Abstract: A conference engine operates in a multiplex processing scheme and requires a signal hardware summer to process all of the programmed conferences, all of the outputs and all of the input channels. The conference engine is only limited by the number of available input channels, the number of desired conference registers, the speed of the implementation, and the amount of memory available on the target system. The conference engine allows the number of available conference channels, the number of conferences per bridge and the number of bridges to be scaled. Each bridge contains a unique set of conference registers and has full access to the input channel data. Additional conference bridges are added by increasing the operating clock/memory to meet the processing requirements of the additional bridge. Conference registers in each conference bridge can be added/removed depending on the system requirements and are independent of the number of input channels.

Abstract: A method configured to continuously receive frames from a plurality of video channels and transmit to each of a plurality of participants in a video conference individual frames containing information concerning each of the video channels. The method only transmits at any given instant new picture data for one of the participants in the video conference.

Abstract: Connecting two or more persons by telephone without any person having to give their telephone coordinates. A messaging server adapted to send to one or more invited persons a message sent by an inviting person and containing an invitation to communicate by telephone, to create automatically a number dedicated to a telephone call in which at least one invited person has agreed to participate, and to broadcast the dedicated number to the inviting person and to the invited person(s) who have agreed to participate. An exchange is adapted to connect at least two calls to the dedicated number.

Abstract: One aspect of the invention is a method of using a computer to identify a conversation. Another aspect is a method for an audio processing system that identifies conversations and enhances each conversation for each user in the conversation.

Abstract: A system for collaboration summarization playback includes a graphical user interface (950) for displaying summarization categories (952) associated with recording cues and clips of the multimedia conference. The categories may be arranged as a list or as thumbnails and typically includes a length of time for each category; a time during the conference when the associated clip was recorded; and a media type. The categories are clickable and allow the associated clip to be played or displayed as recorded. In addition, in certain embodiments, a voice recognition transcription of audio clips may be provided.

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: Method for dynamically optimizing bandwidth allocation in a variable bitrate conference environment. Conference means with two or more outputs are provided, where each one can output data at different rates, in order to support two or more endpoints which may have different media rates. Two or more endpoints are connected to these conference means for participating in the conference. Whenever more than one video rate is used by participants during the conference, each set of output rates is selected from all possible combinations of output rates in the conference means, wherein the lowest output rate in each selected set is the entry rate of the endpoint joining the conference at the lowest rate. A Quality Drop Coefficient (QDC) for each endpoint that joins the conference is determined for each selected set, wherein the QDC is computed according to the endpoint entry rate and the highest rate, among the output rates of each selected set, that is lower or equal to said endpoints' entry rate.

Abstract: An embodiment of the invention is a multipoint conferencing system having the ability to dynamically assign an endpoint to a “closest” one of multiple conference servers is disclosed. In one embodiment, when an endpoint first connects to the conferencing system, its Endpoint ID (e.g., IP address, E164 address) is determined. Based on the Endpoint ID, the conferencing system selects the most appropriate ones of the conference servers to act as the endpoint's “local” conference server. That is, the endpoint will transmit and receive conference data from other endpoints via the selected conference server. If no other endpoints across the network happen to be viewing a particular endpoint's video stream, the stream is not provided to the network, thus conserving network bandwidth. An embodiment further creates appropriate linking between conference servers automatically, manages conference server topology and determines link type (e.g., Unicast vs. Multicast linking) automatically.

Abstract: An exchange capable of increasing the number of participants in a conference without increasing conference circuits for conference communications. The exchange (2) switches connections either between a line wire (11) and a plurality of extensions (10a) to (10c) or between plural extensions and plural line wires.

Abstract: The present invention is directed to a system and method for background noise reduction and performance improvement in conferencing. A method for providing a conferencing session may include receiving inputs from a number of participants in a conferencing session. A number of prominent inputs are determined from the received inputs and the determined prominent inputs are combined into a first output stream. The output stream is suitable for being sent to at least one participant of the number of participants in the conferencing session.

Abstract: A method and an apparatus for controlling a conference call is a system which comprises a conference bridge for combining several audio signals arriving from terminals and for transmitting the combined signal further to at least one of the terminals using at least one transmitting codec. The conference bridge comprises a spatialization unit for creating a spatial effect. The number of simultaneously active audio components, such as speech signals, is determined from the audio signals of the conference bridge inputs. At least one transmitting codec is selected for use on the basis of the number of simultaneously active audio components and the combined signal is transmitted to at least one terminal using the selected transmitting codec.

Abstract: In one embodiment, the invention provides a method comprising loading a set of transport components into a memory; initializing each transport component of the set of transport components to listen on a particular conferencing interface using a listen string to associate with a network interface, each transport component of the set of transport components listening to a different conferencing interface, the listen string encapsulating parameters to one or more calls to listen; receiving an incoming call signal on the network interface having an incoming conferencing interface; processing the incoming call signal to detect the incoming conferencing interface; and launching an application based on the incoming conferencing interface. Other embodiments of the invention are also described.

Abstract: Methods and arrangements to play Voice over Internet Protocol (VoIP) messages are contemplated. Embodiments include transformations, code, state machines or other logic to play VoIP messages by receiving overlapping VoIP messages from a VoIP conference, placing at least one of the overlapping VoIP messages in a queue, and playing the overlapping VoIP messages one at a time. The playing may include retrieving the overlapping VoIP messages from the queue. The embodiments may include displaying the VoIP messages through a graphical user interface and receiving instructions from a user on the playing of the messages through the graphical user interface. In some further embodiments, the instructions may describe the order for playing the messages. In many further embodiments, the instructions may request the deletion of VoIP messages from the queue without the playing of the VoIP messages.

Abstract: In a communications system having a plurality of endpoints (240,242,246) each including a logical entity and its physical counterpart, a system for controlling and managing session between at least two endpoints. The system including a registration manager (202), an application layer router (204), a session controller (206), a group database manager (208), and at least one group entity (210).

Abstract: Instances of media items are generated from a selection of media items. The instances are scaled and concurrently transitioned through a display environment.

Abstract: Techniques to perform rate matching for multimedia conference calls are described. An apparatus may comprise a conferencing server and a rate matching module. The rate matching module may be arranged to adjust bit rates between media communications channels for client terminals in a conference call, with the rate matching module to remove video frames from a set of video information received on a first media communications channel to reduce a bit rate for the video information. Other embodiments are described and claimed.

Abstract: A distributed intelligence conferencing system having a plurality of conferencing nodes to connect groups of participants to a conference. Each of the conferencing nodes provides for the connection of one or more participants to the conference. Each node includes a DSP for distributed signal processing. The nodes DSP includes: A signal measuring device for measuring features of the signals from each of the participants. A single core speaker tracking algorithm identifies the active speaker on a local node based upon the relative signal features of each of the number of participant input signals. A communication device communicates the measured signal features for the active local participant input signals only to all other conferencing nodes. A multi core speaker tracking algorithm compares the features of the signals from the speakers on the local node with the features of the signals from the speakers on other nodes.

Abstract: The present invention provides a communication assistance process that is performed by a communication assistance device connected to a plurality of user terminals via a network. During a communication (or video) conference call, when assistance data associated with a topic in the conference is provided to a plurality of user terminals in the form of a voice data, the communication assistance process adjusts the sound volume of voice data in the assistance data if it was determined that voice data is included in the communication data received from the plurality of user terminals, and if voice data is also included in the assistance data, the assistance data that includes sound volume adjusted voice data is then provided to the plurality of user terminals. Accordingly, the communication assistance process allows communications in a conference system to proceed smoothly.

Abstract: Voice packets transmitted by an originating wireless user for use in a half-duplex communication mode in a one to many communication session are monitored for an unexpected cessation. If an unexpected cessation of such voice packets is detected for a predetermined time period, the session server controlling the communications session is notified to release the supporting half-duplex communication paths associated with the originating user.

Abstract: A distributed intelligence conferencing system is disclosed, having a plurality of conferencing nodes to connect groups of participants to a conference. Each of the conferencing nodes provides for the connection of one or more participants to the conference. Each node includes a DSP for distributed signal processing, eliminating the need for a central processor.

Abstract: A video teleconferencing system for controlling multiple multipoint control units (MCU) from a single apparatus. The system utilizes a Virtual MCU (VMCU) (110) to communicate with a plurality of MCUs (135, 140, 145). A user initiates a reserve conference command with the VMCU (110). If sufficient resources are available, the reservation is made and connection numbers are assigned. When the time for the conference arises, an MCU (135, 140, 145) is assigned to the conference. The participants are then connected to the conference. By using a single VMCU (110) to schedule and coordinate multiple MCUs (135, 140, 145), the present invention is able to efficiently schedule a large number of conferences. This greater efficiency in scheduling may allow users to schedule conferences without the advance notice that is usually required.

Abstract: A distributed conferencing system having a plurality of conferencing nodes to connect groups of participants to a conference. Each of the conferencing nodes provides for the connection of one or more participants to the conference. Each node includes a DSP for distributed signal processing. The node DSP includes: A signal measuring device for measuring features of the signals from each of the participants, such as power, zero crossing rate and short term energy and voice activity determination and feature extraction from the signals of the participants connected to each node. Each node has a single core speaker tracking algorithm for determining the relative features of each of the number of participant input signals. The speaker tracking compares the characteristics of the speakers of the core and determines which speakers are to be included and which speakers are to be excluded from presentation to the other nodes in the conference.

Abstract: In communication system (200), a packet data terminal (102) (e.g., personal computer (PC), personal digital assistant (PDA), telephone, mobile radiotelephone, network access device, Internet peripheral, and the like) initiates, coordinates, and controls the provision of on-demand conference call and call waiting services as a function of user inputs without the need for coordination of network resources that dominate conference call service as currently known and practiced. The packet data terminal permits users to establish call conference services for originating and terminating calls, alike. In addition, multi-party call waiting (two or more parties on-hold) services are established.

Abstract: A telephone exchanging apparatus 30 comprises a joined terminal table 341, a joined terminal table operating portion 352, and a reception side designating portion 355. The joined terminal table 341 stores terminal identification information that identifies a telephone terminal 40 that will join a conference. The joined terminal table operating portion 352 operates the joined terminal table 341 corresponding to joined terminal table operation commanding information. The reception side designating portion 355 designates the reception side of a speech packet transmitted from the telephone terminal 40. Since the joined terminal table 341 is operated by the joined terminal table operating portion 352, even if the number of people who join a conference increases, they can be easily added or deleted.

Abstract: When a service receiving server 1 receives a request of a multi-point communication service from communication terminals 3A to 3C, a service ID corresponding to the multi-point communication service and passwords are respectively sent to the communication terminals 3A to 3C receiving the multi-point communication service and are sent to a communication service unit 2. When the communication service unit 2 receives the service ID and the password from the communication terminals 3A to 3C, the communication service unit 2 collates whether or not the service ID and the password sent from the communication terminals 3A to 3C agree with those sent from the service receiving server 1. In case of the agreement, the communication service unit 2 connects with the group of communication terminals 3A to 3C and provides the multi-point communication service for the communication terminals 3A to 3C according to the service ID.

Abstract: A system enables conferencing of multiple participants within a single voice communication session. A replay module records conference audio during the conference and, at appropriate times, provides this buffered audio to selected participants of the conference.

Abstract: The systematic, and possibly repeated, acquisition of several distinct quantities for exploitation by a user system by utilizing a multiplexer with staged architecture without all inputs hard-wired. Each multiplexer stage is addressed by an elementary counter chained with elementary counters for addressing lower stages. The multiplexer inputs are scanned by regularly incrementing the chain of counters. If no precaution is taken, all the multiplexer inputs are scanned without considering their possible absences. To remedy this drawback a first elementary counter addresses the first stage of adjustable counting capacity switches, the elementary counters can address intermediate stages of the switches with controllable shunting circuits, and a global counter is reconfigured, at the end of each counting cycle of the first elementary counter, by commands adjusting the first elementary counter capacity, and activating or inhibiting the shunting circuits.

Abstract: In a multimedia communications system (100) that supports conference calls that include an audio portion and a video portion, a primary video image is selected from a plurality of video images based on an amount of audio data generated. The amount of audio data is determined by counting a number of audio packets or by counting an amount of audio samples in audio packets (204). A dominant audio participant is selected if the difference in the amount of audio exceeds a predetermined threshold (206). If the difference in the amount of audio does not exceed the predetermined threshold (206), the dominant audio participant may be determined by comparing the loudness or volume for each audio participant (207 212). The primary video image is selected to correspond to the dominant audio participant (208, 214). The primary video image remains constant for a predetermined period of time before the possibility to change (210, 216).

Abstract: A mobile communication device (104) loads an application bundle (200), which includes a software application (202) and an indication of a time limited license (206). To enforce the time limited license, the mobile communication device, upon attempting to invoke the application code (310), obtains a secure time reading (312). It is contemplated that a secure time server (116) may be used to facilitate license enforcement. Once the secure time reading is obtained, the mobile communication device compares the time with the license period, and executes the code only if the time reading is within the license period.

Abstract: A system and method for speaker identification in a conference calling unit. The method identifies the speaker, locates the speaker identifier, transfers the speaker identifier to an outgoing path and sends the speaker identifier to endpoints participating in the conference call. If the endpoints have the capability of displaying caller identification, their displays will be updated dynamically with the speaker identifier in place of the caller identifier. The system includes a conference call unit. The unit has a processing unit operable to extract the caller identification of the line having the speaker and a network interface unit operable to send the caller identification to endpoints participating in the call as speaker identification.

Abstract: An Internet Protocol Telephony Network and public switched telephone network that allows one or more call waiting callers to dynamically join in an existing and to establish a multiple-party conference call including the call waiting call. A call waiting call may also be added to an existing conference call.

Abstract: The present invention relates to a method for generating a charge ID in Internet ONLY service, generating the charge ID that includes printable characters using a call ID in the IP gateway; the process is simple and resulted charge ID is absolutely unique. Wherein including: configuring the charge ID including 32 printable characters, each of which is represented with 4 binary bits; taking the call ID including 16 characters, each of which is represented with 8 binary bits; corresponding each call ID character with two charge ID characters, wherein the upper four from eight bits of one of the call ID characters referring to 4 binary bits of one of the charge ID characters, the lower four bits corresponding to 4 binary bits of another charge ID character. Because the scope represented by four binary bits is from 0 to 15, the charge ID characters can exactly correspond one by one relationship with printable characters of 0-9 and A-F.

Abstract: A method and system for independent participant control of audio during a multiparty communication session includes receiving an audio stream from each of a plurality of participants to a multiparty communication connection. An audio control signal is received from a participant for an audio stream of another participant. A conference output stream is generated for the participant by applying an audio parameter based on the audio control signal to the audio stream.

Abstract: The present invention comprises a method for audio/video conferencing. In a preferred embodiment, the method comprises using a dynamic threshold value to determine whether there is speech on a line. One aspect, the method comprises determining a dynamic threshold value based on one or more characteristics of signals received on a port, associating that dynamic threshold value with the port; and comparing one or more characteristics of signals subsequently received on the port to the dynamic threshold value. Signals received over a plurality of ports are summed, but for ports whose signal characteristics have a specified relationship to the dynamic threshold value associated with that port, signals are not contained in the sum.

Abstract: In video conference system terminals, the conventional difficulty in high-quality display of facial expressions of a plurality of participants is solved. In a multipoint connection device connecting a plurality of video conference terminals, image data transmitted from a terminal is decoded, and a face area in the image data is recognized. Then, quantization coefficients for the face area are set to be greater coefficients than those for other areas than the face area, then the image data is compressed, and delivered to the respective terminals. By this arrangement, in image data, a face area with great significance can be re-compressed without degradation of image quality, while an area with less significance such as background can be compressed with high efficiency. Thus, the total code amount can be reduced. Accordingly, even if the system uses a narrow band communication channel, the significant face area can be clearly displayed at the respective terminals.

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: The present invention comprises a method for audio/video conferencing. In a preferred embodiment, the method comprises using a dynamic threshold value to determine whether there is speech on a line. One aspect, the method comprises determining a dynamic threshold value based on one or more characteristics of signals received on a port, associating that dynamic threshold value with the port; and comparing one or more characteristics of signals subsequently received on the port to the dynamic threshold value. Signals received over a plurality of ports are summed, but for ports whose signal characteristics have a specified relationship to the dynamic threshold value associated with that port, signals are not contained in the sum.

Abstract: A multi-line telephony system and method deliver multiple concurrent telephone conversations to a customer premise via a single traditional twisted-pair communications connection, and support telephone services via a traditional dial-up connection. Improved packet-based communications are performed using the multi-line telephony system and method with better transmission characteristics, such as transmission speed, over traditional twisted-pair wire connections to provide H.

Abstract: A conference system is provided for connection in a digital audio signal transmission system (50), for receiving N encoded audio signals from N conferees, wherein N is a positive integer greater than or equal to three. The system comprises a decoding system (90, 91, 92) for decoding the N audio signals, a selection system (100, 130) for selecting one from the N decoded audio signals, and means for switching (160) the selected encoded audio signal from an input stage (80, 81, 82) of the conference circuit to an output (161) of the conference circuit, for transmission at least to the N−1 conferees from whom the selected audio signal did not originate. This arrangement enables the audio from one conferee to be provided as the conference signal, without undergoing decoding and subsequent re-encoding. A decision is made as to whether one signal is appropriate or whether multiple audio signals should be combined.

Abstract: An apparatus and method for providing multimedia conferencing services with selective performance parameters delivers a different representation of the same video signals to each of a plurality of users depending on the hardware limitations, software limitations, network limitations, and user preferences of each user device. Thus, each user can choose the quality of service he/she wants to receive in view of system limitations and cost considerations. The quality of service may be defined in terms of bandwidth, jitter, frame rate, and delay, for example. The apparatus includes a multimedia bridge that receives video signals from at least one of the plurality of users. The multimedia bridge scales-down the video signals as necessary before sending the video signals to the end users. The multimedia bridge also detects congestion occurring during the multimedia conference and scales-down the video signals, if necessary, to meet reduced network availability.

Abstract: A system, method and computer program product which allows both mixing (e.g., PC-based) and non-mixing (e.g., phone-based) clients to participate in a single audio conference. The system includes a hybrid multi-point control unit (i.e., conferencing server) that performs mixing for phone-based clients and multiplexing for PC-based clients. The method and computer program product determines which clients have the capability to mix multiple audio streams and which do not. For those clients capable of mixing, the server multiplexes the packets of audio data received from each client on the active speakers list into a multiplexed stream. For those clients that are not capable of mixing, the server mixes the packets of audio data received from each client on the active speakers list into one combined packet.

Abstract: A personal videoconferencing system includes a videoconferencing appliance connected to a personal computer through a Universal Serial Bus (USB) or similar connection. The appliance is provided with a video camera and microphone for generating local video and audio streams. The local video and audio streams are compressed by a video and audio encoder residing at the appliance, and the compressed video and audio streams are conveyed to the personal computer over the USB connection. The appliance may be further provided with an audio decoder for decompressing a remote audio stream received by the personal computer through a network interface and conveyed to the appliance via the USB connection. A video decoder, located on the personal computer, is configured to decompress the local video stream (received through the USB connection) as well as a remote video stream received through the network interface. The decompressed video streams are then sent to a monitor for display to a user.

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 method and apparatus for providing unified conferencing services in a telecommunications system is described. A DSP chip in the system contains an internal CPU and on-chip memory storage device. The CPU is programmed such that when a conference has been established, the time slots or channels of PCM information are collected and stored. Channels are sub-sampled at a pre-determined optimal time interval. The energy for each sub-sample, which is a mathematical summation over time utilizing a sum of squares equation, is calculated for each channel. These calculated energies are stored in an associated array. The energies are surveyed to select the N loudest channels. The input signals (the voice information) for those N loudest channels are used to form the conference. More specifically, the sum of the N loudest channels is generated. This sum forms the basis for a broadcast output.

Abstract: An IP (Internet Protocol) conference telephone system feasible for an IP-PBX (Private Branch Exchange) includes a communication control connected to an IP network. When a connection for a conference is set up between a plurality of telephones accommodated in the IP network, each telephone packetizes a voice signal and sends the resulting packet to the network to address it to a conference trunk. An IP bridge interconnected to the network receives packets via the network 10, multiplexes the voice signals of the packets on the channels of a down-going highway each corresponding to a particular telephone, and sends the multiplexed voice signals to the conference trunk. The conference trunk performs addition and subtraction with the above voice signals and returns the results of addition and subtraction on an up-going highway. This allows a conference to be held between the telephones. A multicast router may be substituted for the conference trunk, in which case the network will be adaptive to multicast.

Abstract: A system and method for collaborative multimedia production by users at different geographic locations. The users produce sequencer data at a plurality of sequencer stations connected via a network. The sequencer stations encapsulate sequencer data units into broadcast data units and upload and download broadcast data units to and from a server, in response to user commands received at the sequencer stations.

Abstract: A system and method for selectively muting a call of a wireless communication system. A first predetermined signal is received by the wireless communication system. The phone signal has a user signal associated with it. In response to the first predetermined signal, the wireless communication system pauses communication of the user signal to the end user while still maintaining the phone call between the wireless communication system and the conference bridge. In response to a second predetermined signal, the wireless communication system reiates communication of the user signal to the end user.

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: For allocating radio resources in a time-division multiple access packet mode radio communication system, each remote station stores transmit authorizations for each time slot of a frame in a table. The packets are stored in a plurality of queues in each remote station. The table is duplicated and one table is read during a frame while the other table is being written.

Abstract: A method and apparatus is disclosed for data communication between agents, such as those in an electronic conferencing system. In an electronic conferencing a system wherein data is shared between a plurality of participants during an electronic conference users, a method is disclosed for maintaining consistency of the data among the participants during the electronic conference users.