Abstract: A system and method for delivering digital subscriber line (DSL) service to a subscriber from a remote terminal of a telephone network. Included is a first route for delivering a telephone signal to the subscriber, and a second route for passing the telephone signal through a DSL system before delivering the signal to the subscriber. A disrupter is then used to selectively activate either the first or second route.

Abstract: A data coding device for coding data, to be transmitted as part of data communication, to signals of a first frequency range, not used for voice communication with a first set of performance features, and a conversion device are provided. The conversion device has an interface for connecting a voice terminal having a second set of performace features and for transmission of data coded to signals of a second frequency range that are to be transmitted with the second set of performance features as part of voice communication. The conversion device also has a coding means for recording signals of the second frequency range to signals of a third frequency range that is located outside the first frequency range. Furthermore, a frequency-separating filter for feeding signals of the first and third frequency range into the subscriber line is provided.

Abstract: An IP PBX system that serves enterprise telephones via a landline IP network connection is be expanded to serve cellular wireless communication devices (WCD) via a cellular wireless carrier's radio access network (RAN). Calls to and from the cellular WCD are connected through the cellular carrier's RAN and the IP PBX system, so that the IP PBX system can control and manage the calls just as the IP PBX would control and manage calls involving other extensions on the IP PBX. A cellular WCD thereby becomes an IP PBX client station, i.e., an extension on the IP PBX system. As such, the cellular WCD can seamlessly benefit from many of the same IP PBX features that other more conventional IP PBX client stations (e.g., desk phones) enjoy.

Abstract: The present invention relates to a method for data-signal quality control, a related receiver and a related transmitter in an XDSL communications system, where this XDSL communications system comprises a transmitter and a receiver. The transmitter is coupled to the receiver over an XDSL line, over which the transmitter sends a data-signal on a plurality of data carriers for carrying this data-signal from the transmitter to the receiver. This receiver analyses a quality of the data-signal received by the receiver and subsequently requests the transmitter, in case this quality of the data-signal is beneath a predefined level to improve the quality of the data-signal, by performing an equal transmit power increase over a plurality of the data-carriers. The transmitter in response to the requesting improves the data-signal by performing an equal transmit power increase over at this plurality of data-carriers.

Abstract: A telecommunication system uses at least one dual mode channel capable of supporting variable-rate real-time data (e.g. speech) and non-real-time packet data. The packet data is queued in a buffer and transmitted when the data rate of the real-time data stream is less than the full data capacity of the channel.

Abstract: A playback system in a digital cinema network synchronizes the presentation of visual and aural content by deriving timing information for packets of information that are conveyed in video and audio data streams, examining the timing information to determine if any misalignment between the two data streams is likely to be perceptible and, if the misalignment is deemed to be perceptible, introducing delays into one or both data streams to correct the misalignment. If the audio data stream precedes the video data stream, the audio data stream is delayed by an integer number of audio sample periods. If the video data stream precedes the audio data stream, the video data stream is delayed by an integer number of video frames and the audio data stream is delayed by an integer number of audio sample periods.

Abstract: A method of signal communication includes receiving a selection signal, and selecting a selected frequency range used for a multi-carrier signal communication from a set of predetermined frequency ranges for signal communication depending on the selection signal. The set of predetermined frequency ranges includes a first frequency range and a second frequency range including the first frequency range.

Abstract: This invention provides a simplified communication system that transmits voice and SDSL signals between customer premise equipment (“CPE”) on a single twisted-pair line. The system preferably includes at least one combiner for combining voice and SDSL signals into a combined voice/SDSL signal that can be transmitted to the CPE on a single twisted pair of wires, preferably a pair of wires currently being used to transmit the voice signal to the CPE. One combiner is ideally located at a common wiring closet of the building where the CPE is located. The combiner is ideally coupled to the existing pair of wires for the voice signal. This eliminates the need to add another set of wires between the common wiring closet and the CPE for the SDSL signal. A second combiner, normally located at or near the CPE separates the combined voice/SDSL signal into the separate voice and SDSL signals for processing by the CPE.

Abstract: Aspects of the present invention are directed at establishing a multimedia network session in which the transmission of media streams is synchronized. In one embodiment, a method is provided for synchronizing incoming audio and video streams. The method includes establishing a communication channel between a first computing device that is receiving an incoming audio stream with the second computing device that is receiving an incoming video stream. Once the communication channel is established, the current network conditions that describe attributes of the incoming audio stream are obtained by the first computing device. Then, the delay in the incoming audio stream is calculated. When the delay is known, the method causes the incoming video stream to be delayed to match the delay in the incoming audio stream.

Abstract: A method for processing cells at a user-network interface with automatic identification of virtual circuit identifiers and a testing function is provided. The method includes distinguishing the source of the cell. When the cell is from a first source, the method tests the cell against at least one selected criteria. When at least one test determines the cell is invalid, the cell is marked. When the tests determine that the cell is valid, the method translates a virtual circuit identifier to a default setting and forwards the cell to a queue for further processing.

Abstract: The present invention is directed to a system for reliably transmitting characters encoded as audible signals using voice over Internet protocol communication systems. According to an embodiment of the present invention, tones within a sequence of tones encoding a textual character are transmitted using two data packets. The first data packet includes a segment of time having a starting boundary that is synchronized with the start of the encoded tone and extending for a time period equal to about one-half the total time period of the tone. The second packet encodes the remaining half of the audible tone. At the receiving end, data packets encoding a textual character are ordered according to the position occupied by the data encoded by the packet within the sequence of tones. If any unpaired packets of data are detected at the receiving end, the characteristics of the packets that are unpaired are copied into the time slot that would have been occupied by the missing packet.

Abstract: The present invention provides a novel system, method and apparatus for allocating power between at least two communication services that share a common power output limit. Two communication services particularly suited for the present invention are voice services and data services transmitted on the downlink of a wireless network. An embodiment of the method includes determining the actual consumption of power on the voice channels of the wireless network during a given time period, and allocating substantially the same amount of power to the voice channels for the next time period, thus allowing for the allocation of the remaining amount of power to the data services, and thereby allowing, for example, increased modulation of the data services and thereby improve overall rates of data transfer and/or reliability of data transmission.

Abstract: A method and an intermediate element, or a gateway, for routing an information flow in a telecommunication system including a plurality of switching centres. The switching centres are connected with an Internet Protocol (IP) based connection via the gateways, which are arranged to perform speech coding/decoding for an information flow between the switching centres. The gateways are further arranged to detect if the information flow from the switching centre is a data call and to omit the speech coding if the information flow is a data call. The data call detection is performed based on of a symbol pattern included in the frame structure of the information flow.

Abstract: A Voice-over-Internet-Protocol (VoIP) system includes clients that make management connections to an external manager server using web-browsing protocols. A discovery process tests a variety of transport methods by sending packets using different protocols to several ports of the external manager. Some packets may be blocked by firewalls, while other packets are allowed to pass through. Successful packet replies indicate transport capabilities of the client. When a client makes a VoIP call to another client, the external manager compares transport capabilities of the two clients and selects a transport capability that is common to both clients. Transport methods include direct client-to-client connections using TCP or UDP to an arbitrary port or well-known port 80 or 443, and indirect using a voice-proxy server that each client connects to using web standards such as HTTP or SSL. When a client can make only outgoing connections, that client initiates the connection.

Abstract: A method for communicating data in a wireless communications system having a base station communicating to at least one of a plurality of terminals over a set of carrier frequencies, the method comprising periodically monitoring the carrier energy level of each said carrier frequency received at a given terminal to identify a list of optimal frequencies over which the carrier energy level is strong enough to maintain a viable transmission from said base station to said given terminal, wherein said viable transmission provides a given coding gain; and simultaneously transmitting a given set of packet data from said base station to said given terminal over a plurality of frequencies from said list of optimal frequencies identified for said given terminal.

Abstract: A multiformat-compatible receiving apparatus is quickly compatible with a change in the signal type of a received signal. Responsive to the detection of a stream change to an invalid data stream (ancillary no data), a signal processing means is controlled to have a setting compatible with the signal type of the invalid data stream (F104?F105). Thus, the setting of the signal processing means is predictively changed so as to be compatible with the signal type of real data to arrive next. When a stream change to a real data stream is actually detected, a receiving means is controlled to have a setting compatible with the signal type of the real data stream (F107).

Abstract: An apparatus for providing fault and error tolerance in a network environment is provided that includes an aggregation node associated with a base station controller and operable to receive a packet that includes an error. A portion of the packet is discarded, the packet being associated with a communications flow. The aggregation node being further operable to perform a selected one of injecting a subframe into the communications flow and playing one or more previous frame bits from a frame memory in response to the error.

Abstract: A novel single chip tuner integrated circuit (IC) for multiple video reception in a cable modem. The tuner circuit is well suited for use in cable modem systems adapted to implement the DOCSIS 3.0 specification which specifies multiple simultaneous video channel reception. The single-chip tuner integrated circuit comprises a plurality of tuner sub-circuits wherein each tuner sub-circuit is operative to generate a single output channel, determined in accordance with a tune commend input. The tune command input is used to generate a frequency reference signal that is input to a corresponding tuner sub-circuit. The frequency reference signal for a tuner sub-circuit is mixed with an input RF receive signal from the CATV input to the cable modem to generate a baseband channel. The baseband channel is subsequently filtered, amplified and input to the PHY circuit wherein undergoes analog to digital conversion (ADC) before being input to a baseband processor.

Abstract: An external connection interface emulation method and apparatus of a mobile terminal for exploiting the mobile terminal as a network adaptor of another terminal are provided. The external connection interface emulation apparatus includes a network modem for supporting communication through the mobile communication system; and a controller, which configures the network modem for operating the mobile terminal as an external network adaptor of another terminal or a communication device as it is, according to an input signal. The external connection interface emulation apparatus and method establish a communication link between a WiBro modem and a USB port at the software level making it possible to use the mobile phone as an external modem regardless of whether the WiBro modem and the USB port are compatible or not.

Abstract: The present invention advantageously provides a method and apparatus for optimization of channel capacity in multi-line multi-tone communications such as X-DSL among subscriber lines which are bundled with one another. In an embodiment of the invention an apparatus for optimizing channel capacity of multi-tone communications effected by opposing sets of modems coupled to one another by a plurality of subscriber lines is disclosed. The apparatus includes a spectrum manager coupled to at least one of the opposing sets of modems. The spectrum manager includes: a profiler, a demand module and an optimizer. The profiler obtains from the at least one of the opposing sets of modems the spectral characteristics of each of the plurality of subscriber lines. The demand module determines for each of the plurality of subscriber lines the subscriber demand profile.

Abstract: The invention relates to a line terminating device for a subscriber line which transmits and receives broadband signals via a single subscriber line, a broadband signal being composed of a broadband or narrowband audio-frequency data signal and a broadband higher-frequency data signal and the frequency bands of the voice signal and of the data signal essentially not overlapping. The line terminating device according to the invention has a digital frequency separating filter in the digital section of the line terminating device, which is arranged in the digital section of the line terminating device so that the audio-frequency voice signal is separated from the higher-frequency data signal. In particular, the line terminating device is suitable for separating an ISDN or POTS voice signal from an ADSL data signal.

Abstract: In a communication system for Internet protocol (IP)-based transmission of information formed of a speech part and a data part between interface units with a plurality of ports, the speech part and the data part are transmitted to separate ports of the receiving interface unit. The ports that used are allocated to each other in the communication system as corresponding ports.

Abstract: The home gateway system (60) for home automation and security has a wireless local loop transceiver (62). A home automation controller (64) is capable of sending and receiving a message with the wireless local loop transceiver (62). A home security controller (66) is capable of sending and receiving a message with the wireless local loop transceiver (62) and the home automation controller (64).

Abstract: An integrated access platform includes a first port configured to receive a first signal from a plain old telephone service (POTS) and a second port configured to receive a second signal from an asymmetric digital subscriber line (ADSL). The integrated access platform also includes a controller system configured to process the first signal and the second signal for access with external networks.

Abstract: An information distributing method etc. are provided. An information providing server comprises a first transmission device for transmitting image data to a communication terminal, a first conversion device for converting attached information attached to the image data into converted information of a format suitable for aural read-out, a second conversion device for converting the converted information into voice information, a second transmission device for transmitting the voice information and/or the converted information to the communication terminal, and a third transmission device for transmitting synchronizing information for synchronizing image data with the voice information on a screen of the communication terminal.

Abstract: There is provided a DSL modem for integrating an analog voice band with DSL bands over a telephone line, where the analog voice band consists of 0-4 kHz frequency range, and the DSL bands consist of frequencies beyond 4 kHz. The DSL modem comprises a splitter configured to receive a telephone line signal from the telephone line and split the telephone line signal into an analog voice band signal and a DSL bands signal; a voice module configured to receive the analog voice band signal, digitize the analog voice band signal to generate digitized voice band signal and packetize digitized voice band signal into packetized voice band signal; a DSL module configured to receive the DSL bands signal and retrieve packetized DSL bands signal from DSL bands signal; and a multiplexer configured to multiplex packetized voice band signal and packetized DSL bands signal to generate multiplexed voice-DSL packets.

Abstract: A signal transmitter and a signal receiver comprises a time base compression part (101) for compressing an audio signal on a time base to output the compressed signal as a time base compressed audio signal and a multiplexing part (102) for multiplexing a video signal, a control signal, and the time base compressed audio signal to output the multiplexed signal to the external as a video/audio control multiplexed signal. The thus constituted signal transmitter and the signal receiver realize a signal transmission system for multiplexing and transmitting the video signal, audio signal, and control signal, for consequently transmitting high-quality diverse digital audio signals without conversion into an analog signal, and for transmission at high speed with minimal error.

Abstract: Transmission data are transmitted via a radio interface between an analog telephone link and a user apparatus. A base station for a cordless telephone that is connected to the telephone link has a converter that converts the analog signals into digital signals and vice versa. The digital signals are transmitted, in particular according to the DECT standard, via a radio interface between the base station and a mobile station. In the mobile station, the digital signals are decoded, and thus the transmission data are acquired, and vice versa.

Abstract: According to the data amount stored in a jitter buffer, which temporarily stores obtained data from a received voice packet, an operation timing of an output process of the jitter buffer and a data process of a codec and a modem is adjusted, the codec and the modem processing data that is sequentially output from the jitter buffer.

Abstract: A method for transmitting speech and data simultaneously on a circuit switched two-way connection of a mobile communications system, which transfers speech and other data by use of discontinuous transmission, DTX, in which the sending is discontinued when it is indicated, that there is no information to be sent, and continued when it is indicated that there is information to be sent. In accordance with the method of the invention, the data to be sent are stored in buffer means and wait to be sent. When it is indicated, that there is no more information to be sent, the data stored in buffer means are sent until there is no more data to be sent or until it is indicated that there is information, speech or other data, to be sent, in which case the sending of data is discontinued and the sending of speech or other data is re-started. So the transfer of data takes place as a background process during transfer of speech or other data. The data are arranged into frame structures while being stored in the buffer.

Abstract: A method and apparatus for a computer and telecommunication network which can receive, send and manage information from or to a subscriber of the network, based on the subscriber's configuration. The network is made up of at least one cluster containing voice servers which allow for telephony, speech recognition, text-to-speech and conferencing functions, and is accessible by the subscriber through standard telephone connections or through internet connections. The network also utilizes a database and file server allowing the subscriber to maintain and manage certain contact lists and administrative information. A web server is also connected to the cluster thereby allowing access to all functions through internet connections.

Abstract: Disclosed herein is a system for automatically upgrading the firmware of Internet video phones and a method of managing the system. The system includes one or more Internet video phones, an upgrade server, and a Trivial File Transfer Protocol or Hypertext Transfer Protocol (TFTP/HTTP) server. Each of the Internet video phones has information, including a company name, a language type, a version and an Internet Protocol (IP) address, and transmits information, including changed IP information, via an Internet when the IP information changes. The upgrade server performs an update using information transmitted from the Internet video phones and transmits upgrade commands to the Internet video phones when new firmware is distributed. The TFTP/HTTP server transmits the new firmware, which is requested by the Internet video phones in response to the upgrade commands, to the Internet video phones that requested the firmware.

Abstract: According to one embodiment of the invention, a method for multicasting data includes receiving, at a line card multicast data and an indication of a plurality of destinations for the multicast data. The method also includes transferring the multicast data over a common bus to a plurality of digital subscriber line chipsets associated with the plurality of destinations. Such transfer occurs by selecting two or more of the plurality of digital subscriber line chipsets to receive multicast data, enabling the two or more selected digital subscriber line chipsets to receive the multicast data, and transferring the multicast data over the common bus to the selected digital subscriber line chipsets after selection and enabling of the two or more of the plurality of digital subscriber line chipsets.

Abstract: A tester and related method test an xDSL transceiver unit-central office so as to measure performance of a transceiver unit by automatically converting many output ports by connecting a multiplexer to the xDSL transceiver unit.

Abstract: An interactive for-hire vehicle communication system and method of communicating selectable passenger-related information to a for-hire vehicle are disclosed. The system comprises an audio device located in a for-hire vehicle. A workstation is in communication with the audio devices and the workstation is also in communication with one or more passenger interfaces located in the for-hire vehicle. Either the workstation and/or servers are configured to receive passenger audio information received by the audio device. The passenger audio information is converted to text suitable for passenger-related interactive information exchange to one or more passenger interfaces of the for-hire vehicle. The system also may include a video device configured to receive image information suitable for transmission to one or more servers to provide interactive information exchange to passenger interfaces of the for-hire vehicle.

Abstract: A method and apparatus for transmitting delay sensitive information (DSI) and non-delay sensitive information (NDSI) over a communication link where initial DSI packets are delayed to reduce the number of conflicts between DSI and NDSI. The delay applied to the DSI is based on parameters such as the transmission periodicity of the DSI packets and the packet length of the NDSI being transmitted.

Abstract: A data communication apparatus is provided with a VoIP processor and a modem that interchangeably executes a standard communication mode and a high-speed communication mode, the standard communication mode exchanging the voice packet at a predetermined transfer rate, the high-speed communication mode exchanging the voice packet at a rate faster than the rate of the standard communication mode. When capability information of an opposing apparatus is obtained, in accordance with a predetermined communication protocol, and it is detected that the opposing apparatus has the high-speed communication mode, the high-speed communication mode is selected for execution.

Abstract: A method and device for communicating multiple communications networks from a single chassis is disclosed. In an exemplary embodiment, the communication device includes a processing unit, a network interface, a service interface, an endpoint interface, and a cluster interface. The processing unit controls various data flows in the communication device. The network interface communicates to at least one other communications network. The endpoint interface provides communication to local end devices and between local systems. While the service interface is designated to enhance service capabilities, the cluster interface is devised to expand system capacity. In this embodiment, the processing unit, the network interface, the service interface, the endpoint interface, and the cluster interface are assembled in one unit.

Abstract: A system includes telephony devices that provide services and applications through the execution of scripts in web pages.

Abstract: A communication management system operates with a packet switched local area network and includes a PSTN interface for communicating over a telephone line with a service provider central office. The system comprises means for establishing a logical channel over the packet switched local area network to support a real time communication session with a real time communication device in response to receiving session signaling from the real time communication device; means for controlling the PSTN interface to transition the telephone line to an off hook state in response to receiving session signaling; and translation means for providing digital audio over the logical channel, the digital audio representing a dial tone received on the telephone line from the central office and providing one of a plurality of dual tone multi frequency signals on the telephone line in response to receipt of data from the real time communication device corresponding to the one of a plurality of dual tone multi frequency signals.

Abstract: An XDSL system may include an active unit and a standby unit. The active unit may include a plurality of line circuits, which can be used to form XDSL loops with far end units. The standby unit may be synchronized with the active unit. In response to an indication, such as a fault, the TDM system may switch from the active unit to the standby unit. Then, the standby units may be used to form the XDSL loops with the far end units.

Abstract: Provided are apparatuses and methods for transmitting radiotext plus data (e.g., RT+ data) to a mobile terminal. In one example, radiotext plus data is received and transmitted to the mobile terminal in a Digital Video Broadcast—Handheld (DVB-H) data stream. Program or service data corresponding to a program may be endoded on IP and transmitted in audio/video data in a DVB-H data stream and RT+ data corresponding to the program may be transmitted on Internet Protocol (IP). For example, the RT+ data may be converted to Real-Time Transport Protocol (RTP) on a RTP timed text protocol. In another example, a content stream may be received and divided into a program data stream and a corresponding RT+ data stream. The RT+ data stream may be converted to RTP and synchronized with the program data on audio/video data in a DVB-H data stream. The RT+ data may be delivered to a mobile terminal.

Abstract: A method employs call detail records for developing a report on the usage of VoIP calling along a network path or through a network location. By utilizing, for example, a call-beginning record and a call-ending record for each connection, a report is created of the number of concurrent VoIP calls that a specified equipment or network path carries at any given instance by retrieving batches of stored call detail records, culling the records that are relevant to the specified equipment or network path, considering the culled records in a time sorted manner, and incrementing or decrementing a counter based on whether a considered record is a call-beginning record or a call-ending record, respectively. The state of the counter is reported, illustratively, as a line entry each time the counter's state is modified.

Abstract: IP network communication device which allows facsimile communication conforming to ITU-T recommendation V.34 in the network conforming to ITU-T recommendation T.38. This IP network communication device comprises a call controller for setting the call connection of a voice channel with another communication device, a voice controller which generates packets from voice data and facsimile data, sends the packets to the other communication device, and restores voice data or facsimile data from the packets received from the other communication device, and a digital signal processor which mediates the communication of voice data and facsimile data between an accommodated terminal and the voice controller. The digital signal processor turns ON the facsimile tone detecting function, echo canceling function, and fluctuation absorption buffer function if voice data is communicated, and turns these functions OFF if facsimile data is communicated.

Abstract: An inband signaling modem communicates digital data over a voice channel of a wireless telecommunications network. An input receives digital data. An encoder converts the digital data into audio tones that synthesize frequency characteristics of human speech. The digital data is also encoded to prevent voice encoding circuitry in the telecommunications network from corrupting the synthesized audio tones representing the digital data. An output then outputs the synthesized audio tones to a voice channel of a digital wireless telecommunications network.

Abstract: In preferred embodiments, a system including a transmitter, a receiver, and a serial link, in which the transmitter is configured to transmit video data, embedded-clock auxiliary data (or auxiliary data derived from embedded-clock auxiliary data), and a video clock over the link to the receiver. The transmitter is configured to extract a sample clock from the auxiliary data without use of a phase-locked loop, and to generate time stamp data in response to the sample clock and the video clock. Typically, the auxiliary data are SPDIF (or other) audio data, and the sample clock changes state in response to codes that occur periodically in the audio data. Other aspects of the invention are a transmitter for use in such a system, a time stamp data generation circuit for use in such a transmitter, and a method for generating time stamp data in response to a stream of embedded-clock auxiliary data and a video clock.

Abstract: A network device including a plurality of communication interfaces, including a telephone line interface, a computer data interface, and a broadband network interface; a processor; a machine-readable storage medium which during use stores a call processing application and service profiles, and which stores executable instructions to mediate communications between the plurality of communication interfaces, the instructions causing the network device to detect network signaling events or trigger points in a telephone call and invoke the call processing application in response to the detected network signaling events or trigger points, the call processing application operating according to parameters defined in the service profiles.

Abstract: Network delay is determined in order to synchronize a clock in a mobile station with a reference clock. Tones are generated that represent a sequence of bits in a synchronization flag. The tones are sampled beginning at a selected sample start time. The sampled tones are demodulated to identify the bit values in the synchronization flag. The demodulator is synchronized with the sampled tones in the synchronization flag by shifting the sample start time until the samples generate an optimum synchronization value. A reference time is then identified according to the optimum sample start time.

Abstract: A multi-media terminal adapter couples to a network access module over a communication link. The network access module communicates IP frames over a frame switched network. The multi-media terminal adapter comprises a wide area network interface coupled to the communication link for exchanging IP frames with the access module and a local area network interface for receiving outbound data client IP frames from each of a plurality of data clients. Each outbound data client IP frame comprises a socket information that includes a local area network IP address and a data client port number. A VoIP module generates outbound VoIP frames. Each outbound VoIP frame comprises socket information that includes a VoIP port number selected from a first group of port numbers exclusively reserved for use by the VoIP module. A router module is coupled between the wide area network interface and each of the VoIP module and the local area network interface.

Abstract: A method and system that communicates adaptive transmit-side filter updates between a receiver and transmitter inserts additional versions of control codes into a back channel for encoding updates. Since the control codes are required in the back channel, no additional bandwidth of the back channel is used to communicate the updates.