server determines whether an address of the other party inputted by a user is a communication terminal address or a telephone number, and transmits a voice communication request to a communication terminal of the other party when the address is a communication terminal address.

Abstract: A computerized telephony bridge unit has a Data Network Telephony (DNT) Port and a Connection Oriented/Switched Telephony (COST) trunk port, each associated with circuitry for receiving a placing calls in the data format required by connected networks. The bridge unit further comprises conversion circuitry for converting data dynamically between network protocols compatible with each connected network. Control routines executable on the computerized bridge unit are adapted to receive a first call from one of the COST and DNT networks, to place a call associated with the received call on the network other than the network on which the call is received, and to dynamically convert data between a call connected at one port and a call connected at the other port. The data network can be the Internet, and the COST network can be any publicly or privately switched dedicated-connection-oriented telephone network.

Abstract: A method and system for providing intelligent network control services in IP telephony, wherein the system includes a location manager and an IP telephony proxy server. The location manager includes an interface to a legacy telephony service control entity and the IP telephony proxy server includes an IP interface to the location manager. If the IP telephony proxy server requires intelligent network services, then the IP telephony proxy server sends an IP telephony session initiation request to the called party at the location manager. The location manager uses the information to query the legacy telephony service control entity for routing information. When the location manager receives a routing response from the service control entity, the location manager maps the response to an IP telephony session control message back to the IP telephony proxy server.

Abstract: A communication system for an organization having multiple sites uses a dual-mode device capable of both cell phone communication and telephone communication on a local area network (LAN). IP LANS are established at organization sites such that a temporary IP address is assigned to a dual-mode device that logs onto an organization LAN, and the IP address is associated at a PSTN-connected server on the LAN with the cell phone number of the communication device. The IP server notifies a PSTN-connected routing server when a device logs on to a LAN, and also provides a destination number for the IP server. Cell calls directed to the device are then redirected to the IP server and directed to the device connected to the LAN.

Abstract: A system is provided for providing communication event routing and transfer capability in a multi-site communication-center environment. The system utilizes a presence protocol application and a routing application for determining availability of an agent or system and for setting up the transfer from a point of transfer on a network to a destination of transfer on the same or connected network. In a preferred use the presence protocol application is shared by communication-center sites cooperating in the transfer and routing of events and the presence protocol enables at least one event-handling process normally performed by the routing application.

Abstract: A software suite is disclosed for routing communication events over a data-packet-network using an IP session initiation and management protocol. The software suite comprises, a server application running on the network for computing and serving routing determinations per request, a session management application running on the network for initiating and managing routed and established session events, a parsing application running on the network for parsing request data received under session initiation protocol and a conversion application running on the network for converting data received under session initiation protocol into a routing request. All received communication requests for routing are in the form of the session initiation protocol wherein they are parsed and converted into routing requests processed by the server application and routed to determined destinations and wherein events are established as session events conducted under the session initiation and management protocol.

Abstract: User data is transmitted between an interconnected data transmission network and a data package transmission network. A receiver of an original signaling message generates a partner signaling message, which has been defined for a data package transmission and is suitable for transporting the original signaling message. The partner signaling message excludes a portion of the original signaling message.

Abstract: A packet-data network is made intelligent in the sense of a connection-oriented, switched telephony (COST) network by enhancing one or more interconnected IP routers in the network with computer-telephony integration (CTI) processors executing CTI applications. No-charge-to-calling-party IP addresses are assigned and sponsored by various enterprises, who may also maintain call centers having at least one CTI-enhanced IP router connected to the network, and agent stations having IP telephones connected to the call-center-located IP router. With appropriate software and the CTI link to IP routers the performance of well-known conventional telephone systems may be provided in packet networks like the Internet.

Abstract: Accordingly, the present invention relates to a monitoring system and method for a trunk gateway. In the monitoring system of the present invention, a trunk gateway is interposed between the Internet and a PSTN and is adapted to packetize a trunk line into packets and to transmit the packets to the Internet, or to perform a reverse procedure. The monitoring system includes a VoIP agent connected to the Internet and adapted to originate and terminate a VoIP call and to generate an MOS value using a voice file received from a counterpart. A circuit network agent is connected to the PSTN and adapted to originate and terminate a call and to generate a MOS value using a voice file received from a counterpart. A center server performs various types of control operations related to measurement of service quality of the trunk gateway, including establishment and termination of a call, and collects the MOS values.

Abstract: A data transmission controlling device for controlling a data transmission of an electronic device is provided. The data transmission controlling device includes a switch generating a code and a switching signal; a bias circuit generating a bias voltage for the electronic device; and a body controlling the bias voltage generated by the bias circuit in response to the code and the switching signal received from the switch.

Abstract: In one aspect, a digital device comprises a wireless station, a universal serial bus (USB) port, a memory, and a processor. The processor is configured to detect a coupling of a flash device to the USB port and obtain wireless network configuration information from the flash device. The wireless network configuration information comprises a network name and a security key for a wireless communications network. The processor is further configured to store the wireless network configuration information in the memory, configure the wireless station using the wireless network configuration information, and connect the wireless station to the wireless communications network based on the wireless network configuration information.

Abstract: Systems and methods for determining that a maximum number of IP sessions have been established are provided. There are instances when the mobile device transmits a request to the wireless network. According to an aspect, in the event that the maximum number of IP sessions is already established for the mobile device, the wireless network transmits a response indicating that the request cannot be fulfilled. The mobile device determines based on the response that the maximum number of IP sessions is already established, which allows the mobile device to determine the maximum number of IP sessions that can be supported so that IP sessions can be managed accordingly.