Abstract: A time-slot interchanger (TSI 100) is used to control both time-slot access and distribution of signals to physical or virtual signal-processing circuits (106-116). The TSI includes a control table (200) with an entry (202) for each time slot, that specifies the source (204, 205) and destination (230, 231) of signals received during the corresponding time-slot interval and the processing (208-228) of those signals. The TSI is coupled to the signal processing circuits, and sends the received signals to the signal processing circuits for processing along with control signals to effect the processing specified by the corresponding control-table entries. For conferencing, the TSI sends all signals for each conference in sequence to the conferencing circuit (108) before starting and sending any signals for any other conference, thereby allowing a single conference accumulator to support any number of conferences of any size.

Abstract: A receiver manages multiple speakers in a packet network telephone call. A packet gateway receives audio packets from the multiple speakers over the packet network. Memory in each receiver stores the audio packets and information about the multiple speakers in the telephone call. A processor selects which audio packets and what speaker information to retain in memory. The processor determines which of the selected audiopackets to store in memory and mix together to produce an audio output signal by determining from the speaker information which of the multiple speakers are actively talking and which speaker has not actively talked for the longest period of time.

Abstract: A system and method for multi-media half-duplex conferencing in a computer network comprising a plurality of stations and a control server.

Abstract: Endpoints, such as clients and gateways, maintain local codec priority lists (500), receive remote codes priority lists (504), and maintain common communication attribute priority lists (502). When opening a transmittal channel, a local endpoint determines the highest priority codec on its local priority list (500) that is also on the remote priority list (504). The local endpoint also determines the highest priority codec on the remote priority list (504) that is also on the local priority list (500). If the codes types match, then the transmitting channel is opened with that codec. If there is no match, then the common communication attribute priority list (502) is used, and the local or remote codec that has a corresponding highest or lowest desired attribute is used for the communication.

Abstract: A bidirectional digital signal transmission system includes a transmission line, and video instruments respectively connected to both ends of the transmission line. In each of the video instruments, a digitized signal is multiplexed on a time division basis and compressed in a time base, thereby generating a transmission signal constituted by repetition of a signal period and a quiescent period. The transmission signal generated by one of the video instruments is transmitted during the quiescent period of the transmission signal generated by another one of the video instruments, thereby realizing bidirectional digital signal transmission. The system further includes a repeater circuit inserted in the transmission line at an intermediate location thereof to compensate for deterioration of the signal occurring during transmission over the transmission line. The repeater circuit includes a switch device for preventing a transmission signal feedback loop from being formed in the repeater circuit.

Abstract: A method for controlling subscription services delivered to a user by an Internet Service Provider (ISP) coupled to an Automated Intelligent Network (AIN) telephone system with at least one central office switching system. An intelligent peripheral subsystem is connected to the central office switching system, via a call connection channel. The intelligent peripheral subsystem providing at least one auxiliary call processing capability via the call connection channel and provides a telephony speaker authentication method for selectively authorizing updates to user subscription services types provided by the ISP, wherein each of the subscription services types include one or more service options.

Abstract: Systems and methods by which voice/data communications may occur in multiple modes/protocols are disclosed. In particular, systems and methods are provided for multiple native mode/protocol voice and data transmissions and receptions with a computing system having a multi-bus structure, including, for example, a TDM bus and a packet bus, and multi-protocol framing engines. Such systems preferably include subsystem functions such as PBX, voice mail and other telephony functions, LAN hub and data router. In preferred embodiments, a TDM bus and a packet bus are intelligently bridged and managed, thereby enabling such multiple mode/protocol voice and data transmissions to be intelligently managed and controlled with a single, integrated system. A computer or other processor includes a local area network controller, which provides routing and hub(s) for one or more packet networks. The computer also is coupled to a buffer/framer, which serves to frame/deframe data to/from the computer from TDM bus.

Abstract: A system and method for establishing a communication path for a call between a video telephone/teleconference call and a packet network telephone terminal through a packet network. The call is routed through a multimedia gateway which performs a conversion process between a video telephone/teleconference domain and a packet network telephone domain.

Abstract: A PC-based home automation system uses a low data-rate transport layer and COM-based software components for control of devices in a home automation network. The home automation system is merged with a messaging-based HAVi-network that uses IEEE 1394 as a high data-rate transport layer. The HAVi-network controls audio/video equipment in a home entertainment system. The home automation services and devices are registered as a HAVi-compliant elements with the HAVi network's FAV or IAV device. The home automation resources (devices and services) have both COM OLE Automation Interfaces and HAVI-compliant interfaces to permit control of the home automation system from the HAVi-network.

Abstract: The present invention provides a multiparty network communication system, which is constructed by connecting system users in serial. Various types of information are packed and encoded according to protocol into information packets. When information packets are transmitted within the network, the receiving party distinguishes them according to the coding to be voice or data information and forwards them to various modules at user's end for further processing. Voice information is processed by virtue of the superimposing technology that mixes the receiving information with the voice information received at user's end and forwards to other adjoining users. A multiparty voice communication is then realized. Moreover, the present invention provides methods of manipulating user's system failure so in case one user system among conference participants fails, his neighboring users will be automatically connected to allow other participants to continue the multiparty network conferencing.

Abstract: A communications task resides on an audio/communications board of the computer conferencing system. A communications manager and a communications application programming interface reside on a general-purpose host-processor of the computer conferencing system. The communications manager receives local compressed video signals from a video subsystem of the computer conferencing system and passes the local compressed video signals to the communications task. The communications task receives local compressed audio signals from an audio task of the computer conferencing system, the audio task residing on the audio/communications board. The communications task transmits the local compressed video signals and the local compressed audio signals over a communications link to a remote computer conferencing system. The communications task receives remote compressed video signals and remote compressed audio signals over the communications link from the remote computer conferencing system.

Abstract: The present invention is embodied in a digital communication system where multiple media data sources are time multiplexed into a packetized data stream, each packet having an assigned priority and the packetized data stream transmitted in substantially the order of assigned priority. At both the transmit side, and the receive side, audio packets are given priority processing over video packets, which in turn have priority over text/graphics data packets. Continuous real time audio playback is maintained at the receiver by delaying the playback of received audio in a first in/first out (FIFO) buffer providing a delay at least equal to the predicted average packet delay for the communication system. Optionally, the average system delay is continuously monitored, and the audio playback delay time is adjusted accordingly. Audio playback is slowed or accelerated in order to shrink or grow the difference in time between the sender and receiver.

Abstract: A file is logically partitioned into data channels where each data channels holds a sequence of data of a particular data type. The data channels are logically partitioned into delivery times. The format of the file explicitly sets forth the synchronization between the data channels and the delivery times of data held within the channels. The file format is especially well adapted for use in a distributed environment in which the file is to be transferred from a server to a client. Channel handlers are provided at the client to process respective data channels in the file. The channel handlers are data type specific in that they are constructed to process data of an associated data type. The data in the file may be rendered independently of the delivery time of the data.

Abstract: A distributed digital switching system is provided that includes a plurality of service controllers (20) which are interconnected to each other through a network (10). A network arbitrator is provided that controls the flow of traffic on the network (10). Each of the service controllers (20) is operable to provide a switching configuration between inbound data to the service controller (20) and network terminations associated with each of the service controllers (20). A processor (150) on each of the service controllers (20) controls the operations thereof with the voice information stored in a memory (208) which can be configured with a circuit (204). Each of the service controllers (20) has associated therewith a conference circuit for defining and creating a conference that can access all inbound data and provide interconnections therebetween and output this on a single outbound conference channel. There can be multiple conferences created at each service controller (20).

Abstract: A time division multiplexing ("TDM") digital matrix intercom system for connecting intercom stations. The system includes a digital switching system receiving audio input communications signals from at least some of the intercom stations, and sending audio input communications signals to at least some of the intercom stations. The digital switching system includes a TDM system for broadcasting on a TDM bus multiple digital communications signals including such digital audio input signals. The TDM system includes one or more interfaces connecting intercom stations with the TDM bus.

Abstract: A signal transmitting-receiving system which has a signal transmitting side system constructed by signal transmitter T1, voice processor T2 and signal receiver T3 and also has a signal receiving side system constructed by signal receiver R1, synthesis circuit R4, signal transmitter R5 and delay circuit R3. The signal transmitting-receiving system is constructed such that no processing for making a video signal coincide with a voice used in communication using voices in position on a time axis is performed.

Abstract: A method and system for integrating occasional user-initiated data transfers with a generally continuous video transmission in a multimedia communications session includes enabling a party of the session to select at least two levels of video quality of service for the session. A first level is selected for time segments in which the user-initiated data transfers are active. A second level of video QoS is selected for time segments having user-initiated data transfers. Connectivity for the session is established based upon the first level. However, upon initiation of a data transfer, the video transmission requirements are switched to requirements based upon the second level of video quality of service. Present-time quality of service is monitored to determine whether the data transfer should be regulated so as to ensure that the present-time video quality of service remains at or above the second selected level.

Abstract: The present invention is embodied in a digital communication system where multiple media data sources are time multiplexed into a packetized data stream, each packet having an assigned priority and the packetized data stream transmitted in substantially the order of assigned priority. At both the transmit side, and the receive side, audio packets are given priority processing over video packets, which in turn have priority over text/graphics data packets. Continuous real time audio playback is maintained at the receiver by delaying the playback of received audio in a first in/first out (FIFO) buffer providing a delay at least equal to the predicted average packet delay for the communication system. Optionally, the average system delay is continuously monitored, and the audio playback delay time is adjusted accordingly. Audio playback is slowed or accelerated in order to shrink or grow the difference in time between the sender and receiver.

Abstract: A method for determining a talk/listen state using voice detection includes receiving an audio sample and detecting whether the audio sample includes voiced sound. The audio sample represents sound measured during a sample time interval. The method further includes deriving an audio level from the audio sample and comparing the audio level to a threshold level. The audio level represents an average power level of the audio sample. The method further includes determining the talk/listen state depending on a relation of the audio level to the threshold level and depending on whether the audio sample includes voiced sound.

Abstract: A multipoint teleconference system dispensing with an MCU, in which terminals are connected into a loop configuration through an ISDN in such a way that each terminal is coupled with each of the adjacent terminals by a single B channel. Thus, a communication frame is revolved through a looped communication channel. Each terminal enters control information for controlling a multipoint teleconference, into a predetermined area within the communication frame, and it relays the resulting communication frame to the next terminal.

Abstract: An apparatus for embodying integrated time switching and conference calling functions in a private exchange system, comprising a first converter for converting serial mode channel data from a plurality of incoming highways into parallel mode channel data, and for multiplexing the parallel mode channel data in time division to form multiplexed data in time slots within a frame. A data memory having a plurality of memory locations is connected to the first converter to store the multiplexed data in groups of said time slots received from the first converter during a write operation, and to supply stored multiplexed data in groups of the time slots during a read operation. An address counter generates a write address during the write operation for storing the multiplexed data in groups of the time slots in the data memory. A control memory stores control information to read and select stored multiplexed data during the read operation.

Abstract: In a communication system having a plurality of communication terminals connected by a communication circuit in the form of a loop or a bus, data circulated through the communication circuit has a multi-frame structure having a multiplicity of frames in which data fields are assigned to the terminals, and the circulated data is relayed and updated. If all data generated by the terminals is circulated through the communication circuit, each terminal may receive data generated from itself, which is essentially meaningless for the terminal. Data which has become essentially meaningless is deleted during communication to prevent such meaningless data from always occupying a part of the communication circuit to reduce the communication speed and the efficiency of utilization of the communication circuit.

Abstract: A system that enables a portable station of the personal handy phone system to support a conference call. The present invention provides this capability by empowering a portable station to communicate with up to four cell stations simultaneously enabling the user of the portable station to communicate with the users of four other telecommunication devices. Specifically, the present invention modifies the internal circuitry of a portable station to increase the number of receiving channel circuits to four and the transmitting channel circuits to four while adding a digital signal processor to mix the different received and outgoing voice signals. The increased number of receiving and transmitting channel circuits enables a portable station to utilize all the communication slots located within the frames of the radio communication interface of the personal handy phone system. Communication is time-multiplexed across these slots.

Abstract: The present invention provides a video and audio signal multiplex transmitting apparatus having a video and audio signal multiplex sending apparatus for grouping audio data to packets of a fixed length in the sending side, and multiplexing video and audio signals by multiplexing horizontal and vertical phase signal outputs from horizontal and vertical counters that are in synchronization with a video signal to an audio data as video phase information, and a video and audio signal multiplex receiving apparatus provided with an audio packet interpolating device for separating the signal received to video and audio signals, detecting, if any audio data is lost, a lost audio data from a data block number of the audio data and audio clock phase information, and recovering an audio clock by means of horizontal and vertical counters and the clock phase information transmitted.

Abstract: A multipoint video conferencing system in which the video conferencing terminals are in a daisy-chain arrangement with the head-of-chain terminal connected to the ATM network. ATM cells are relayed from the ATM network downstream to the last terminal, and from each terminal upstream to the network. Each video conference terminal has a video program input port provisioned with a unique ATM address, and a plurality of video program out ports, each provisioned with one of the video program input port addresses. ATM cells generated from each video program input port are transmitted upstream to the ATM network where they are relayed back to the head-of-chain terminal and then downstream to the last terminal. Each video conferencing terminal may also receive ATM cells having addresses that match the addresses provisioned for its video program output ports. These cells are converted to analog video programs and transmitted out the associated video output ports.

Abstract: In a multipoint video conference controlling system which is supplied with video packets and audio packets from conference terminals, multiplexes the video packets to produce a composite video packet, and mixes the audio packets to produce a mixed audio packet, a sequence number collating device collates the sequence numbers of the video packets and the audio packets to produce a coincidence signal when the sequence numbers of the video packets and the audio packets coincide with each other. Video and audio transmitting devices transmit synchronously the composite video packet and the mixed audio packet in response to the coincidence signal. The multipoint video conference controlling system may comprise video and audio sequence number processing devices which add one of sequence numbers to the composite packet and the mixed audio packet when the video packets and the audio packets have the sequence numbers.

Abstract: In a receiving node coupled to a number of transmitting nodes, a method of managing audio data received from the transmitting nodes is provided. In the method, a plurality of transmissions are received by the receiving node, with each transmission representing an utterance of audio information. Each transmission is transmitted from a different one of the transmitting nodes and includes a number of audio data packets. One of the utterances is designated as a ready-to-play utterance based on an order in which the utterances were received. The data packets of any utterance designated as the ready-to-play utterance are provided to a ready-to-play queue in real time. Data packets of the other utterances are held in separate queues and are not provided to the ready-to-play queue until after the packets of the ready-to-play utterance have been provided to the ready-to-play queue.

Abstract: A video teleconferencing system uses digital transcoding to obtain algorithm transcoding, transmission rate matching, and spatial mixing. The video teleconferencing system comprises a multipoint control unit (MCU) for allowing multiple audiovisual terminals, which send and receive compressed digital data signals, to communicate with each other in a conference. The MCU has a video processing unit (VPU) that performs algorithm transcoding, rate matching, and spatial mixing among the terminals within a conference. The VPU includes a time division multiplex pixel bus and a plurality of processors. Each processor is assignable to an audiovisual terminal in the conference and is coupled to the pixel bus. In a receive mode, each processor receives and decodes compressed video signals from its assigned terminal and puts the decoded signal onto the pixel bus. In a transmit mode, each processor receives from the pixel bus uncompressed video signals from any terminal in the conference.

Abstract: A multimedia switching apparatus switches multimedia signals including audio and video signals received from a public telecommunications network to distribute the multimedia signals automatically. The gateway receives the multimedia signals and generates a separate audio and video signal output. The audio signal is switched by an audio switch to a destination under the control of an automatic call distribution arrangement to automatically control the switching of the audio signal. A separate video switch is provided for switching the video signal to a destination logically associated with the destination of the audio signal. The video switch is controlled by monitoring the events in the audio switch to control the switching of the video signal so that it is switched to the logically associated destination. The switching operations of the audio switch are copied by the video switch.

Abstract: A communications system having a first data subsystem; a data transmission controller coupled to the first data subsystem; a data line coupled to the data transmission controller having a bandwidth with a first portion and a second portion; and, a second data subsystem coupled to the data transmission controller. A method for dynamically estimating and allocating the bandwidth between the first data subsystem and the second data subsystem having the steps of measuring a utilization level of the first portion of the bandwidth by a first application; detecting whether the utilization level is outside of a predetermined range; and, changing an allocation of the first portion and the second portion of the bandwidth.

Abstract: A method for determining a talk/listen state using a dynamic threshold is provided that can be implemented in a multipoint control unit (12). An audio sample is received and represents sound measured during a sample time interval. An audio level is generated from the audio sample The audio level represents an average power level of the audio sample. A dynamic threshold is set by processing the audio level. The audio level is compared to the dynamic threshold. A talk/listen state is determined to be a talking state if the audio level is above the dynamic threshold and to be a listening state if the audio level is below the dynamic threshold. The previous steps are repeated for each audio sample in a sequential stream of received audio samples such that the dynamic threshold is maintained and used to determine the talk/listen state.

Abstract: A communication terminal apparatus for multiplexing a plurality of different types of data and for performing communications using the multiplexed data includes a network control unit connected to a communication network for transmission and reception of the plurality of different types of data. A multiplexing unit generates transmission data by multiplexing the plurality of different types of data and outputs the transmission data to the line control unit. A synchronizing unit generates a transmission synchronizing signal of the multiplexing unit. A separating unit separates reception data received from the network control unit in units of data type synchronously with the transmission synchronizing signal generated by the synchronization unit. A loop back path can be used for looping back and inputting the separated reception data to the multiplexing unit, and an adding unit adds dummy data to the looped back reception data.

Abstract: Modem-equipped computers which can initiate an audio channel using the modem data connection. The connection is initiated with a new protocol called the voice-over-data protocol. The new protocol does not require any additional modem hardware or telephone line features, and is not tied to any proprietary hardware/software compression or transmission schemes. The voice-over-data protocol negotiates an audio compression/decompression scheme and then sets up an audio channel over an existing data connection using a socket. Compressed audio data is then delivered to the remote computer where it is decompressed and output. The voice-over-data protocol significantly reduces the latency which disrupts normal speech patterns when voice data is sent over a data connection. This protocol also reduces the bandwidth required to send voice over a data connection.

Abstract: A multimedia teleconference system serving sound data to an existing telephone terminal as well as multimedia terminals, and makes the existing telephone terminal join in a multimedia telephone conference by the use of sound data. The multimedia teleconference system includes a first multimedia terminal for generating various kinds of media data including at least, sound data. The teleconference system further including a plurality of second multimedia terminals for setting connections corresponding to each of the various kinds of media data, a telephone terminal for receiving only sound data, and a switch for receiving the various kinds of media-data sent from the first multimedia terminal and distinguishing a connection for the sound data and extracting only the sound data transferring on the distinguished connection to send the extracted sound data to the telephone terminal.

Abstract: A conference call sound level balancing method and system for selectively muting a conferee whose sound environment is loud enough to dominate other conferees of a conference call. The method is operable on a system that includes a network-based conference bridge and a sound equalization platform that is coupled to each telephone of a conference call. If one telephone connected to the conference call has a higher noise level than the other branches of the call, thus making it so the other branches cannot obtain speaking privileges, a predetermined touchtone key is entered at the noisy telephone. The sound equalization platform responds by automatically measuring the ambient sound levels of the respective branches of the conference call and balancing the speaking privileges for the respective call branches of the conference call.

Abstract: There are proposed a communication station apparatus which can perform a multipoint intercommunication by connecting a plurality of points via a communication line, having an inquiry unit for sending, in advance, an inquiry as to participation in the multipoint intercommunication to the stations of members relating to the multipoint intercommunication, and a registration unit for receiving inquiry results sent from the members who received the inquiry from the inquiry unit, and registering a group of personal information of the members relating to the multipoint intercommunication in correspondence with the received results, and a communication station apparatus which can perform a multipoint intercommunication by connecting a plurality of points via a communication line, having an inquiry unit for sending, in advance, an inquiry as to participation in the multipoint intercommunication to stations of members relating to the multipoint intercommunication, and a communication path setting unit for receiving inq

Abstract: In a networked videoconferencing system, a receiver with a number of receiving stations receives calls from a number of different calling stations. To get appropriate bandwidth, two calls are connected at the same time from the calling station to one of the receiving stations. The receiver includes an access switch that uses information from the calling station to pair calls together so that two calls from one calling station are routed to the same receiving station, regardless of the order in which the calls are received by the access switch. The access switch can use one or a combination of routing methods including storing all pairs of incoming numbers in a table and/or by using comparisons of similar numbers.

Abstract: The present invention is embodied in a digital communication system where multiple media data sources are time multiplexed into a packetized data stream, each packet having an assigned priority and the packetized data stream transmitted in substantially the order of assigned priority. At both the transmit side, and the receive side, audio packets are given priority processing over video packets, which in turn have priority over text/graphics data packets. Continuous real time audio playback is maintained at the receiver by delaying the playback of received audio in a first in/first out (FIFO) buffer providing a delay at least equal to the predicted average packet delay for the communication system. Optionally, the average system delay is continuously monitored, and the audio playback delay time is adjusted accordingly. Audio playback is slowed or accelerated in order to shrink or grow the difference in time between the sender and receiver.

Abstract: A video teleconferencing system uses digital transcoding to obtain algorithm transcoding, transmission rate matching, and spatial mixing. The video teleconferencing system comprises a multipoint control unit (MCU) for allowing multiple audiovisual terminals, which send and receive compressed digital data signals, to communicate with each other in a conference. The MCU has a video processing unit (VPU) that performs algorithm transcoding, rate matching, and spatial mixing among the terminals within a conference. The VPU includes a time division multiplex pixel bus and a plurality of processors. Each processor is assignable to an audiovisual terminal in the conference and is coupled to the pixel bus. In a receive mode, each processor receives and decodes compressed video signals from its assigned terminal and puts the decoded signal onto the pixel bus. In a transmit mode, each processor receives from the pixel bus uncompressed video signals from any terminal in the conference.