Abstract: To reduce power consumption of a first line driver/receiver (LD1) adapted to communicate with a second line driver/receiver (LD2) via a transmission line (1) in a multitone communication system, wherein, during a training sequence, the second line driver is adapted to transmit a known multitone signal pattern to the first line receiver, a signal processor (DSP1) connected to the first line driver/receiver (LD1) is adapted to measure the signal-to-noise ratio of the received signal pattern and, in response hereto, determine an appropriate bit rate to be used, to decrease the quiescent current of the first line driver in steps, to detect a decrease of the measured signal-to-noise ratio, and to set the quiescent current of the first line driver to the value it had before the decrease of the signal-to-noise ratio was detected.

Abstract: A method of converting voice signals into voice packets. The method includes providing one or more of voice or fax digital signals, at a first location, processing the provided digital signals at the first location, transmitting the processed digital signals to a second location, adding at least one header to the processed digital signals transmitted to the second location, at the second location and transmitting the processed digital signals with the added header on a communication network.

Abstract: A method and apparatus for performing directory number/Internet Protocol (IP) address resolutions to facilitate voice over IP communications. A message requesting the IP address of an IP gateway is built at an originating Time-Division Multiplex (TDM) switch (that forms part of either a private or public network). The message is forwarded over a TDM telecommunications network to a terminating TDM switch, where the requested IP address information is retrieved. The IP address is forwarded across the TDM telecommunications network to the originating TDM switch. The IP address is then used to establish a call across an IP network. By using existing TDM protocols and infrastructure to request and retrieve IP gateway information, the present invention does not suffer from the scalability problems inherent in other DN/IP address resolution solutions.

Abstract: A method for data communications includes receiving a time-division-multiplexed (TDM) input signal carrying a payload comprising data and determining whether the data comprise synchronous or non-synchronous data. A first encapsulation scheme is selected if the data comprise synchronous data, and a second encapsulation scheme is selected if the data comprise non-synchronous data. The data are encapsulated for transmission over a packet-switched network in accordance with the selected encapsulation scheme.

Abstract: In a digital wireless communication system wireless communication stations or radio base stations constitute frames to be conveyed over an air-interface for simultaneously comprising full rate speech data and dedicated user data. A speech coder in the digital wireless communication system generates the speech data while operating at a full rate. A user data application in the digital wireless communication system generates the dedicated user data. The system can set different service configurations supporting simultaneous transmission of speech data and dedicated user data, and can also include signaling data in the frames.

Abstract: A system and method for associating notes to a call with a portable information device, such as a personal digital assistant. In one embodiment, notes taken during a call are associated with information about the call, such as parties to the call, the date and time, and the subject matter of the call. Notes taken may include user input or PID data communicated across a telephony network, for example. In one embodiment, a portable information device links to a data network telephone to initiate a phone call. Information about the phone call is stored with notes taken to allow for later retrieval.

Abstract: Digital Subscriber Line (DSL) modems allow high-speed data to be sent over telephone lines normally used for analog voice service. In cases of impaired DSL line performance, diagnostic information is passed over a more stable part of the channel, i.e. the voice band channel. For example, analog modems may be used to communicate over the voice band channel. The diagnostic information may include various types of information that the two non-connecting DSL modems see on their respective sides of the line. For example, all the parameters that each of the DSL modems establish during start-up can be part of the diagnostic payload transmitted by the analog modems. Parameters such as settings of frequency domain equalizer, line bit rates, settings of time domain equalizer, SNR, location within startup of start-up failure, signal shape, Reed-Solomon values, etc. may be passed.

Abstract: A wireless communications system comprises a first splitter separating at least two signals, a wireless circuit for processing a first signal as a wireless signal and a combiner connected to the wireless circuit and the splitter, combining the two signals and transmitting the signals into a service area. The wireless communications system further comprises a second splitter separating the wireless signal from a second signal, and a transmitter for broadcasting the wireless signal.

Abstract: A system and method places signaling message frames in a traffic channel carrying voice frames so that the signaling message transmission does not seriously impact voice quality. This is accomplished by producing a first stream of voice encoded packets and a second stream of signaling message packets. The signaling message packets are prioritized in the second stream relative to the voice encoded packets in the first stream. An arbitration element arbitrates between the voice encoded and prioritized signaling message packets, which are multiplexed as a function of this arbitration. The multiplexed voice encoded and prioritized signaling message packets are then transmitted.

Abstract: In the communication system which performs the data communication by the discrete multi-tone modem scheme between a plurality of data communication units using the time-division half-duplex communication function, the ratio between the data transmission time suitable for data communication and the quasi-data transmission time dynamically changes within one period. Further, bits are assigned in such a manner that the data of one period can be transmitted during the data transmission time of that period. Dummy bits are assigned to the portions of the data transmission time to which the data to be transmitted has not been assigned.

Abstract: An apparatus for transmitting voice data on a Next Generation Network Asymmetric Digital Subscriber Line (ADSL) subscriber board and a method thereof enable a Channelized Voice over xDSL (CVoDSL) function as a voice service function of an ADSL subscriber, stable signaling of the CVoDSL, and synchronization of an ADSL subscriber board and a Public Switched Telephone Network (PSTN) to embody the CVoDSL function. The service to the voice subscriber using an ADSL line can be maintained with the same voice quality as that of an existing copper cable subscriber and the same service can be carried out with less expensive equipment. Also, the need for a splitter to provide services using a separate voice band frequency on the ADSL line can be eliminated, and a plurality of users can be provided with voice services by sharing and dividing a service frequency band on the same line.

Abstract: A portable telephone includes a speech transmission circuit, data transmission circuit, earphone/microphone detection circuit, and control circuit. The speech transmission circuit performs speech communication through an antenna. The data transmission circuit performs data communication through the antenna simultaneously with speech communication by the speech transmission circuit. The earphone/microphone detection circuit detects an obstacle that causes a deterioration in transmission efficiency of the antenna. When one of a data communication request during speech communication and a speech communication request during data communication is generated, the control circuit 11 determines on the basis of a detection result obtained by the detection circuit whether to permit/reject the request. A data communication control method for the portable telephone is also disclosed.

Abstract: A method for selecting a transmission protocol for a voice over packet system (VOPS) is disclosed. The method comprises generating a first data comprising a plurality of transmission protocols. The method also comprises transmitting the first data to a remote VOPS. Additionally, the method comprises receiving a second data from the remote VOPS. The second data selects at least one of the plurality of transmission protocols.

Abstract: Methods, systems and computer program products for configuring a modem communication session when conditions associated with echo noise on a digitally discontinuous channel are detected are provided. The condition is detected at the local modem by reference to the local modem's respective transmit and receive rates. A low transmit rate is an indication of a channel problem. Furthermore, where the transmit rate is also lower than the receive rate, the problem is likely at the remote modem end not just a uniformly noisy channel. This combination of conditions has been found to result, for example, from middle echo noise.

Abstract: In one aspect of the present invention, a network gateway is configured to facilitate on line and off line bi-directional communication between a number of near end data and telephony devices with far end data termination devices via a hybrid fiber coaxial network and a cable modem termination system. The described network gateway combines a QAM receiver, a transmitter, a DOCSIS MAC, a CPU, a voice and audio processor, an Ethernet MAC, and a USB controller to provide high performance and robust operation.

Abstract: A data transmission apparatus for sequentially transmitting data in units of packets each containing transmission data to the receiving end, comprises: a reception unit for receiving the transmission data as an input signal; a packet formation unit for receiving the transmission data received, and forming an uncompressed packet in which predetermined transmission data is stored as uncompressed data, and a compressed packet in which at least a portion of transmission data that follows the predetermined transmission data is compressed and stored as compressed data; a reference information management unit for holding and managing, as reference information, information relating to the uncompressed packet formed by the packet formation unit; and a transmission unit for transmitting the respective packets formed by the packet formation unit, as a transmission signal, to the receiving end.

Abstract: Disclosed is a method and apparatus for inserting a data signal into an audio signal to provide a tagged signal. The method includes removing a band of frequencies centred at a predetermined notch frequency from the audio signal, spectrally shaping the data signal so that it takes on the precise shape and magnitude of the envelope of the audio signal at the removed band, and inserting the shaped data signal into the removed band of the audio signal. The method may be used to identify an audio segment or it may be used to encode the audio signal with other desired data. The method of the invention provides for the data signal to be virtually inaudible to the listener of the audio segment yet robust enough to survive severe audio signal processing.

Abstract: A device is provided for coupling between a telephone line and the network, to exchange data between them. The invention includes software for the device, and methods of operation. The device detects when, in a fax/modem type transmission, the transmitted data encode silence or voice. This determines the end of the fax/modem type transmission, and then the device is switched to a more economical voice mode. The device analyzes the energy of the encoded signals. The device makes a discrimination of the type of signals (fax/modem, silence or voice) based on the time patterns of the energy. For fax/modem the pattern is relatively uniform, for silence it is uniformly low, and for voice the pattern oscillates.

Abstract: A communication system for transmitting voice and data information is disclosed. The communication system includes subscriber modems, system converters and at least one system data pump. The system also includes buffer memory interposed, in the upstream direction, between the converters and the data pump to allow subscriber modems to send a predetermined amount of information to the data pump on demand.

Abstract: A system that provides additional voice lines with data over a single two-wire subscriber loop. At the subscriber premise, one or more telephone instruments and data devices are connected to a digital modem into either a voice or a data interface. Analog voice signals are PCM sampled then packetized, and voice and data packets are transmitted to a CO modem over the subscriber loop. The CO modem depacketizes the voice packets into PCM format and transmits PCM voice information to a CO switch over a standard digital trunk interface. Data packets are routed to a data network. Voice packets are given transmission priority over data packets to ensures that voice is transported in the presence of data with low latency.

Abstract: In the communication system, data transmission time suitable for data transmission and the quasi-data transmission time are set in one period in accordance with the transmission path, and first data and second data are transmitted in multiplexed fashion. Bits are assigned in such a manner that the first data of one period can be transmitted during the data transmission time for one period, and also in such a manner that the second data for a predetermined period can be transmitted in the portion not assigned the first data in the data transmission time for the particular predetermined period.

Abstract: Methods and apparatus for performing call-processing functions of wide-area mobile voice calls over a wireless local-area network (WLAN) are provided. Such methods allow the extension of wide-area call-processing protocols such as Global System for Mobile Communications (GSM) or Code Division Multiple Access (CDMA) into a WLAN which uses a completely different air-interface than GSM or CDMA. Such methods enable wide-area mobile voice communications to be available in a WLAN without the use of any Voice over IP (VOIP) related technologies such as SIP or H.323.

Abstract: There is provided a server apparatus that is connectable to client apparatuses via a radio transmission medium. The server apparatus includes a storage unit storing list data indicating a priority order of the client apparatuses in allocating an exclusive period thereto, the exclusive period being a period in which exclusive use of the radio transmission medium is permitted, a control unit configured to allocate the exclusive period to the client apparatuses in accordance with the priority order indicated by the list data, and a processing unit configured to rearrange the priority order of the client apparatuses on the list data using situation data indicating a situation of one of the client apparatuses, when the situation data is transmitted from the one.

Abstract: A system for deriving multiple channels from four-wire residential telephone wiring. The invention provides two voice channels and two, or more, data channels on common residential telephone wiring. The voice channels occupy a normal telephone bandwidth, in the range of approximately zero Hz to 3500 Hz. With this channel placement, ordinary telephonic devices can use the channels, without a requirement of frequency-shifting. Above these voice-channel frequencies, data channels are provided, for internal communication within the residence.

Abstract: To stimulate a market introduction of Voice-over-Internet-Protocol (VoIP) service-orientatedly, systems with terminals (1,2) and networks (3) are provided with storage-stations (4) for storing information to be supplied in response to trigger-packets originating from sending terminals (1) via the Internet Protocol to receiving terminals (2) in the form of data-packets via the Internet Protocol during calls comprising voice-packets exchanged between said terminals (1,2) via the Internet Protocol, and as a result an additional service for VoIP has been created. Said prestored information comprises video, like pictures of users or actors or cartoon characters, and possibly audio, like sayings or exclamations, for supporting said calls.

Abstract: A system and method for communicating screen display images on a computer to another computer using a telephony network. A screen share button on a data network telephone initiates a screen shot request to a first computer associated with the data network telephone. The first computer uses a screen shot application to retrieve the image on the computer display and send it to the data network telephone. The data network telephone receives the screen shot image and sends the image to another data network telephone which is a party to a telephone conversation with the first data network telephone. The first data network telephone preferably uses the SIP invite with a session description protocol that permits the transfer of image data on SIP messages. The second data network telephone implements the image data session description protocol and communicates the screen shot image to the second computer.

Abstract: A method and apparatus for simultaneous voice and data (SVD) service in mobile communication system are disclosed. SVD service can be provided by designating a new voice RLP frame type and adding the new voice RLP frame type in the RLP frame transmittal standard for packet data in CDMA mobile communication system. Further, by designating the new SVD service option or additional signaling message exchange, the signaling procedures or multiplexing and de-multiplexing procedures used for SVD service can be simplified. Thus, mobile terminals and base stations, which could not provide SVD service, can be enabled to provide SVD service by performing SVD call processing between mobile terminals and base stations with only one service option.

Abstract: A method and apparatus are disclosed for dynamically adapting the play-out delay for voice packets transmitted over a local area network using the Ethernet standard as an access mechanism. It has been observed that the distribution of voice packet delays in a LAN (shared by voice and data traffic) follows a log-normal distribution. An adaptive algorithm is disclosed to estimate the parameters of the log-normal distribution and to apply a dynamic play-out delay to improve the quality of packetized voice conversations, quantified by minimum delay and packet loss. For example, the size, B, of the play-out buffer can be established to ensure that the packet loss does not exceed one percent (1%). The distribution parameters are continuously updated and the size of the play-out buffer, B, is modified at the beginning of every calibration interval according to the illustrative 99% percentile of the delay distribution.

Abstract: A communication system for transmitting voice and data information is disclosed. The communication system includes a system converter and a system data pump configured to transform and transmit information between a subscriber and a central location. The system converters convert information between appropriate analog and digital formats. The data pump modulates and demodulates information and is configured to operatively couple to one or more system converters.

Abstract: A method of operating a packet network for carrying voice traffic, wherein the packets carrying voice traffic include voice samples. The method identifies a replacement packet opportunity, creates a replacement packet based on a selected packet, and inserts the replacement packet in the replacement packet opportunity. The replacement packet includes samples based on samples of the selected packet, but in an order that differs from the order of the samples in the selected packet. The method may further comprise identifying another replacement packet opportunity directly following the replacement packet opportunity, creating another replacement packet based on the replacement packet, and inserting the another replacement packet directly after the replacement packet. The another replacement packet differs from the replacement packet.

Abstract: An apparatus and method to implement a highly efficient low power line driver. In a first embodiment, the invention provides a method to increase the power efficiency of a line driver. The method includes supplying a digital signal processor output to a first subtractor; supplying the first subtractor output as an input to a modulator of a line driver; subtracting the line driver output from the digital signal processor output at the first subtractor; filtering the line driver output with a low pass filter; routing the line driver output to an impedance match filter; providing a first analog-to-digital converter and a second subtractor to subtract the line impedance match filter output from the low pass filter output; providing a digital filter and a second analog-to-digital converter; and subtracting the digital filter output from the first analog-to-digital converter output at a third subtractor to output a feedback signal to the digital signal processor.

Abstract: The present invention relates to a packet data communication system and is primarily intended for packet data radio communication systems, such as a GSM-system (PLMN) supporting a GPRS-service. A data frame received from a connected second packet data network (Internet) is divided into a sequence of data segments (DS). Speech packets (SPP) with an origin from a connected telecommunication network and the segments (DS) and are transmitted over a packet data link (PL1) to a radio base station (BTS) with priority given to the speech packets (SPP). The data segments (DS) shall be transmitted on a corresponding radio link (RL2) to a designated mobile station (MS). The problem is that during high traffic loads the segments are subjected to long delays and as a result no segments can be transmitted on the corresponding radio link. Giving priority to a first few data segments of the sequence when transmitted over the packet data link (PL1) solves the problem.

Abstract: A method for a broadband modem to process incoming calls. The broadband modem is coupled to a PSTN (public switched telephone network) and a VoIP network. First, an call is incoming by the broadband modem. Then, the broadband modem detects whether the incoming call is routed from the PSTN network or the VoIP network. When the incoming call is routed from the PSTN network, the broadband modem establishes a connection between a telephone coupled to the broadband modem and the PSTN network through a switch. When the incoming call is routed from the VoIP network, the broadband modem establishes a connection between the telephone and the VoIP network through the switch.

Abstract: A host processor operating together with a network processor to detect packet media streams on a link of a packet network in accordance with a trigger. The network processor captures substantially all packets on the link in real-time to thereby capture communications transmitted via packets on the link. Each communication has at least one corresponding packet media stream. The host system communicates with the network processor to detect respective packet media streams from the captured packets in accordance with a trigger.

Abstract: A closed-loop message distribution method is disclosed for accepting documents from a variety or sources via a plurality of media, presenting the documents to a plurality of recipients through their preferred communications channel, document personalization, and suppressing delivery of follow-up documents to those recipients who have responded to an earlier related message.

Abstract: Methods and apparatus for allowing a packet data connection to be established by sending an indication of a network address through a telephony path. In a first embodiment, a protocol stack initiates the establisment of an Internet connection by sending a data segment through a public switched telephone network (PSTN) telephony path and then operates and maintains the Internet connection on separate packet connection. Dialing digits are used to indicate the address of a remote computer or wireless device via the telephony path. This invention also enables mixed PSTN/internet multimedia telephone calls. In an exemplary embodiment, when a point-to-point telephone PSTN connection is established, a screen of information automatically appears at one or both ends of the connection via the Internet.

Abstract: A two-wire field-mounted process device with multiple isolated channels includes a channel that can be an input channel or an output channel. The given input or output channel can couple to multiple sensors or actuators, respectively. The process device is wholly powered by the two-wire process control loop. The process device includes a controller adapted to measure one or more characteristics of sensors coupled to an input channel and to control actuators coupled to an output channel. The controller can be further adapted to execute a user generated control algorithm relating process input information with process output commands. The process device also includes a loop communicator that is adapted to communicate over the two-wire loop.

Abstract: A source encoder schedules its access to a network IP stack. During that scheduled period, the source encoder blocks the transport of non-real-time TCP and UDP packets. After at least one real-time packet has been transported through the stack, the source encoder allows the non-real-time packets to access the physical layer until the beginning of the next scheduled period. Consequently, the packet transport delay is reduced significantly, which results in a substantial increase in throughput.

Abstract: To improve the quality of audio signals transmitted over a computer network, a source computer (12) and a destination computer (30) that participate in the call exchange information concerning the gain and the frequency characteristics response of the audio components of each participating computer. The parameters of the audio components of the source and/or destination computer are then adjusted to produce an audio signal having one or more desired characteristics.

Abstract: A wireless adjunct for routing communications data between a switching unit and a radio transmit point over a packet based network. The wireless adjunct includes a switching unit interface, a network interface, and a processor. The switching unit interface is for communicating with the switching unit and receiving the communications data. The network interface is for communicating with the packet based network and transmitting the communications data to the radio transmit point; and the processor, in communication with the switching unit interface and the network interface, is operable to manage call processing and provide communications data from the switching unit interface to the network interface.

Abstract: A method and apparatus to manage a remote resource over a network is described.

Abstract: An isochronous output data stream format in which a single n bit word is transmitted as m multiple synchronous data streams where the length of the word transmitted in each data stream is n/m, the output data stream containing multiple isochronous different data rate data streams and one or more asynchronous data stream or streams. One bit from each of the m multiple synchronous data streams are combined and transmitted as a single signal at one of 2m analog levels. Data which is it desired to have a lower error rate than other data is transmitted in the synchronous data stream corresponding to the most significant bit associated with the 2m analog levels. Calibration data is transmitted in the synchronous data stream corresponding to the most significant bit associated with the 2m analog levels, the calibration data also being transmitted distributed over each of the m synchronous data streams.

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: A method for transmitting audio data between a computer and a GPRS card. The computer is connected to the GPRS card through corresponding connectors. The method includes after the computer receives an audio signal, transforming the audio signal into a digital signal, and packeting the digital signal into a plurality of audio units, then transmitting the plurality of audio units to the GPRS card through the connectors. Finally, the plurality of audio units are transformed to become the original digital signal, and the digital signal is transformed to become the original audio signal.

Abstract: A system for efficient sale of devices that comply with licensed standards. A preferred embodiment of the invention uses a generic, or highly adaptable, hardware device. The device can be adapted to adhere to a specific standard, e.g., code-division multiple access, time-division multiple access, etc., after manufacture such as at the point-of-sale to an end user, prior to distribution, or at some other point in a distribution and sales network. This allows manufacturers, retailers and end users to benefit from more competitive selection of standardized communication, data and other formats. Reduction of manufacturing costs and elimination of shipping, or other transfer and storage costs, is also realized.

Abstract: The invention provides a technique for power cutback of DSL transceivers (e.g., modems) using public switched telephone network (PSTN) signaling. For example, one or more dual-tone/multiple frequency (DTMF) signals generated by a touch-tone telephone, or other device, can cause a DSL transceiver to reduce the power of upstream data communications. A user of a DSL modem can cause the modem to transmit upstream data while listening to the microphone of a telephone coupled to the same twisted-pair line. Data transmission power can be adjusted by use of one or more buttons on the telephone. The DSL transceiver, in response to one or more predetermined DTMF signals, can adjust transmission power. Thus, the power cutback of the DSL transceiver can be controlled using PSTN signaling.

Abstract: A QoS management apparatus in which the quality of the requested QoS for each path and different values of the state are managed. A QoS monitor, a registered QoS, and an operation determining function are provided for each path, and the QoS is precisely managed.

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: The present invention provides a new transmission method for both voice and data packets as an enhancement of IEEE 802.11 protocol of wireless LAN network, which includes access point station (AP) and wireless stations. This method utilizes the transmission of standard beacon packets for embodying allocation information. The allocation information and transmission cycle are determined according to various criteria, among them packet's type and the traffic congestion. The transmitted allocation information includes scheduling information of reserved transmission periods for voice or data packets of both wireless and AP stations. For each wireless station voice session at least two time slots are reserved within each transmission cycle: one for uplink transmission of voice packets from the wireless station to the AP station and a second one for downlink transmission of voice packets from the AP station to the wireless station.

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 PacketCable-compliant functionality, including enhanced user privacy, compliance with CALEA, E911 and other mandated services not available in conventional distributed PacketCable 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 lightspeed 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.