Abstract: A speakerphone includes a processor, a memory device, a power management unit, a first microphone to receive audio waves, a second microphone to receive audio waves, and a third microphone to receive audio waves. The speakerphone may also include a digital signal processor (DSP) to detect a single-user mode activated at the speakerphone, process the audio waves received by the first microphone, second microphone, and third microphone to determine the wave phases of the audio waves received by the first microphone, second microphone, and third microphone, calculate a direction of a voice of a single user relative to the speakerphone; and process the voice of the single user and filter other voices detected by the first microphone, second microphone, and third microphone from the user's voice.

Abstract: Offset cartridge microphones are provided that include multiple unidirectional microphone cartridges mounted in an offset geometry. Various desired polar patterns and/or desired steering angles can be formed by processing the audio signals from the multiple cartridges, including a toroidal polar pattern. The offset geometry of the cartridges may include mounting the cartridges so that they are immediately adjacent to one another and so that their center axes are offset from one another. The microphones may have a more consistent on-axis frequency response and may more uniformly form desired polar patterns and/or desired steering angles by reducing the interference and reflections within and between the cartridges.

Abstract: Systems and methods for dynamic voice accentuation and reinforcement are presented herein. One embodiment comprises one or more audio input sources; one or more audio output sources; one or more band pass filters; and a processing control unit that includes an audio processing unit, and which executes a method: differentiating between audio input sources as vocal sound audio input sources and ambient noise audio input sources; increasing the gain of the vocal sound audio input sources; inverting a polarity of an ambient noise signal received by each of the ambient noise audio input sources; and adding the inverted polarity to either an output signal of at least one of the one or more audio output sources, or to an input signal of at least one of the vocal sound audio input sources, to reduce ambient noise.

Abstract: An utterance in any of various languages is processed to derive a predicted label using a generated grammar. The grammar is suitable for deriving meaning of utterances from several languages (polyglot). The utterance is processed by an encoder using word embeddings. The encoder and a decoder process the utterance using the polyglot grammar to obtain a machine-readable result. The machine-readable result is well-formed based on accounting for re-entrances of intermediate variable references. A machine then takes action on the machine-readable result. Ambiguity is reduced by the decoder by the well-formed machine-readable result. Sparseness of the generated polyglot grammar is reduced by using a two-pass approach including placeholders which are ultimately replaced by edge labels.

Abstract: The amount of far-field noise transmitted by a primary communication device in an open-plan office environment is reduced by defining an acoustic perimeter of reference microphones around the primary device. Reference microphones generate a reference audio input including far-field noise in the proximity of the primary device. The primary device generates a main audio input including the voice of the primary speaker as well as background noise. Reference audio input is compared to main audio input to identify the background noise portion of the main audio signal. A noise reduction algorithm suppresses the identified background noise in the main audio signal. The one or more reference microphones defining the acoustic perimeter may be included in separate microphone devices placed in proximity to the main desktop phone, microphones within other nearby desktop telephone devices, or a combination of both types of devices.

Abstract: The application discloses a multichannel echo cancellation circuit and method and smart device. The multichannel echo cancellation circuit comprises signal extraction circuits, one ends of which are connected to the audio channels of the corresponding loudspeakers, to extract part of the audio signals from the audio channels of the corresponding loudspeakers as echo cancellation reference signals; the other ends of the signal extraction circuits are connected to the isolating circuits; the isolating circuits are interconnected to form one noise channel by which the echo cancellation reference signals extracted by each of the signal extraction circuits are formed into one channel of noise signal and then outputted to the processor; and the processor subtracts the noise signal from the sound signal collected by the microphone according to the noise signal inputted and a sound signal collected by a microphone, to obtain a signal that has been denoised.

Abstract: Various embodiments describe techniques for switching microphones in a multiple microphone system. The techniques incorporate sampling audio signals from multiple microphones, determining a microphone that has the greatest incoming amplitude during the analysis window, and switching the microphone to that greatest amplitude microphone. The transition point for switching microphones may be determined when either the amplitude of the incoming signal is within an error bound of zero or at a “zero-crossing” in the input amplitude stream.

Abstract: An audio system embodiment includes a loudspeaker cabinet having at least one loudspeaker transducer and defining a longitudinal axis. Several microphones are distributed around the longitudinal axis, defining an array of microphones. A reference microphone is positioned in the loudspeaker cabinet, e.g., in a rear chamber of the at least one loudspeaker transducer. The audio system includes a processor and a memory having instructions that, when executed by the processor, cause the audio system to receive an audio signal from each distributed microphone and the reference microphone, and therefrom to estimate a direction, relative to the plurality of microphones, of a nearby, acoustically reflective surface. Responsive to the estimated direction, the audio system affects a mode of operation, e.g., beam forms an audio output in a selected direction corresponding to the estimated direction of the acoustically reflective surface. Related principles are described by way of reference to exemplary embodiments.

Abstract: A conferencing system includes a display device that displays video received from a remote communication device of a communication partner. An audio stream is transmitted to the remote communication device. The audio stream includes real-world sounds produced by one or more real-world audio sources captured by a microphone array and virtual sounds produced by one or more virtual audio sources. A relative volume of sounds in the audio stream is selectively adjusted based, at least in part, on real-world positioning of corresponding audio sources, including real-world and/or virtualized audio sources.

Abstract: An audio processing system includes a group of microphones associated with a dynamic network of microphones and a receiver. The receiver is configured to identify a first signal received by a microphone in the plurality of microphones that is designated as a primary microphone, and identify a subset of microphones included in the plurality of microphones, where each microphone in the subset is associated with a respective signal corresponding to the first signal. The receiver is further configured to calculate a weighting factor for each microphone included in the subset based on the first signal and the respective signal and opportunistically establish a connection with a microphone associated with the dynamic network of microphones that is not included in the plurality of microphones; and, based on a signal received from this microphone, adjust a weighting factor for at least one of the microphones in the subset.

Abstract: In one embodiment, a method includes capturing sound using a plurality of microphones, wherein the plurality of microphones is associated with a computing system. The method also includes determining energy levels for the plurality of microphones, and determining signal-to-noise ratios (SNRs) for the plurality of microphones. Finally, the method includes selecting a particular microphone of the plurality of microphones based on the energy levels and the SNRs, wherein selecting the particular microphone includes providing audio signals obtained by the particular microphone to the computing system for use.

Abstract: A method of extending battery life of a wireless microphone unit includes muting the wireless microphone unit responsive to a mute signal from a base station unit, transmitting, by the wireless microphone unit, compressed muted audio data, wherein the compressed muted audio data is compressed via a first compression scheme, determining, by the wireless microphone unit, whether an unmute signal has been received from the base station unit, and responsive to a determination that the unmute signal has been received, unmuting the wireless microphone unit. The method further includes discontinuing transmission of the compressed muted audio data and transmitting compressed audio data via a second compression scheme, wherein the first transmitting step causes the wireless microphone unit to consume less power per unit of transmission time than the second transmitting step.

Abstract: Embodiments of the present invention pertain to an apparatus, method, and a computer program configured to connect a translator located in a remote area to a conference holding area, and also connect a customer and customer service representative to the translator in order for the translator to translate between the customer service representative and the customer, when the customer services representative determines that a translator is needed.

Abstract: The present invention provides systems and methods employing Voice over Internet Protocol (VoIP) technology to provide multi-channel, multi-access voice communication capabilities.

Abstract: A method and an audio conference device for carrying out an audio conference are disclosed, whereby classification information associated with a respective audio date flow is recorded for supplied audio data flows. According to a result of an evaluation of the classification information, the audio data flows are associated with at least three groups which are homogeneous with regard to the results. The individual audio data flows are processed uniformly in each group in terms of the signals thereof, and said audio data flows processed in this way are superimposed in order to form audio conference data flows to be transmitted to the communication terminals.

Abstract: A conference call system with a plurality of endpoints, a switch and a media mixer is disclosed. One or more entities in the conference call system comprise part or whole of a feature module. The feature module executes various telephony features like adjusting the volume of a selected participant in a conference call, initiating a sidebar with a participant wherein other participants cannot hear the conversation of the participants in the sidebar, putting a selected call participant on hold from an endpoint other than the selected participant's endpoint, disconnecting a selected participant from an endpoint other than the selected participant's endpoint and transferring a call with two or more participants to a new participant's endpoint.

Abstract: A method, a system, and computer readable medium comprising instructions for optimizing on-premise conferencing are provided. The method comprises receiving at least one call from at least one caller, identifying at least one call from at least one caller, and connecting the at least one call to a conference call bridge via a single connection.

Abstract: Processing a communication is disclosed. A first request to establish communication is received at a redirection node from a requesting communication participant. A destination communication participant associated with the request is determined. A communication session in which the requesting communication participant, the redirection node, and the destination communication participant are participants at the same time is established, including sending from the redirection node to an equipment associated with the destination communication participant a second request to establish communication between the redirection node and the equipment associated with the destination communication participant.

Abstract: One embodiment of the invention is a method that includes a mobile device receiving via a wireless communication channel a request for an identifier corresponding to a network device. The mobile device transmits the identifier. The mobile device receives a network identifier corresponding to the network device. The mobile device utilizes the network identifier to establish communication with the network device via the wireless communication channel.

Abstract: A method of and system for adjusting audio volume in a conference call environment are disclosed. The method comprises associating respective gain factors with each source of a plurality of incoming audio streams. The method further comprises automatically adjusting the volume of each incoming audio stream in accordance with the associated gain factor. In accordance with example embodiments, the method may be performed either at a telephony endpoint such as a VoIP telephone or at a conference bridge.

Abstract: Data processing and communications devices are configured to generate, using command signals received from at least one user input device, a signal set representing a meeting request, the meeting request signal set comprising signals representing one or more parameters for a proposed meeting and configured for causing a second data processing and communications device to initiate a voice call on receipt of command signals representing a response to the meeting request; and forward the generated meeting request signal set to the second data processing and communications device.

Abstract: A multi-carrier receiver radio monitors frequency channels associated with at least one talkgroup to which a multi-carrier receiver radio subscribes. The talkgroup is associated with a set of frequency channels that may carry group call activity. The radio monitors each frequency channel of the set of frequency channels associated with the talkgroup for group call activity. During monitoring, when a new call is detected on one frequency channel of the set of frequency channels associated with the talkgroup, the radio joins the new call. While joined to the new call, the radio continues to monitor the set of frequency channels associated with the talkgroup for additional transmissions to the talkgroup and to monitor a set of frequency channels associated with another talkgroup to which the multi-carrier receiver radio also subscribes for group call activity on the other talkgroup.

Abstract: A GPS mobile terminal (2) that a person (1) carries monitors position information on a dog (3) to whom a GPS terminal (4) is attached. When the person (1) and the dog (3) are separated more than a preset distance, the GPS mobile terminal (2) that the person (1) carries uses PoC communication enabling intragroup communication such as a transceiver to make a call request to mobile terminals (6, 7, 8) of all guardians belonging to any one of a plurality of groups (A, B, C) made up of the guardians. When there is a response from a group of the guardians, the communication is enabled in the group, so that the guardians can guide the person (1) to a safe place.

Abstract: A method and apparatus are provided for recording an exchange between a central monitoring station and a protected premises. The method includes the steps of transmitting an alarm notification from a protected premises to an alarm routing service, the alarm routing service establishing a voice connection between the protected premises and a central monitoring station in response to the alarm notification and the alarm routing service recording audio on the voice connection between the protected premises and the central station.

Abstract: The invention relates to a method for audio switching and conferencing. The method comprises: providing a plurality of audio channels comprising at least one active audio channel, the active audio channel comprising at least one of an input audio stream and output audio stream; converting the input audio streams from the at least one audio channel in input data; providing audio channel communication requests between parties of the at least one active audio channel; determining a set of Boolean values depending on the parties of the audio channels; determining output data for the respective active audio channels by combining the elements of the Boolean set and the input data; and encoding the output data in output audio streams for the respective active audio channels.

Abstract: It is recognized in the disclosed technique that the way impairments on a conference call are diagnosed and handled in the prior art can sometimes be as distracting as the impairments themselves. The disclosed technique attempts to address some of the disadvantages in the prior art by monitoring the signal path quality on a conference call and, based on the quality of the signals received, performing one or more actions that are intended to minimize distractions while handling the impairments. In particular, in accordance with the illustrative embodiment of the present invention, the technique performs a multi-threshold evaluation of media signals that are received at a conferencing server. If a measure of the impairment being considered exceeds a first threshold, but not a higher second threshold, then a first set of actions is taken to handle the impairment. If the measure exceeds the second threshold, then a more-extreme second set of actions is taken.

Abstract: Some embodiments provide an architecture for establishing multi-participant audio conferences over a computer network. This architecture has a central distributor that receives audio signals from one or more participants. The central distributor mixes the received signals and transmits them back to participants. In some embodiments, the central distributor eliminates echo by removing each participant's audio signal from the mixed signal that the central distributor sends to the particular participant.

Abstract: Processing a communication is disclosed. A first request to establish communication is received at a redirection node from a requesting communication participant. A destination communication participant associated with the request is determined. A communication session in which the requesting communication participant, the redirection node, and the destination communication participant are participants at the same time is established, including sending from the redirection node to an equipment associated with the destination communication participant a second request to establish communication between the redirection node and the equipment associated with the destination communication participant.

Abstract: In one embodiment, a method comprises receiving through a telephony user interface from a user at the telephony user interface an indication of a first direction of a sound source along a first axis with respect to the user and a second direction of the sound source along a second axis with respect to the user. The first axis is substantially perpendicular to the second axis. The method further involves communicating the indication of the first direction and the second direction to enable audio communication of sound originating at the sound source to the user as if coming from the first direction and the second direction with respect to the user.

Abstract: A teleconference terminal apparatus including: an input unit which receives a speech signal; an analyzing unit which calculates a target size on a predetermined segment basis of a speech signal; a coding unit which codes the speech signal to generate a data stream, so that the coded data size on a predetermined segment basis becomes the target size corresponding to each predetermined segment; a stream transmitting unit which transmits to a network the data stream; a receiving unit which receives the data stream transmitted from another terminal apparatus; a filtering unit which determines whether segment data is to be decoded based on data size for each predetermined segment in the received data stream, the segment data being included in the data stream; a decoding unit which decodes segment data determined to be decoded to generate a speech signal; and an output unit which outputs the generated speech signal.

Abstract: Processing a communication is disclosed. A first request to establish communication is received at a redirection node from a requesting communication participant. A destination communication participant associated with the request is determined. A communication session in which the requesting communication participant, the redirection node, and the destination communication participant are participants at the same time is established, including sending from the redirection node to an equipment associated with the destination communication participant a second request to establish communication between the redirection node and the equipment associated with the destination communication participant.

Abstract: A conference control method includes: an Application Server (AS) sets a digit collection template for the conference session; a Media Resource Server (MRS) detects and collects dialing operations of participants on the conference session according to the digit collection template set by the AS, and reports the digit collection result to the AS; the AS controls the conference according to the received digit collection result. A conference control system includes an AS and an MRS, where the AS is adapted to set a digit collection template for the conference session and send the set digit collection template information to the MRS, and adapted to receive the digit collection result from the MRS and control a conference according to the received digit collection result; the MRS is adapted to detect and collect dialing operations of participants on the conference session according to the digit collection template set by the AS, and send the digit collection result to the AS.

Abstract: Data processing and communications devices are configured to generate, using command signals received from at least one user input device, a signal set representing a meeting request, the meeting request signal set comprising signals representing one or more parameters for a proposed meeting and configured for causing a second data processing and communications device to initiate a voice call on receipt of command signals representing a response to the meeting request; and forward the generated meeting request signal set to the second data processing and communications device.

Abstract: A teleconference between at least two participants is evaluated by monitoring the teleconference and detecting a triggering event if more than one participant desires to speak at the same time. Upon detecting a triggering event, e.g., detecting an attempt of one or more participants to interrupt another participant, or detecting a request initiated by one or more participants to interrupt another participant, a dominant source is selected from among the ones of the participants attempting/requesting to speak based at least in part upon a determined hierarchal order of those participants attempting/requesting to speak. A participant is then selected from the participants to have the role of the dominant participant based upon the selected dominant source and only the speech of the dominant participant is conveyed to each participant through the teleconference system.

Abstract: A separate peer-to-peer multi-directional audio channel is established between a subset of the participants to an existing multi-party communication session on the same channel as the original multi-party communication session. Since the secure channel is established on the existing call, the same NAT/firewall traversal mechanisms may be used, and even the same socket may be used. Participants on the secure side conversation can hear both the secure side conversation and the main multi-party communication session. Participants that are not on the secure side conversation may receive packets from the secure side conversation, but are not part of the security association for the secure side conversation and, hence, cannot hear the audio from the secure side conversation.

Abstract: Devices are able to identify other devices in a shared audio space (e.g., a room) by emitting a signal and receiving messages from other devices over a data connection indicating which of the other devices received the emitted signal. Devices that are determined to be in the same audio space can selectively mix audio in a teleconference to inhibit mixing of audio signals for devices that are determined to be in the same audio space.

Abstract: A method, a system, and computer readable medium comprising instructions for optimizing on-premise conferencing are provided. The method comprises receiving at least one call from at least one caller, identifying at least one on-premise call from at least one caller on the premise, collecting the at least one on-premise call from the at least one call, and connecting the at least one on-premise call to a conference call bridge via a single connection.

Abstract: Disclosed herein are methods, systems, and devices for improved audio, video, and data conferencing. The present invention provides a conferencing system comprising a plurality of endpoints communicating data including audio data and control data according to a communication protocol. A local conference endpoint may control or be controlled by a remote conference endpoint. Data comprising control signals may be exchanged between the local endpoint and remote endpoint via various communication protocols. In other embodiments, the present invention provides for improved bridge architecture for controlling functions of conference endpoints including controlling functions of the bridge.

Abstract: An apparatus includes a base unit and cordless handsets configured to establish a call with a communication apparatus, and a storage unit configured to store information corresponding to the communication apparatus indicating whether a multi-party call among the communication apparatus and at least two of the base unit and the cordless handsets is permitted. Also, a processing unit is configured to determine whether to enable a multi-party call among one communication apparatus, and at least two of the base unit and the cordless handsets based on the information. A method for use with the apparatus can include, in response to receiving a call from one communication apparatus, determining whether to enable a multi-party call to be established among the one communication apparatus and at least two of the base unit and the cordless handsets based on the information.

Abstract: A network entity, method and computer program product are provided for effectuating a conference session. The method may include receiving a plurality of signals representative of voice communication of the participants. In this regard, the signals may be received from a plurality of terminals of a respective plurality of participants at one of the locations, each of at least some of the terminals otherwise being configured for voice communication independent of at least some of the other terminals. The method of this aspect also includes classifying speech activity of the conference session according to a speech pause, or one or more actively-speaking participants, during the conference session. The signals of the respective participants may then be mixed into a at least one mixed signal for output to one or more other participants at one or more other locations, the signals being mixed based upon classification of the speech activity.

Abstract: The invention relates to a method for managing a packet switched, centralized conference call between a plurality of terminals 13. In order to enable an enhancement of the user comfort, it is proposed that the method comprises at a conference call server 12 receiving data packets from all terminals 13. Based on these data packets, then at least one terminal 13 currently providing voice data is determined. In a next step, the data received in the data packets is mixed, and the mixed data is inserted into new data packets together with at least one identifier associated to one of the terminals 13 which were determined to provide voice data, such that the at least one identifier can be distinguished from any other information in the data packets. Finally, the new data packets are transmitted to terminals 13 participating in the conference call.

Abstract: A digital audio system configured to facilitate a collaborative audio session comprises a first digital audio device operable by a first user, a second digital audio device operable by a second user independently of the first digital audio device, and a server configured to communicate with the first digital audio device and the second digital audio device over a network, receive a request from at least one of the digital audio devices for participation in a collaborative audio session, and facilitate communication between the first digital audio device and the second digital audio device.

Abstract: Processing a communication is disclosed. A first request to establish communication is received at a redirection node from a requesting communication participant. A destination communication participant associated with the request is determined. A communication session in which the requesting communication participant, the redirection node, and the destination communication participant are participants is established, including sending to equipment associated with the destination communication participant a second request to establish communication. A communication session processing feature not provided with respect to the communication session by a telecommunication service provider associated with a service provider network used to establish and provide the communication session is made available to at least one of the requesting communication participant and the destination communication participant.

Abstract: A multiparty calling method, which includes initiating a multiparty call between at least a first, second and third terminal, and setting a flag in the first terminal to inform a processor of the first terminal to skip a process of closing an existing communication path between the first terminal and the second terminal, when the first terminal is disconnected with the third terminal. Further, the set flag is released when the multiparty call between the at least first, second and third terminals has been established for a predetermined time period. In addition, the first terminal preferably sets the flag when the multiparty call join command is transmitted to the network.

Abstract: Disclosed herein are methods, systems, and devices for improved audio, video, and data conferencing. The present invention provides a conferencing system comprising a plurality of endpoints communicating data including audio data and control data according to a communication protocol. A local conference endpoint may control or be controlled by a remote conference endpoint. Data comprising control signals may be exchanged between the local endpoint and remote endpoint via various communication protocols. In other embodiments, the present invention provides for improved bridge architecture for controlling functions of conference endpoints including controlling functions of the bridge.

Abstract: A method is disclosed that breaks the “one line, one location” paradigm of teleconferencing in the prior art. The teleconference bridge in the illustrative embodiment is able to utilize more than one audio channel from each location. By having access to more than one audio channel from a teleconference location's sound field, the bridge is able to create a multi-channel effect during a conference call. When more than one microphone is present in the first sound field, they can be used to create a multi-channel effect in a second or other sound field involved in a conference call that has more than one loudspeaker. With the bridge mimicking the multi-channel imaging in one room with the multi-channel imaging in another, the conference call participants can experience audio depth during a conference call and can experience two-dimensional imaging, depending on the microphone or speaker separation that is present.

Abstract: A voice conference apparatus includes: a communication unit that transmits and receives a first voice signal to and from another apparatus via a network; an output unit that reproduces the first voice signal received by the communication unit and outputs the reproduced first voice signal as a first voice; an input unit that collects the first voice and a second voice generated by a voice source, as a second voice signal; a separating unit that separates the second voice signal into a third voice signal originated from the first voice and a fourth voice signal originated from the second voice; and a selecting unit that selects one of the third voice signal and the fourth voice signal. The communication unit transmits the one of the third voice signal and the fourth voice signal selected by the selecting unit to the other apparatus.

Abstract: A system for establishing a conference call is provided. The system includes a handset operable to send a message containing a list of prospective participants for the conference call. The system also includes an application server operable to receive the message and invite at least some of the prospective participants. The application server is further operable, in response to receiving an acknowledgement from at least one of the prospective participants to participate in the conference call, to include the at least one of the prospective participants in the conference call.

Abstract: A method and system for identifying conference participants who dial in to a telephone conference of an electronic conference that includes a web conference is provided. To identify a conference participant, a conference system displays to a user the names of those conference participants who have not yet been associated with a telephone line of the telephone conference. The conference system plays to the user the identification announcement of a conference participant who is not yet associated with a telephone line. When the user hears the identification announcement, the user recognizes the name of the conference participant and selects the name of that conference participant from the displayed names. The conference system can then associate that conference participant with the telephone line associated with the identification announcement that was played to the user.

Abstract: Disclosed herein is an apparatus. The apparatus includes a user input region. The apparatus is configured to move a first icon displayed on the apparatus in response to a received input at the user input region. The apparatus is configured to assign the first icon to a first personal contact. The apparatus is configured to assign a second icon to a second personal contact. The apparatus is configured to initiate a conference call between the first personal contact and the second personal contact in response to a movement of the first icon onto the second icon.