Abstract: In one embodiment, a method is provided. The method of this embodiment may include, in response, at least in part, to receipt at a first node of a request issued from a second node to change from one mode of operation to another mode of operation, selecting an amplitude of a signal to be propagated between the first node and the second node during the another mode of operation. The amplitude of the signal may be different from another amplitude of the signal during the one mode of operation. The method of this embodiment also may include, in response, at least in part, to the receipt of the request, selecting a frequency of a clock signal to be supplied in the first node during the another mode of operation. The frequency may be different from another frequency of the clock signal during the one mode of operation.

Abstract: A module and kit for coupling at least one telephone service signal to at least one telephone device over a wiring. The network includes: a wiring having at least two conductors for carrying multiple time-domain multiplexed digitized voice channels; an exchange side device coupled to the wiring and operative to couple at least one telephone service signal to at least one digitized voice channel; and at least one subscriber side device coupled to the wiring and operative to couple the at least one telephone device to at least one digitized voice channel.

Abstract: In a communication network, the network management system NMS keeps one or more element management system EMS in synchronization of network topology changes using requests that are automatically generated or manually requested and then sent from the NMS to the affected EMS(s) whenever a node topology change is made at the NMS. Examples of changes are node creation, node move, node group move, node group renaming, etc. As the changes are completed in the EMS topology database, the EMS sends acknowledgments of the request to the NMS. This functionality requires verification of the validity of the request for that specific EMS, before the change is implemented at the NMS.

Abstract: System management functions are remotely performed on customer premises equipment (CPE), such a residential gateway, even if the CPE has not been authenticated or dynamically allocated an IP address or otherwise is not in a state that enables normal communication with the network. CPE management information is communicated using an IP address identifying a destination CPE but exclusively associated with CPE management. The access node that receives the CPE management information reconfigures the packets to translate the address to a fixed, non-public address. The same fixed address can be shared by all of the CPE's connected to the access node. The CPE recognizes the fixed address as though it were being addressed locally rather than through the network and is therefore able to receive the CPE management information even if the CPE has not been authenticated or dynamically allocated an IP address.

Abstract: The invention features a telephone, intercom, and clock. The telephone comprises a receiver adapted to receive a signal representing encoded first voice data; a processor comprising a media access controller adapted to obtain the encoded first voice data from the signal, and a digital signal processor adapted to decode the first voice data obtained by the media access controller; an output circuit; a digital-to-analog converter adapted to convert the first voice data decoded by the digital signal processor to a first analog signal, wherein the first analog signal is output to the output circuit; an input circuit; an analog-to-digital converter adapted to receive a second analog signal from the input circuit, and to convert the second analog signal to second voice data; wherein the digital signal processor is further adapted to encode the second voice data; and a transmitter adapted to transmit signals representing the encoded second voice data.

Abstract: Video, voice and data are integrated using existing telephone connections by using a video network server to receive and transmit video signals to and from clients using telephone wires. Client frequency division multiplexing couplers, each associated with one of the clients, receive video and telephone signals from the client and frequency multiplex them onto the telephone wires for transmission. They also receive video signals from the server and frequency demultiplex them for transmission to the client. A video frequency division multiplexing coupler connected between the video network server and each of the client frequency division multiplexing couplers receive the video and telephone signals from the clients and transmit only the video signals to the video network server.

Abstract: Provided is a method for controlling a video server providing video and audio data to a plurality of clients via a communication network in real time. Analog audio signals and analog video signals are respectively converted into digital audio data and digital video data. The digital audio data and digital video data are compressed. The compressed digital audio data and digital video data are packetized to generate an audio packet and a video packet. The audio packet and video packet are converted into an audio/video RTP packet according to real-time transport protocol. The audio/video RTP packet are transmitted to the clients according to TCP/UDP-IP.

Abstract: A data carousel contains multiple data files having a particular arrangement. These files are cyclically broadcast to a number of receivers of the data files. A procedure modifies the arrangement of data files based on information obtained from receivers of the data files.

Abstract: Apparatus is used in communicating in a data and voice communication system. The apparatus includes a line card and a crosspoint switch in communication with the line card. The apparatus also includes an accelerator card in communication with the line card and the crosspoint switch. The crosspoint switch is configured to provide a redundant communications path between the line card and the accelerator card.

Abstract: A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. Such quality is ensured by inserting gaps between successive data packets in a stream of multiplexed data and/or voice packets. A gap after a particular data packet is proportional to the size of that particular data packet. In this way, bandwidth is preserved for any voice packets that may have arrived during the transfer of the data packet as well as for any voice packets that arrive during the gap. The unconstrained upstream data bandwidth and the bandwidth used by voice calls may each be estimated by taking a plurality of instantaneous measurements of the available bandwidth and/or taking individual direct measurements. The size of data packets may be limited to a maximum size in order to ensure that time-sensitive voice packets experience only an acceptable delay in queue for transmission.

Abstract: A network switch. The network switch includes a processor configured to execute a set of instructions to access a recorded announcement with attendant text and to transmit the recorded announcement with the attendant text to a telecommunications device.

Abstract: The VOIP drop amplifier connects end user equipment to a broadband system, such as that provided by a cable provider. The amplifier includes a splitter for connecting the cable signal to multiple output connectors, and RF amplifiers compensating for losses in the splitter and other passive components in the amplifier. The drop amplifier includes an input connection for accepting a broadband cable signal from a cable system and returning broadband signals to the cable system. The drop amplifier includes an amplification path connecting the input connection to the plurality of output connections through RF amplifiers and a splitter, and a bypass path that bypass the splitter and forward and reverse amplifiers in the amplification path to connect the input connection directly to the output connection for the essential circuits. A sensing circuit monitors the amplifiers and the supply voltages, and selects the bypass path when a failure is detected.

Abstract: A system for delivering broadband service, including standard telephone service and telephone/DSL service, to a subscriber. The system includes a pair gain system and a service distribution system. In some embodiments, the service distribution system is located such that the standard telephone service and telephone/DSL service are routed from the service distribution system without being routed through a cross-connect system. In other embodiments, the functionality of a cross-connect system is incorporated into the service distribution system. The service distribution systems include terminating modules and switching modules. In some embodiments, the switching modules include protector plugs that protect the system from over-voltages and/or over-currents, and provides upstream and downstream diagnostic testing.

Abstract: A method and system are provided for preventing data loss in a VoIP system. In particular, during a VoIP call, it is determined whether incoming ringing on a POTS line causes an unacceptable level of signal loss or errors. If so, for subsequent VoIP calls, the CO handling calls to the POTS line is instructed to either answer each call with a busy signal or automatically forward calls to the POTS line to the VoIP line or other selected telephone. Calling returns to normal upon ending of the VoIP call. In this manner, incoming ringing on the POTS line does not result in call dropping or lengthy retraining processes.

Abstract: The invention concerns a data transmission technique for real-time applications and non real-time applications in a communication network comprising several nodes connected by communication channels. The invention is characterized in that the data are cyclically and deterministically transmitted, the data for the real-time applications being processed in priority, such that, during one transmission cycle, all the data for the real-time applications are first sent, then, in the time interval remaining prior to the next transmission cycle, the data for non real-time applications are sent.

Abstract: The present invention provides a mobile phone system, which can give a desired voice message to a user of an opposite party terminal without speaking in a state that limitation to usable voice messages has been reduced, while allowing a user existing under an environment that he/she can not speak to listen to a voice from the opposite party terminal. A mobile phone terminal 1 is provided with means 12 for transmitting a connection request of a packet switching line to the side of a network and means 11 for transmitting data corresponding to a desired voice message, while the network is provided with data/voice message converting means 4 for converting transmitted data to a voice message corresponding thereto and means for merging the converted voice message to a circuit switching line of the opposite party terminal to transmit the same.

Abstract: A system for automatically monitoring provisioning in a DSL network is disclosed. The system is operable to query network elements in the DSL network for information related to the provisioning of DSL services. Thereafter, the system analyzes the information to identify points of delay and/or failure in the provisioning process. If a delay exceeds a threshold value, an administrator is notified.

Abstract: An Internet controlled telephony system employing a host services processor connected to a subscriber via the Internet and further connected to the public switched telephone system (PSTN). The subscriber employs a web interface to populate a database with preference data which is used by the host services processor to handle incoming calls and establish outgoing telephone connections in accordance with the preference data provided by the subscriber. Incoming calls to a telephone number assigned to the subscriber may be automatically forwarded to any telephone number specified by the preference data.

Abstract: A method for reducing bandwidth when transmitting data between a sending terminal and a receiving terminal over a voice connection path using a digital data network. The coding of user information is converted within the digital data network between data transmission in the voice connection path and data transmission in the digital data network so that the user information is transmitted on one section of the digital data network via the coded voice connection path and the user information is transmitted on another section of the digital data network using a method suitable for the digital data network.

Abstract: A data session with a wireless device ordinarily cannot be initiated from the network side because the wireless device has no predetermined IP or other network address. Methods are disclosed for instead causing the wireless unit to initiate a data session with a target destination, thereby obviating the IP addressing problem. A voice call is first initiated by a requester application to a remote wireless unit. The requester transmits a request message to the remote wireless unit via the wireless voice channel using in-band techniques; preferably including a payload in the polling message that identifies the target destination. In response, the remote wireless unit initiates a packet data session with the target destination.

Abstract: Upon receiving a call while receiving a digital broadcast, an audio path switching control unit switches an audio path to an output side of a telephone audio processing unit, and outputs telephone audio data received, when the user presses the call start/end button to end the call, the audio path switching control unit switches the audio path to the side of a digital broadcast audio processing, and outputs audio data extracted from the digital broadcast.

Abstract: One embodiment of the present invention uses an abbreviated blanking period, in comparison to the standard VESA and CEA-EIA blanking periods, in order to send data, including low bandwidth, non-timing information, over one or more channels of the digital video link. By shortening the blanking period, the amount of time available for sending data in each scan line is increased, enabling the system to send more data over each channel. The inactive video portion of a scan line sent during vertical sync may also be used to send additional digital data. Shortening the blanking periods and/or using the inactive video sections of the horizontal scan lines adds to the overall data capacity of the link and may be used to send other digital data, such as multichannel audio, video, control, timing, closed captioning or other digital data.

Abstract: A system and method for delivering digital subscriber line (DSL) service to a subscriber. In a telecommunications network, the signal passing through a protector field is diverted to a cross connect block that can be selectively configured to establish a communication route between the cross connect block and a splitter selectively receiving a DSL signal.

Abstract: A multi-carrier CDMA radio transmitting method replicates each information symbol and disposes thus-obtained information symbols along a frequency axis, multiplies the thus-obtained information symbols with a spreading code along the frequency axis, spreads the information symbols into components of a plurality of sub-carriers having different frequencies. The transmission rate of the information is changed by controlling the amount of information transmitted simultaneously by controlling the number of the information symbols to be used for spreading for the plurality of sub-carrier components for each user to which the information is to be transmitted.

Abstract: An integrated voice and data application delivery system that can deliver converged voice/data application services to at least one user device over at least one communications network, and optimize the delivery of voice/data content during a current voice/data session based on the bandwidth of the network and/or the capacity of the user device. The integrated application delivery system allows users to switch from one voice/data session to another in response to a predetermined event, as required to avoid a loss of information or a permanent interruption of service. The integrated voice and data application delivery system includes a unified XML-based integrated voice and data media converging switch, a controller coupled to the switch, and a data storage device coupled to the controller. The switch transfers voice/data content between voice/data applications on a first network and one or more user devices on a second network.

Abstract: A method for utilizing a household's existing telephone line wiring and existing analog telephone sets to make VoIP calls and PSTN calls is provided. The method allows the installation of a simple VoIP routing device in the existing wiring without restructuring of the existing wiring or a deployment of a totally separate network. The method deploys a number of mapping devices to take advantages of the un-used wires in the existing wiring. The mapping devices are installed between the wall jacks and the phone sets, and between a wall jack and the VoIP routing device. The installation of the mapping devices is easy and requires no special tools and experienced personnel.

Abstract: A method and apparatus for enabling a local area network wiring structure to simultaneously carry digital data and analog telephone signals on the same transmission medium. It is particularly applicable to a network in star topology, in which remote data units (e.g. personal computers) are each connected to a hub through a cable comprising at least two pairs of conductors, providing a data communication path in each direction. Modules at each end of the cable provide a phantom path for telephony (voice band) signals between a telephone near the data set and a PBX, through both conductor pairs in a phantom circuit arrangement. All such communication paths function simultaneously and without mutual interference. The modules comprise simple and inexpensive passive circuit components.

Abstract: A method for encoding ATS-QSIG standard-based content for transmission over a 56 kb/s clear channel. The method encodes each of the three 16 kb/s ATS-QSIG voice sub-channels on to corresponding 16 kb/s voice sub-channels in the 56 kb/s clear channel. The method encodes the 16 kb/s ATS-QSIG data sub-channel on to a 8 kb/s data sub-channel in the 56 kb/s clear channel. Using the method, the ATS-QSIG standard can be used in a network such as, for example, the North American PSTN that utilizes 56 kb/s clear channels. In addition, a system for adapting ATS-QSIG standard-based content transmitted on a 64 kb/s clear channel for transmission on a 56 kb/s clear channel and a system for adapting ATS-QSIG standard-based content transmitted on a 56 kb/s clear channel for transmission on a 64 kb/s clear channel. The systems can be used, for example, as gateways to permit ATS-QSIG standard-based communications between 56 kb/s and 64 kb/s based networks.

Abstract: An Internet controlled telephony system employing a host services processor connected to a subscriber via the Internet and further connected to the public switched telephone system (PSTN). The subscriber employs a web interface to populate a database with preference data which is used by the host services processor to handle incoming calls and establish outgoing telephone connections in accordance with the preference data provided by the subscriber. Incoming calls to a telephone number assigned to the subscriber may be automatically forwarded to any telephone number specified by the preference data.

Abstract: A method is described by which digital data can be sent between mobile phones concurrent with speech during a same phone call. The digital data can be from a digital camera that is integrated with the mobile, or cellular, phone. The discontinuous transmission system of the mobile phone system is modified to include a transmission handler that is used to control the packetizing of the digital data to be transmitted within the voids in the transmitted speech determined by a voice activated detector. A reception handler is added to the discontinuous transmission system to allow the extraction of the digital data from the transmitted mobile phone system signal.

Abstract: A method and apparatus for enabling a local area network wiring structure to simultaneously carry digital data and analog telephone signals on the same transmission medium. It is particularly applicable to a network in star topology, in which remote data units (e.g. personal computers) are each connected to a hub through a cable comprising at least two pairs of conductors, providing a data communication path in each direction. Modules at each end of the cable provide a phantom path for telephony (voice band) signals between a telephone near the data set and a PBX, through both conductor pairs in a phantom circuit arrangement. All such communication paths function simultaneously and without mutual interference. The modules comprise simple and inexpensive passive circuit components.

Abstract: A system receives multiple data samples and determines time stamp values associated with each of the multiple data samples. The system identifies an earliest time stamp value and uses that time stamp value as an initial system clock value. The system may also subtract a delay factor from the earliest time stamp value to account for delay in decoding the multiple data samples. The earliest time stamp value is also communicated to an audio decoder and a video decoder. The audio and video decoders provide clock data back to a component that maintains the system clock value. If the difference between the clock data received from the decoders and the system clock value exceeds a threshold value, the system clock value is recalculated.

Abstract: An integrated Set-Top Box DSL VOIP WIFI device including a housing; a telecommunications line input for accepting a telecommunications line containing a video content signal and a data content signal from a provider, the data content signal for providing the voice over internet protocol to a telecommunication's device; a virtual local area network for splitting the video content signal and the data content signal; a video output for accepting a video line for transmitting the video content signal to a video device; a data output for accepting an audio line for transmitting the data content signal to at least one of a telecommunications device and a digital computing device; an antenna for transmitting the data content signal to a wireless communications device; and a passive pass-through filter for connection for providing transmission of E911 service voice signals.

Abstract: Video, voice and data are integrated using existing telephone connections by using a video network server to receive and transmit video signals to and from clients using telephone wires. Client frequency division multiplexing couplers, each associated with one of the clients, receive video and telephone signals from the client and frequency multiplex them onto the telephone wires for transmission. They also receive video signals from the server and frequency demultiplex them for transmission to the client. A video frequency division multiplexing coupler connected between the video network server and each of the client frequency division multiplexing couplers receive the video and telephone signals from the clients and transmit only the video signals to the video network server.

Abstract: A method for providing and transmitting data in a data transmission system by transmitting several data blocks from a data-transmitting device to terminals. Before transmission of the data to the terminals an interleaving process of at least two data blocks is conducted. For the purpose of interleaving for each data unit of the data blocks, a transmission time is established with the help of an algorithm in such a way that for the transmission of k time critical data blocks with k=2, 3, 4, . . . and for one non-time critical data block, the transmission times of the data units of each time critical data block are respectively set at a fraction 1/k of the transmission time of the non-critical data block.

Abstract: A network for coupling at least one telephone service signal to at least one telephone device over a wiring. The network includes: a wiring having at least two conductors for carrying multiple time-domain multiplexed digitized voice channels; an exchange side device coupled to the wiring and operative to couple at least one telephone service signal to at least one digitized voice channel; and at least one subscriber side device coupled to the wiring and operative to couple the at least one telephone device to at least one digitized voice channel.

Abstract: A multi-carrier CDMA radio transmitting method replicates each information symbol and disposes thus-obtained information symbols along a frequency axis, multiplies the thus-obtained information symbols with a spreading code along the frequency axis, spreads the information symbols into components of a plurality of sub-carriers having different frequencies. The transmission rate of the information is changed by controlling the amount of information transmitted simultaneously by controlling the number of the information symbols to be used for spreading for the plurality of sub-carrier components for each user to which the information is to be transmitted.

Abstract: After a predetermined period of time has lapsed after transmission of a voice data packet representing a first voice section a second voice data packet representing the same voice section is transmitted. On the receiving end, a value for the transmission quality is determined for the first and for the second voice data packet and the one of the two voice data packets is processed that has the better transmission quality.

Abstract: A control method of synchronizing communications between or among a plurality of devices in a communication system includes detecting beacons from the plurality of devices in the communication system, and establishing a reservation for at least a portion of the plurality of devices in the communication system. Each reservation is a frame interval in which to transmit symbols from one device to one or more of the other devices in the communications system. Each frame interval and intra-frame interval is set according to the established reservation. Each frame interval and intra-frame interval is a plural, integral number of symbol periods in duration.

Abstract: A method for dynamically varying the allocation and distribution of a base station's resources in response to fluctuations in demand for services. The method includes transmitting a status signal from a base station to a base station controller over an Iub interface. The status signal may correspond to the demand of the base station's transmit power for a data transmission service, such as HSDPA, and for at least another service, such as a dedicated voice channel. In response to receiving the usage information signal, the base station controller may vary the transmit power of the base station allocated for HSDPA and/or for the dedicated voice, data and/or both integrated voice and data channel. In response to receiving users' status information for HSDPA services, the base station controller may vary its call admission status for HSDPA and/or dedicated voice, data and/or both integrated voice and data users.

Abstract: A method and apparatus for user data traffic handling for an EVDO enabled hybrid mode mobile device, the method having the steps of: blocking a system selection at the mobile device until no EVDO traffic state exists, the blocking step includes either monitoring whether a user changes a scan mode on the mobile device; and if the user changes the scan mode, checking whether the mobile device is in an EVDO traffic state; and if yes delaying a system selection until the EVDO traffic state is ended, or monitoring whether a user is in an EVDO traffic state, and if yes, preventing a user from accessing a scan mode menu on the mobile device.

Abstract: A method and system for communicating information via a plurality of different networks may comprise requesting in a mobile terminal capable of receiving and processing cellular frequency band services and VHF/UHF broadcast frequency band services, media having a specified quality of service. A description of said requested media and/or a quality of service associated with the requested media may be displayed via a user interface in the mobile terminal. An input may be received in the mobile terminal that indicates at least one required quality of service associated with the requested media. Based on information stored in the mobile terminal, at least one required quality of service associated with the requested media may be determined. One or more required quality of service associated with the requested media may be stored in a profile within the mobile terminal.

Abstract: Presently described is a system and method for switching multimedia data communications, including but not limited to Voice over IP (VoIP) telephony, cable TV, digital audio and video. The system utilizes a single, integrated device to provide all Packet Cable-compliant functionality, including enhanced user privacy, compliance with CALEA, E911 and other mandated services not available in conventional distributed Packet Cable systems. High speed and efficient, low cost operation are provided by means of an optimized data unit encapsulation scheme for internal switching and routing. A proprietary fiber optic backplane and removable optical connectors are used to enable light speed internal communications hot-swapping of components. Furthermore, the present system is extensible to all forms of digital data switching and is secure, resistant to Denial of Service attacks, and fault-resilient.

Abstract: Systems and methods for monitoring, making, managing and controlling telephone communications with a computer call assistant with an integrated voice/data communications system are disclosed. A call assistant computer application preferably runs on a personal computer (“PC”) coupled to the integrated system over a packet bus. The call assistant exchanges control and/or status packets with the integrated system preferably over a packet bus. The call assistant enables the user to make, receive and control telephone calls, monitor the status of the user's extension, voice mail, etc., and preferably operates with integrated systems capable of transmitting and receiving voice and data in multiple modes.

Abstract: The hybrid DSL system (10) for transmission of hybrid DSL data via a subscriber access line (2030) has a baseband transceiver (101) for transmission of first data in a base frequency band via the subscriber access line (2030) at a first data transmission rate, a non-baseband transceiver (102) for transmission of second data in a non-base frequency band via the subscriber access line (2030) at a second data transmission rate, and a data link (103) for interchanging parameters between the baseband transceiver (101) and the non-baseband transceiver (102) in order to increase the overall data transmission rate of both transceivers (101, 102).

Abstract: System and method for partitioning a DSLAM network. In this regard, one such network can be broadly summarized by a representative communication system comprising a digital subscriber line access multiplexer (DSLAM) that is communicatively coupled on its line-side to a high-speed digital link. The DSLAM is communicatively coupled through the high-speed digital link with the trunk side of a remote line access unit (RLAU). The RLAU is communicatively coupled on its line-side to a first digital subscriber line (DSL).

Abstract: A system and methods for providing services through an integrated digital network. The present invention embodies a system and methods for providing telecommunications services through an integrated digital network that provides the superset of all previous network types, and can continue to support being the core for existing networks—now used simply as edge access methods. The method includes connecting a multi-service core to existing telecommunications networks, wherein the core can communicate with conventional network components; interconnecting a gateway between the multi-service core and multiple existing network components, wherein the gateway is configured to share multiple service aspects with each network component and to utilize multiple transport protocols to communicate with different core network components; coordinating communications between the gateway and existing network components; and resolving differences between protocols or communications between existing core network components.

Abstract: An assembly (10) of three telecommunications modules (12, 14, 16) each having an equal number of contacts for connecting wires (18, 20, 28) therewith, comprises splitter circuits (30?, 30?), the number of splitter circuits (30?, 30?) being equal to half of the number of contacts of each module (12, 14, 16), one third of the contacts (24, 74) of the assembly (10) being adapted to transmit a line signal, one third of the contacts (22, 72) of the assembly (10) being adapted to transmit a POTS signal, and one third of the contacts (26) of the assembly (10) being adapted to transmit a DSLAM-signal. A module, particularly for use within an assembly, is open at a side other than a front side, at which contacts are exposed, so as to allow at least one splitter circuit (30?, 30?) to be at least partially inserted into the module.

Abstract: In order to measure the rate at which a terminal 2 can receive data from a source 5, 6, 7 the source temporarily increases it transmitting rate. The time taken to receive these higher-rate transmissions is measured at the terminal (or acknowledged by the terminal and measured at the transmitter) and the rate calculated. Inter alia, these measurements can be used for control of layered coding where apparatus for transmitting signals to video or other terminals 2 has coding means 5 for receiving the signals to generate a base signal stream B which is a coarse representation of the signal to be transmitted and at least one higher order stream E1, E2 containing finer detail of the signal not present in any lower order stream. These streams are sent by transmitting means 6 to the terminals, but at least some of the terminals do not receive all of the higher order streams.

Abstract: In some embodiments, an apparatus includes a connect control module adapted to receive a dynamic service addition message containing a reserved bandwidth amount of a reserved bandwidth allocation for an uplink service flow. The reserved bandwidth allocation is capable of accommodating two or more Voice over Internet Protocol (VoIP) calls and the uplink service flow is substantially in accordance with an Institute of Electrical and Electronic Engineers (IEEE) 802.16 standard. The apparatus also includes a call session module coupled to the connect control module and adapted to generate a connection request message for a VoIP connection. The connect control module, in response to the connection request message, is further adapted to send a dynamic service change request message to activate a call-session bandwidth portion of the reserved bandwidth allocation for the VoIP connection.