Abstract: A system and method for determining DTMF coefficients stores a plurality of reference templates, at least some of said reference templates representative of likelihood ratios. A plurality of LPC coefficients is received. The LPC coefficients are representative of an input signal. The system and method determines a plurality of current likelihood ratios based, at least in part, upon the LPC coefficients. The system and method also determines an initial tone based, at least in part, on the minimum of the current likelihood ratios and the minimum of the reference likelihood ratios. A plurality of LSF coefficients is received. The LSF coefficients are determined, at least in part, upon the input signal. The system and method verifies the validity of the initial tone based, at least in part, upon the LSF coefficients.

Abstract: A network interface is presented that receives packet data from a shared medium and accomplishes the signal processing required to convert the data packet to host computer formatted data separately from receiving the data packet. The network interface receives the data packet, converts the analog signal to a digitized signal, and stores the resulting sample packet in a storage queue. An off-line processor, which may be the host computer itself, performs the signal processing required to interpret the sample packet. In transmission, the off-line process converts host-formatted data to a digitized version of a transmission data packet and stores that in a transmission queue. A transmitter converts the transmission data packet format and transmits the data to the shared medium.

Abstract: Transcoder apparatus is designed to transcode a receive signal (UR) into a transmit signal (UE). For this purpose, it includes a signaling transmitting member (14) for producing a signaling message and for introducing it into the transmit signal by stealing bits from said signal. According to the invention, the signaling transmitter member (14) also comprises a tone generator (15) to add a tone signal (UTS) to the transmit signal (UE) prior to applying the signaling message to said transmit signal.

Abstract: A multi-carrier transceiver, such as a DMT-based ADSL transceiver, has separate processing paths for different information types, such as voice, data, video, etc. Each processing path can include separate scrambling, interleaving, bit loading or tone ordering, and constellation encoding. In this manner, specific bit loading or tone ordering can be performed for each of the different information types, such that transmission carriers are selected for optimal bit rates and error rates.

Abstract: A signal detector includes a transform unit and a post processing test unit. The transform unit is adapted to a transform unit adapted to receive at least first and second frames of input samples, determine a first plurality of energy values for the first frame based on a first set of transform coefficients, and determine a second plurality of energy values for the second frame based on a second set of transform coefficients. The post processing test unit is adapted to identify at least one target frequency based on the first plurality of energy values. The second set of transform coefficients is based on the target frequency, and the post processing unit is adapted to determine the presence of a first signal in the input samples based on the second plurality of energy values. A method for detecting a signal is provided. A first frame of input samples is received. A first set of transform coefficients is determined based on a first plurality of frequencies.

Abstract: A system and method is provided for collecting and transporting DTMF digits between two network end points in a packet network in order to preserve the integrity of the DTMF digits. The system and method use a Session Initiation Protocol (SIP), which is currently defined by the standard body of Internet Engineering Task Force. SIP applications utilize certain predetermined messages, such as an INFO message, and templates. DTMF digits can be collected and transported by using the SIP INFO messages. This works especially well for telephony applications which may require mid session signaling.

Abstract: The present invention is directed to a system and method for determining when a particular tone in a discrete multi-tone communications system is being used to transmit a control signal. Broadly, the system and method of the present invention are realized by a digital signal processor that is configured to detect a constant phase shift on individual tones when a control signal condition is present.

Abstract: A periodic signal detector includes sample counting means for counting a number of samples between adjacent zero cross points in an input signal, the sample counting means outputting the number of samples between each time the number of samples between the adjacent zero cross points is obtained and continuing outputting the number of samples until a next zero cross point is detected, and periodic signal section detecting means for detecting a section, as a periodic signal section, in which the output from the sample counting means has a constant result.

Abstract: In optical two-way communication using a single optical fiber, part of the light transmitted from an apparatus is received as stray light by the apparatus itself. This occasionally hampers accurate detection of disconnection from the partner apparatus and of the start of data communication by the partner apparatus after the establishment of connection. To overcome this, the timing with which an apparatus transmits a tone signal after the establishment of connection is compared with the timing with which the apparatus receives a tone signal from the partner apparatus, and the two apparatuses use different sets of data transmission properties (for example, different wavelengths) to accurately detect disconnection. Moreover, based on the properties of the received light and the characteristics of the receiving amplifier, repetition of the same code for longer than a predetermined period is detected to accurately detect disconnection.

Abstract: A tone detector able to detect multicomponent tones for various tone formats is disclosed. An input signal is received and frequency and amplitude components, if any, of each sample of the incoming signal are estimated. In estimating the components, a current digit value is used to appropriately filter the input signal and estimate frequency and amplitude components. Using the component estimates, a new digit value is determined using a table in storage corresponding to the current tone format. The new digit value corresponds to the symbol whose components are closest in value to the estimated components corresponding to the current digit value. The new digit value is fed back as the current digit value and used in combination with a next sample of the input signal to determine a new digit value. A decision unit is used to process the digit values to determine when a valid symbol is detected.

Abstract: A method and apparatus is provided for locking a local oscillator to a reference oscillator. The reference oscillator is arranged to transmit a signal containing two reference pilot tones in a signal sideband. The apparatus samples the received signal and makes a frequency shifted copy where the shift is equal to the expected separation between the reference tones. The non-shifted and shifted signals are then combined in the frequency domain to constructively reinforce one of the reference pilot tones which is then searched for. Once the tone has been identified, a correction to the local oscillator frequency is calculated. The apparatus may be of particular use in a radio telemetry system in which it is important that the frequency of the local oscillators in the transmitters and receivers are very accurately matched to each other and where, in order to reduce interference, all the clocks used in the system are locked to a master clock.

Abstract: A method for detecting a tone signal includes determining a correlative value of an input signal obtained for each frame of a predetermined length, determining an estimating expression that represents an output power of a second order FIR band elimination filter that uses the correlative value of the input signal as an input and a frequency thereof as a parameter, determining a frequency at which a minimum value of the estimating expression is obtained, determining a prediction gain of the FIR band elimination filter at the determined frequency, and determining that the input signal is a tone signal if a value of the determined prediction gain exceeds a preset threshold value.

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: A noise calculation mechanism is incorporated into a cadence tone detector which upon a miss in either a energy or silence interval allows for retention of those periods corresponding to noise and augments the appropriate energy and silence interval.

Abstract: The present invention discloses a system and a method for transmitting DTMF signals over high speed digital networks using voice compression algorithms, and particularly to a system for ensuring the integrity of DTMF signals at the destination node of a network after compression and decompression of data on a voice connection. A DTMF Detector is placed in parallel with a voice compression unit performing the compression algorithm. When a candidate DTMF signal is detected, the signal component at the higher frequency in the candidate is filtered out to eliminate the possibility of double DTMF detection at end user equipment. When the candidate DTMF signal is finally validated by the source node as representing a true DTMF signal, only the data essential for reconstituting the DTMF signal are transferred to the destination node.

Abstract: A tone processing method and communications system which reduce the probability of falsely detecting modem and facsimile communications that occur on a communications link. The link connects a calling endpoint to an answering endpoint. In a preferred embodiment, when the answering endpoint goes off-hook in response to receiving a call from the calling endpoint, an elapse of time is measured while monitoring the link. If a predetermined amount of time elapses without detecting at least one tone on the link, tone detection criteria is modified to reduce tone detection sensitivity, or to ignore tones altogether. In another embodiment, the tone detection criteria is similarly modified if voice activity is detected on the link. In yet another embodiment, when tones and voice activity occur at different times during a call, at least one characteristic is modified to optimize the link for the type of communications occurring on the link.

Abstract: A tone detection device is provided which includes a Fourier Transform means for performing a Fourier Transform on an incoming signal. The Fourier Transform means generates a frequency spectrum of the incoming signal. A normalizing means receives the generated frequency spectrum and normalizes the spectrum for magnitude. The normalizing means then generates a normalized frequency spectrum. An integrator means receives the normalized frequency spectrum and generates a mean of the normalized frequency spectrum over time. A comparator means then determines whether the mean of the normalized frequency spectrum exceeds a predetermined threshold value. If so, a signal is generated indicating that a tone is detected. If the mean of the normalized frequency spectrum does not exceed the predetermined threshold value, a signal is generated indicating that no tone is detected.

Abstract: A signal detector includes a transform unit, a gain control unit, and an energy level threshold unit. The transform unit is adapted to receive a plurality of input samples and calculate a first plurality of delay values based on the input samples. Each delay value is based on at least one previous delay value and a current input sample. The transform unit is adapted to calculate a first energy value based on at least a subset of the first plurality of delay values. The gain control unit is adapted to determine if a selected delay value of the first plurality of delay values exceeds a predetermined threshold and scale the selected delay value, the at least one previous delay value used to calculate the selected delay value, and subsequent input samples received after the current input sample by a first amount in response to the selected delay value exceeding the predetermined threshold.

Abstract: An apparatus for determining energy threshold detection values for detecting at least one tone having a known frequency and duration in an input signal. The input signal is input over a period of time that is divided into frame portions including at least an initial frame portion and a last frame portion. An energy value is determined for the at least one tone during each frame portion. A set of control flag signals are set for the at least two frame portions based on the energy values being below a threshold value in previous frame portions and the presence of noise in previous frame portions. An offset into a data table is determined based on the control word, the offset being used to retrieve a scaling coefficient and an address of an energy value for the at least two frame portions from the data table.

Abstract: Two related voiceband compression techniques are employed in order to enable an RF telecommunications subscriber unit to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the subscriber unit to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Time slot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: The present invention is directed to a voice tone discriminator for distinguishing between call progress tones and voice. The voice tone discriminator is useful in various applications involving the ability to automatically charge a telephone user based on the exact time the user starts speaking. According to another aspect of the present invention, a prediction algorithm may be implemented for distinguishing voice and tone based on the fact that tones are more accurately modeled with a linear filter than voice signals. Thus, a low order filter or predictor may accurately model redundancies in tones but not in voice. A normalized error between an original and a predicted signal may be used to distinguish voice from tones. Voice may be detected when the error is above a preset threshold for a time greater than a preset fixed duration.

Abstract: Modems, methods, and computer program products for recovering from errors during a tone reversal sequence between a modem pair detect a false tone reversal during a tone reversal sequence at a first modem. If a false tone reversal is detected, then the first modem requests that a second modem restart the tone reversal sequence. Tone reversal sequences are used, for example, in the startup protocols of both the V.34 and V.90 standards to calculate the round trip delay between two modems. Neither standard provides for any recovery procedure, however, should the tone reversal sequence fail. The modems, methods, and computer program products can allow errors in a tone reversal sequence to be addressed immediately rather than proceeding to later phases in the communication protocol where the errors may ultimately cause the communication session to breakdown.

Abstract: A radio communication device provides extended burst tone detection for a demodulated I and Q input signal. The device includes a first burst detector coupled with the input signal and provides a first detection signal when a FCB tone is detected. A frequency shifter is coupled with the input signal and frequency translates the input signal by a predetermined amount. A second burst detector is coupled with the translated input signal and provides a second detection signal when a FCB tone is detected. A combiner is coupled with the first and second detection signals and indicates FCB tone detection when either of the first and second detection signals indicate FCB tone detection.

Abstract: A method and apparatus for automated sectionalization of a DS1/DS3 data path based upon information received at a single location along the path. A test and monitor device is located at a point of demarcation between an LEC and an IEC. A Remote Module is located at a point of demarcation between the LEC and CPE. The test and monitor device is fully ANSI compatible. The information that is received is processed in a three step process in order to generate a “Sectionalizer Report”. In preparing the Sectionalizer Report, the information output from a filter is used to determine where particular Events originated.

Abstract: A method of transmitting DTMF tones over an Internet telephone call includes the step of monitoring an audio signal in a transmitting IP Gateway in order to determine when tone is suspected, and then deleting the audio signal including such tone if tone is confirmed. After the audio signal including the tone is deleted, a digital out of band signal is transmitted including the appropriate tone information.

Abstract: A method of facilitating telephony communications over a packet-switched network includes detecting signaling tones in an audio stream and blocking them from the stream in response to the detecting. After the blocking, the audio stream is sent over an undependable channel in the packet-switched network. The detected signaling tones can be sent over a dependable channel instead of the undependable audio channel.

Abstract: A method identifies a pulse sequence having known values and a known length in a signal. According to this method, the mathematical sign of the phase difference between samples of the signal is used to estimate whether the transmitted pulse is a 1 or a 0. Undersampling, carried out to a selectable extent, produces a relatively insensitive response to adjacent channel interference. The sum of the pulses in a window which is proportional to the length of the pulse sequence and to the extent of the undersampling is determined, with the pulse sequence being regarded as being identified at the point in time at which the sum of the pulses in this search window exceeds a threshold value.

Abstract: Two related voiceband compression techniques are employed in order to enable an RF telecommunications system to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the telecommunications system to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Timeslot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: A bit rate converter capable of avoiding slip of data in a memory for use in bit rate adjustment is disclosed. A phase comparator compares a write address and a read address of the memory to produce a phase difference. The write address is ahead of the read address in the memory. A stuffing rate controller selects one of a plurality of preset fixed stuffing rates depending on the phase difference. A stuff pulse inserter inserts a stuff pulse into readout data from the memory at the selected stuffing rate. When normally operating, a normal fixed stuffing rate is selected. When the phase difference is small than a lowest threshold value, a higher fixed stuffing rate is selected. When the phase difference is greater than a highest threshold value, a lower fixed stuffing rate is selected.

Abstract: A system and method to detect a call trigger during a call without requiring a service platform to remain connected to the call. A first communication device transmits a call having call signaling and user communications. A signaling processor receives the call signaling from the first communication device and selects a connection to a second communication device. An interworking unit receives the user communications from the first communication device and converts the user communications from a time division multiplex format to asynchronous transfer mode cells that identify a selected connection to the second communication device and transports the user communications to the second communication device. The interworking unit detects a call trigger in the user communications. The interworking unit transmits call trigger data to the signaling processor.

Abstract: A method of transmitting DTMF signals over a compressed computer network (24) comprises sending a packet containing a compressed DTMF signal followed by a confirmation packet. The confirmation packet preferably comprises an uncompressed DTMF signal which is transmitted on the same logical channel as the packets that represent voice signals. To detect the DTMF signal transmitted, a telephony Internet server (28) decompresses packets as they are received. If a packet appears to contain a compressed DTMF signal, the telephony Internet server looks for the presence of the confirmation packet before processing the previously received packet as a DTMF signal. In an alternative embodiment, the confirmation packet comprises a code that represents a DTMF signal. The packets that represent the voice signal and the code representing the DTMF signal are sent on separate virtual channels of the same logical channel.

Abstract: In order to support both signalling processing on a per channel basis and multiple adaptation protocols, an interface (10) is modularised principally by function. Incident channels (60-62, Cho—Chn) are applied to a routing device (16) that consults a connection map (64) to determine an appropriate path, via an interface (90), for signal processing of each channel, as shown in FIG. 2. Modularised processing platforms (70-74) each contain a number of signal processors (80-86), with each processing platform providing an additional level of indirection with respect to channel handling. In this latter respect, one of the signal processors (86) on each platform is assigned to distribute channel processing to appropriately configured other ones of the signal processors and such that signal processing functions may be distributed between the other ones (80-84) of the signal processors.

Abstract: A network interface is presented that receives packet data from a shared medium and accomplishes the signal processing required to convert the data packet to host computer formatted data separately from receiving the data packet. The network interface receives the data packet, converts the analog signal to a digitized signal, and stores the resulting sample packet in a storage queue. An off-line processor, which may be the host computer itself, performs the signal processing required to interpret the sample packet. In transmission, the off-line process converts host-formatted data to a digitized version of a transmission data packet and stores that in a transmission queue. A transmitter converts the transmission data packet format and transmits the data to the shared medium.

Abstract: A method and system in a network telephony system for transporting audio signals with lower delay when DTMF is not present. The method of transporting audio signals determines whether DTMF signals are included in the audio signal, or are likely to be transported according to the called or calling parties. A delay is imposed in the audio when DTMF signals are sensed or determined to be present as indicated by the calling party, the destination telephone number, the called party, and various other indicators. A timer may be used to trigger a reduction and elimination in the audio delay after an initial period in the call for calls in which DTMF signaling is used for an initial period of the call. The timer may also trigger the reduction and elimination in the audio delay when no DTMF signals have been received after a period of time. The system uses a DTMF detector to sense DTMF signals in the audio signal and to signal the delay generator to impose a delay in the audio stream.

Abstract: An apparatus and method for selectively compressing information to be transmitted over a communications network includes an analog line monitor connected to an analog telephone and an analog data communications device. The analog line monitor is configured for detection of carrier tone signals associated with data transmissions. An analog-to-digital/digital-to-analog (AD/DA) converter is connected to the analog line monitor to convert the information contained in transmissions from the analog telephone and the analog data communications device from an analog format to a digital format. Optionally, the AD/DA circuitry is configured for demodulating data prior to transmission over the network. Compression circuitry is responsive to detection of a carrier tone signal by the analog line monitor, such that detection of a carrier tone signal suppresses compression of data transmitted by the data communications device.

Abstract: A DTMF transmission structure of a W-CDMA WLL system in which a DTMF signal is transmitted and received to and from a switching system through a terminal station and a base station controller, each of the terminal station and the base station controller comprising: an echo canceller for extracting an echo component from a received signal; a DTMF detector for discriminating whether a signal to be transmitted is a DTMF signal and controlling the operation of the echo canceller; an adder for adding the signal to be transmitted and the output of the echo canceller; a VOCODER for coding the output signal of the adder to a bit stream form or decoding the received signal in the bit stream form to an analog signal; an adjusted DTMF detector for detecting a digit component from the output of the VOCODER; a DTMF generator for reproducing a normal DTMF signal according to the detected digit component; and a silence generator for generating a silence for a predetermined time immediately after the DTMF signal is reproduce

Abstract: A method of facilitating telephony communications over a packet-switched network includes detecting signaling tones in an audio stream and blocking them from the stream in response to the detecting. After the blocking, the audio stream is sent over an undependable channel in the packet-switched network. The detected signaling tones can be sent over a dependable channel instead of the undependable audio channel.

Abstract: Whenever a subscriber associated with a plurality of subscriber features receives or originates a call, an application module associated with the serving local exchange links into the established call. The application module thereinafter intercepts all incoming calls towards that subscriber and transmits control data over the existing call connection to the subscriber terminal informing it of the new incoming call. The control data are transmitted over the existing call connection using out-of-band voice frequencies or during silent portions of the speech connection. Over the same existing call connection, the application module receives an instruction from the subscriber terminal for providing a particular terminating call treatment for this incoming call. The application module then provides the instructed terminating call treatment as instructed by the subscriber.

Abstract: In system having a plurality of telephones coupled to a communications controller which is connected to an integrated services digital network (ISDN) line of a telecommunications network, the ISDN line having two bearer channels of predetermined bandwidth and a data channel and the plurality of telephones being represented by a plurality of identifiers, a method for convenient incoming call acceptance comprises the communications controller associating each identifier for the plurality of telephones with a distinctive ring pattern. The controller receives, on the data channel, a dialed identifier of the plurality of identiers which was dialed in connection with an incoming call and rings all idle telephones of the plurality of telephones according to the distinctive ring pattern associated with the dialed identifier. The controller connects a first telephone of the idle telephones which goes off-hook, through a voice subchannel extracted from the bandwidth of the two bearer channels, to the incoming call.

Abstract: Two related voiceband compression techniques are employed in order to enable an RF telecommunications system to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the telecommunications system to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Timeslot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: A method for reducing probability of clipping in an IFFT-based (inverse fast Fourier transform based) multitone communication transmitter, the method including applying an IFFT to an original sequence thereby to generate an IFFT output sequence, if the IFFT output sequence fits a predetermined clipping profile, applying at least one transformation to the IFFT output sequence, thereby to define a transmitted sequence, the transformation-applying step including decomposing the IFFT output sequence into a first component sequence and a second component sequence, rearranging at least one of the first and second component sequences with respect to the other, thereby to define third and fourth component sequences, and combining the third and fourth component sequences into a final output sequence.

Abstract: A signal detector includes a transform unit and a signal alignment unit. The transform unit is adapted to receive a plurality of frames of time domain input samples and generate a plurality of energy values based on the frames of time domain input samples. Each energy value is associated with one of the plurality of frames. The signal alignment unit is adapted to identify at least a first partial energy frame and a second partial energy frame from the plurality of frames. The first partial energy frame is associated with a first frequency and has a first energy value. The second partial energy frame is associated with a second frequency and has a second energy value. The signal alignment unit determines a misalignment between the first and second partial energy frames based on the first and second energy values. A method for detecting a signal is provided. The method includes generating a plurality of energy values based on the frames of time domain input samples.

Abstract: A system for providing an IR-link between a mobile phone and its accessories to transmit audio, control, and data signals between them in an apparent full-duplex real time mode using a conventional half-duplex IR-transceiver with suitable FIFO registers and control means. The "direction" of the half-duplex channel is "switched" by the system which may be embodied in an IR-link module, having transmission channels for digital full-duplex audio signals, slow data messages and fast data messages, and for some control data. The module is connected to the baseband of a phone through a baseband interface (BBIF) and communicates through a media interface (MIF) and an infrared interface (IRIF) with free air space or a lightguide. A link-specific low-level protocol is used for audio and slow data message transmission but the link hardware of the module may also support other protocol implementations. The system may be implemented in hardware and/or software.

Abstract: A transmultiplexer is connected between digital facilities and analog facilities for translating information between TDM and FDM formats on the digital and analog facilities, respectively. Information in the TDM format is representative of a plurality of digital telephone signal channels and information in the FDM format is representative of a plurality of analog telephone signal channels. The total capacity of the digital facilities is smaller than the total capacity of the analog facilities. Any one of the digital channels can be selectively coupled to any one of the analog channels. The transmultiplexer is installed at a telephone central office facility as an interface between a cable telephony transmission system and a digital telephone switch. A host computer controls both the transmultiplexer and the digital telephone switch.

Abstract: Improved telecommunication apparatus is realized with a structure that is tailored to interact with the telecommunication network strictly in packet format. The apparatus includes circuitry for creating either control or information packets, where each packet comprises a header portion and a payload portion. The apparatus also includes an identifier module that enables the apparatus to identify itself to the telecommunication network with an identifier signal that is unique to it. The apparatus further includes encoding and decoding circuitry to create, and decode, highly compressed digital representations of audio signals and, optionally, encryption and decryption features to enhance security of communication. Still further, the apparatus includes circuitry to enable users to interact with the network and partake of telephony service.

Abstract: A joint signal detection and channel parameter estimation scheme is provided for multiple subcarrier signaling with pilot symbol-assisted modulation schemes. Known symbols located in the subcarriers are used to estimate a pair of parameters associated with the generation process of the fading frequency selectivity common to all of the subcarriers. These known symbol estimates are used to derive parameter pair estimates for unknown symbols located in the subcarriers. This parameter estimation thus effectively extracts information regarding the fading frequency selectivity through the pilot symbols received not only by the subcarrier of interest but by other subcarriers as well. The fading complex envelope with each subcarrier is then derived from the estimates of the parameter pair.

Abstract: A handshake between sending and receiving devices (10) that exchange DTMF signals across a network (14) is accomplished by having the receiving device send a DTMF ready tone (digit) for a prescribed interval following receipt of a call set-up message received after the sending device has initiated a call. Upon receipt of the DTMF ready digit, the sending device sends to the receiving device a data message includes a string of digits of interest prefixed by M header digits. After receiving at least a portion of the data message, the receiving device stops transmitting the ready digit. Once the receiving device has received the entire data message, the receiving device strips the protocol digits (the M header digits and any of the ready digits received due to echo). By analyzing the stripped protocol digits, the quality of the handshake between the receiving and sending devices can be determined.

Abstract: End-to-end call processing of an ISDN-BRI central-office switch is simulated using a polled multi-tasking method which is executed by a single microcomputer in an ISDN-BRI central-office emulator which includes (i) a packet handler, (ii) a time-space multiplexer, (iii) an in-band tone generator, and (iv) a plurality of dual-interface line modules, each of which includes a U-interface loop termination device and an S/T-interface network termination device. The method includes controlling the packet handler to simulate packet-handling functions required for ISDN signalling and ISDN data exchange, controlling the time-space multiplexer and in-band-tone generator to simulate at least one of the circuit mode voice and data connections functions of the ISDN-BRI central-office switch, and activating either the U-interface loop termination device or the S/T-interface network termination device of at least one of the dual-interface line modules.

Abstract: A first validation circuit utilizes residual energy unaccounted for by a linear predictive coding model for a specified order of model as a criteria on which to make a validity determination of whether a valid tone or digit has been received. A second validation circuit operating in conjunction with the first validation circuit utilizes discrete Fourier transforms at the center and adjacent frequencies to energy peaks associated with tones. A ratio of the peak energy to the adjacent energies is utilized in combination with corresponding thresholds to make a validity determination. The combination of the first and second validation circuit provides a high level of confidence that a particular tone or digit has been detected.

Abstract: An ADSL receiver (200) receives an upstream modified ADSL signal and an ISDN signal from a remote terminal (32) on a twisted-pair copper wire (18). An ADSL transmitter (100) of the remote terminal (32) transmits the ADSL signal in a frequency range above an ISDN frequency range so that the ADSL signal does not overlap the frequency range of the ISDN signal. In one embodiment, the ADSL receiver (200) includes a band pass filter (201), an analog-to-digital converter (203), a decimator (205), a fast Fourier transform (210), and a digital signal processor (212). The decimator (205) converts the ADSL signal back to base band, thus allowing an ADSL signal source to simultaneously utilize the telephone line with an ISDN signal source, without significantly reducing ADSL throughput.