Abstract: A communication manager establishes a call between two or more participants on two or more communication devices. The call can be an audio or video call. A call request is sent from one of the communication devices to conference an additional communication device to the call. For example, a participant in the call decides to conference an additional person to further discuss an idea. An information manager gets a roster of the participants. In addition to the call request, the roster of the participants is sent to the additional communication device. The additional communication device receives the call request and the roster. The roster is displayed to the user of the additional communication device. The user can then indicate to answer the call request. If answered, the additional communication device sends an indication that the call was answered and the additional communication device is conferenced into the call.

Abstract: In an embodiment, a user equipment (UE) participating in a server-mediated communication session maintains an identifier of the session and a list of UEs that are currently participating in the session. The UE determines to suppress its participation level in the session, and configures a member update message to include the identifier and to indicate the suppressed participation level of the UE. The UE transmits the configured member update message to each other UE currently participating in the session. In another embodiment, the UE receives a request to modify one or more of a given set of control parameters associated with the session. The UE determines whether the UE belongs to a subset of UEs that are currently participating in the session and to which permission to modify the given set of parameters is restricted. The UE selectively grants the received request based on the determination.

Abstract: The present invention provides a method, system, and program product for managing conference calls. Specifically, prospective conference call participants (e.g., moderators and/or invitees) will initially provide user profiles. A participant's profile will include, among other things, a set of contact numbers at which the participant can be reached. Thereafter, a moderator and a set of invitees of a conference call to be held are identified. This is typically done based on synchronization with a calendar application used to send and accept an invitation to the conference call. When the moderator later initiates the conference call, the present invention will access the profile of each invitee that accepted the invitation, and then call the invitees at the respective contact numbers set forth therein. When contact is made with an invitee, the call will be merged into the conference call with the moderator.

Abstract: A method for join selection of a conference call from a communication device, wherein the communication device includes a memory for storing an event record. The method includes: identifying from the event record one or more addresses for contacting a conference call server for establishing a scheduled conference call session; automatically selecting, based on specified criteria, one of the addresses as a default address associated with a default join option for establishing the scheduled conference call session; generating an identifier for associating the selected default address with the default join option; and storing the identifier in the memory.

Abstract: Methods and systems for dialing into conference calls are presented. A meeting request is received by an attendee, and the dial-in information in the meeting request is stored. A meeting reminder is displayed at the proposed date and time or a predetermined period prior to the proposed date and time, and a conference call confirmation instruction or cancellation instruction is received from the attendee. The conference call dial-in number is automatically dialed at the proposed date and time. A network connection is established and the conference call meeting identification number is transmitted.

Abstract: In one embodiment, a conference bridge receives a call initiated by a first caller endpoint. The conference bridge identifies a conference to which the call requests entry and determines whether the conference bridge has available resources to process the call. The conference bridge enables the call to join the conference via a second conference bridge associated with a second caller endpoint participating in the conference according to whether the conference bridge has available resources to process the call.

Abstract: Systems and methods that enable high quality audio teleconferencing are disclosed. In one embodiment of the present invention, a signal processor receives signals from a spatially dispersed set of directional microphones, processing the microphone signals and the far-end received audio into a signal for transmission to a far-end party. The processing may comprise the use of one or more algorithms that reduce conference room noise and may selectively increase participant audio levels by processing the microphone signals using beamforming techniques. An embodiment of the present invention may also comprise one or more omni-directional microphones that may be used in cooperation with the directional microphones to adjust for background noise, acoustic echo, and the existence of side conversations.

Abstract: Products for conferencing between a local and a distant party through an electronic channel include a portion installed to a fixed conferencing environment and another portion removable from the fixed environment for roaming conferencing. A breakout box detachable from the removable portion may be used having connections for installation-type conferencing elements such as speakers, microphones and other conferencing devices. The removable portion may include an acoustic echo canceler. The removable portion may be connected to a communication-handling device, such as a computer with VOIP software, or may be directly connected to a communication channel. Speakers or microphones in the removable portion may be automatically or manually disabled on connection with the fixed portion.

Abstract: A communication system automatically mutes a microphone in response to a key event detected from a coupled keyboard to prevent delivery of noise generated by the keyboard to a far end of the communication session. A timer is initiated for a time period when the key event occurs, and the microphone remains muted for the time period of the timer. The timer may be restarted to maintain the microphone mute in response to a subsequent key event detected while the timer is running. If the timer expires before a subsequent key event is detected, the microphone can be restored to its previous state (usually un-muted). The system can be applied to a computer coupled to the keyboard and a microphone and having a communication application operating on the computer. Alternatively, the system can be applied to a computer coupled to the keyboard and coupled to an external conferencing unit having a communication application.

Abstract: A method for protecting moderator access using a communication device. The method includes: displaying an interface for editing a conference call profile, the conference call profile including conference call scheduling information including one or more addressees for communicating with a conference call server, wherein the conference call scheduling information further includes a moderator access code and a participant access code; generating an indicator for the conference call profile for excluding sending of the moderator access code to invited participants; and storing the conference call profile including the indicator in a memory of the communication device. A communication device can be configured to perform the method.

Abstract: A conference diagnosis system can correlate information about a current teleconference with information about past conferences to make an intelligent assessment as to the potential cause of a poor audio or other condition. A conference bridge conducting the conference can then change various configuration settings on the conference bridge to resolve the problem. A running average of the results of past adjustments can be measured and direct future configuration changes. Thus, past successful adjustments can provide a starting point for changes to a new conference and real-time digital signal processing (DSP) adjustments can help tailor changes to the specific conference.

Abstract: A conference call system comprises an input interface for receiving during a conference call at least two input streams of audio signal, each from another source. A selection unit is connected to the input interface, for selecting a number of dominant speaker streams out of the input streams, the number being less than or equal to a maximum number of dominant speakers value and each of the dominant speaker streams representing speech from a respective dominant speaker. A mixer is connected to the selection unit, for mixing the selected streams into an output stream. The conference call system comprises an output interface for outputting the output stream and a selection control unit connected to the selection unit and the input interface, for dynamically setting, during the conference call, the maximum number of dominant speakers value based on dynamics of the conference call.

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 validation service offers the ability to validate a telephone conference. The validation service can verify that a telephone number and a pass code can be used to access the telephone conference at the time the telephone conference is scheduled. The validation service can also inform users about changes to the scheduled telephone conference, and provide alternative telephone numbers that can be used to access the telephone conference.

Abstract: A system and method for merging a first voice call and a second voice call. A first topic discussed during the first voice call is compared to a second topic discussed during the second voice call. If the first topic matches the second topic, the first voice call may be merged with the second voice call. A semantic analysis may be performed on the first topic and the second topic to determine if the first topic and the second topic match one another. Filter criteria may be defined by participants of the first voice call and the second voice call that identify criteria that may preclude merger of the first voice call and the second voice call.

Abstract: A method for joining a conference call from a communication device, the communication device having a locked state and an unlocked state. The method includes displaying an interface on the communication device while the communication device is in the locked state, the interface including an option to join a scheduled conference call, receiving an input for selection of the option while the communication device is in the locked state, unlocking the communication device to the unlocked state, and sending a communication to a second communication device for establishing a conference call session.

Abstract: A method and apparatus for enabling an improved experience by better matching of the auditory space to the visual space in video viewing applications such as those that may be used in video teleconferencing systems using window-based displays. In particular, in accordance with certain illustrative embodiments of the present invention, one or more desired sound source locations are determined based on a location of a window in a video teleconference display device (which may, for example, comprise the image of a teleconference participant within the given window), and a plurality of audio signals which accurately locate the sound sources at the desired sound source locations (based on the location of the given window in the display) are advantageously generated.

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: 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 hierarchical order of those participants attempting/requesting to speak. A participant is then selected from among 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 communication system that includes multiple conferencing devices connected in a daisy-chain configuration is communicably connected to far-end conference participants. Multiple conferencing devices provide improved sound quality to the far-end participants by reducing aural artifacts resulting from reverberation and echo. The daisy-chain communication system also reduces the processing and transmission time of the near-end audio signal by processing and transmitting the audio signal in frequency domain. Each conferencing device in the daisy chain performs signal conditioning on its audio signal before transmitting it in the frequency domain to a mixer. The output signal of the mixer is converted back to the time domain before being transmitted to the far-end. The daisy-chain configuration also provides a distributed bridge to external communication devices that can be connected to each conferencing device.

Abstract: A teleconferencing system for establishing a facilitator-led teleconference involving a plurality of callers, a facilitator, and at least one party that can receive transfers. The teleconferencing system is operated by the facilitator under the control of application software through a user interface. A facilitator-led teleconference is formed and the callers are informed of how to initiate a transfer to the party, such as by touching of a key on their telephone keypad. The system detects the callers who initiate transfer. If the party is available the initiating caller is automatically transferred to that party. If the party is not available, such as would happen if numerous callers initiate transfer to that party, the initiating callers are placed into a transfer queue. When the party becomes available the callers in the queue are then transferred in queue order.

Abstract: A multi-point to multi-point intercom system, formed by at least one intercom server and a plurality of intercom terminals, the intercom terminals registered as talking or listening intercom terminals in an intercom session table. The intercom server sends to listening intercom terminals some or all of the unmixed audio packets received by talking intercom terminals according to the intensity signal value found in the header of the unmixed audio packets.

Abstract: A method of automatically establishing an input language for a handheld electronic device includes receiving a request to initiate the composition of a new message, receiving a selection of a particular contact that is the intended recipient of the message, and determining whether the particular contact has a stored preferred input language. If a preferred input language is stored, the method includes determining whether the preferred input language is different than the current input language (which may be a pre-established default input language). If the preferred input language is different, the method further includes switching the current input language to the preferred input language. Also provided is an improved handheld electronic device having one or more routines adapted to implement the method. Also provided is a method of indicating a preferred language in association with a call received by or placed to the device.

Abstract: Architecture for accessing and purchasing multimedia services in a mobile terminal cellular environment. The architecture brings together the on-the-fly discovery of music and connectivity for the purchase or preview of multimedia services (e.g., music tracks). Provided is a music client on a cell phone or other mobile communications device that facilitates the discovery, storage, and play of multimedia content or portions thereof such as music. A user can listen to music transmitted to the mobile terminal device (e.g., cell phone) via broadcast, satellite radio, streaming internet radio, etc., review the music, and buy tracks or whole albums over-the-air via the mobile terminal.

Abstract: Disclosed are apparatus and techniques for providing presence and location information from available media resources to participants in a media conference so that the participants can use then use such available resources in the conferencing interaction. When participating in such rich media conferences with presence and location capabilities, apparatus and techniques are provided for determining the device(s) being used and their capabilities and determining available resources presently connected to a network that is available to the participants and presenting such presence information to the conference participants. Additionally, apparatus and techniques are provided for obtaining presence and location knowledge of potential resources that are accessible but not connected to the network or shutdown and presenting such information to the participants.

Abstract: A system for conducting a conference call based on a community document. The system includes a data network, a first conference client device and a second conference client device communicatively coupled to the data network, wherein the first conference client device is associated with a first user of the conference call and the second conference client device is associated with a second user of the conference call, and a conference server device communicatively coupled to the data network.

Abstract: A method for facilitating access to a conference call is disclosed. The method can comprise allowing a prospective participant to join the conference call after the prospective participant provides a personal ID rather than a conference ID. The method can comprise using at least one of a prospective participant's personal ID or voice for authentication, accessing the prospective participant's electronic calendar, presenting conference call choices from the prospective participant's electronic calendar to the prospective participant, and adding the prospective participant to a selected conference call.

Abstract: The invention relates to a conferencing unit for multi-point communication in the course of a telephone conference. Previously, it was not possible to call up and document agreements and decisions in voice-based telephone conferences in a computer-aided manner. The reason was that in traditional conferencing units, all audio channels were mixed into an overall channel and therefore an undisturbed interaction of individual participants with the system is not possible. The invention enables an individual association of a conference participant with a decision processing unit, in which decisions can be announced in a dialog system and stored in a computer-readable format.

Abstract: Conference calls are managed using a permanent or default password associated with a particular call-in phone number and at least one temporary, dynamically-generated password for that same call-in phone number. The temporary passwords are preferably applicable for a particular time period or interval. If a temporary password is not defined for the time period in which a conference call takes place, the default password may be used. Conference call access is restricted to participants who provide the proper password—namely, the temporary password during a time period for which a temporary password has been generated, and the default password otherwise.

Abstract: A method of audio scene management in a teleconference or other group sound telecommunication is presented, in which teleconference at least a first transmitting party, a second transmitting party and a receiving party participates. The method comprises receiving of signals representing sound of the first transmitting party and sound of the second transmitting party. The method further comprises obtaining of measures of sound activity for the first and second transmitting parties, respectively and selecting a first angle and/or a second angle based on the obtained measures of sound activity. The method further comprises processing of the received signals into processed signals such that sound from the first transmitting party is experienced by the receiving party as emanating from the first angle while sound from the second transmitting party is experienced as emanating from the second angle, with respect to the receiving party. Finally signals representing the processed signals are outputted.

Abstract: A method and apparatus for performing active speaker selection in teleconferencing applications illustratively comprises a microphone array module, a speaker recognition system, a user interface, and a speech signal selection module. The microphone array module separates the speech signal from each active speaker from those of other active speakers, providing a plurality of individual speaker's speech signals. The speaker recognition system identifies each currently active speaker using conventional speaker recognition/identification techniques. These identities are then transmitted to a remote teleconferencing location for display to remote participants via a user interface.

Abstract: A computer-implemented method for auto play of a status in a telephone conference is provided. The method includes receiving, at an e-mail server, a status message of a party to a telephone conference, and determining from the status message whether the party will be able to attend the telephone conference. When it is determined that the party cannot attend the telephone conference, the method further includes forwarding the status message to a voice box for conversion of the status message into audio that is saved in an audio file. The method may also include notifying a teleconference bridge that the party associated with the status message will not be able to attend the telephone conference. The teleconference bridge creates an entry in a repository indicating that the party will need a voice proxy from the audio file, and, during the telephone conference, the status of the party is provided to other parties to the telephone conference by playing the audio file.

Abstract: There is provided a system for facilitating a conference call. The system includes a module to generate a real-time voiceprint from a voice input of a participant in the conference call, and a module to provide information indicative of the participant based on the real-time voiceprint.

Abstract: Rules-based teleconferencing that may include receiving criteria regarding invitees to a teleconference call, generating rules based on the criteria, and generating a notification regarding the invitees to the teleconference call based on the rules. The criteria may include a quorum of a minimum number of invitees that must be in attendance at the teleconference call or a list of specific invitees required to be in attendance at the teleconference call. A processing device may include a network interface, one or more second interfaces configured to establish a connection between at least three telephones, and a controller configured to establish a teleconference call between the at least three telephones, receive criteria regarding invitees to the teleconference call, generate rules based on the criteria, and generate a notification regarding the invitees to the teleconference call based on the rules.

Abstract: Call replay systems and methods are disclosed. A call replay system can record a telephonic communication and replay the recorded communication upon request. The entire communication may be recorded, or only a portion, such as the most recent few minutes. The entire recording can be transmitted to a requestor, or only a portion. The recording may be altered before it is transmitted to a requestor. Alterations may include actions taken to reduce playback time, such as removing spaces in the recording where no conversation is taking place, or speeding up the recording. Other alterations may include adjusting the sonic characteristics of the recording, translating the recording, or transcribing the recording to create a text version.

Abstract: Techniques for multi-party conferencing are provided. A plurality of audio streams is received from a plurality of conference-enabled devices associated with a conference call. Each audio stream includes a corresponding encoded audio signal generated based on sound received at the corresponding conference-enabled device. Two or more of the audio streams are selected based upon an audio characteristic (e.g., a loudness of a person speaking). The selected audio streams are transmitted to each conference-enabled device associated with the conference call. At each conference-enabled device, the selected audio streams are decoded into a plurality of decoded audio streams, the decoded audio streams are combined into a combined audio signal, and the combined audio signal is played from one or more loudspeakers to be listened to by a user.

Abstract: Conference participant finder methods and apparatuses are provided. The conference participant finder maintains a record of expected conference participants. After the scheduled start time of a conference, expected conference participants are automatically contacted and reminded to join the conference. The contact address used to contact an expected conference participant can be obtained with reference to presence information. In addition, information regarding the status of conference participants can be displayed to a conference host.

Abstract: System and method for enabling push-to-talk (PTT) private calls in a wireless communications network are described. One embodiment includes selecting a called party private identification for a private call by a calling party on a mobile device; pressing a PTT button on the mobile device; transmitting a SIP SUBSCRIBE including the calling and called parties private identifications to request a speech token; redirecting the SIP SUBSCRIBE to the PTT Server for purposes of removing the calling party and the called party from a multicast group; receiving an acknowledge message that includes a speech token; transmitting calling party information from the PTT Server to the called party; communicating speech packets from the calling party to the called party in a half-duplex manner; releasing the PTT button on the mobile device; transmitting a SIP SUBSCRIBE to release the speech token; and notifying the calling and called parties that the group's speech token is available.

Abstract: A method and apparatus for performing lawful intercept in a group call is described. A group call for is established and at least one target member device of the group call is identified as corresponding to a subject to be tracked. Call content for communications corresponding to the call is then tracked from the beginning of the call. A query is presented to a law enforcement authority to determine whether the law enforcement authority wishes to receive call data collected before the target joined the call or after the target leaves the call.

Abstract: Systems and methods for facilitating transferring of a conference call. In some embodiments, users of a secure mobile device may be provided with the ability to transfer a media session for use in a conference call to a non-secure communication device. The method may include displaying a user interface on the primary communication device, the user interface including a transfer option, receiving a command selecting the transfer option, sending a transfer request from the primary communication device to the conference call controller to link an alternate communication device within the conference call session, and determining whether the transfer request requires authentication, and if so performing authentication of the transfer request.

Abstract: A conference call system comprising at least a gateway, a conference reservation agent and a multiport conference unit, can establish conference calls by calling conference call participants at a phone number specified by the conference call participants. When a user initiates a conference, an identifier is assigned to the conference and a URL referencing this identifier is transmitted to each conference participant. Upon clicking the URL, conference participants are presented with a web-form within which the participants can entered a preferred phone number and further join the conference. The multiport conference unit receives each participant phone number and calls the participants at that phone number. Upon establishing a telecommunication connection with each participant, the multiport conference unit bridges the multiple call legs to establish the conference call.

Abstract: A system is disclosed for enabling communication between server resources and a wide spectrum of end-terminals to enable users access to the resources of both converged and non-converged networks via voice and/or electronically generated commands. An electronic personal assistant (ePA) incorporates generalizing/abstracting communications channels, data and resources provided through a converged computer/telephony system interface such that the data and resources are readily accessed by a variety of interface formats including a voice interface or data interface. A set of applications provide dual interfaces for rendering services and data based upon the manner in which a user accesses the data. An electronic personal assistant in accordance with an embodiment of the invention provides voice/data access to web pages, email, file shares, etc.

Abstract: A teleconferencing system that enables facilitator-led teleconferences in which a caller can transfer their call to a different conference group, facilitator, or another party by initiating a transfer command, such as by pushing a particular key on their telephone keypad. Application software controls a switch that automatically transfers a caller to a different conference group, facilitator, or party when a transfer-initiating act, such as pressing a key on a keypad, is sensed. The transfer is reported to the current facilitator and to the new facilitator, if any, who is provided with contact information regarding the transferring-in caller.

Abstract: A method, a device, and a system for implementing a video call are provided. The method includes: when a video call initiated by a calling party is received, performing media negotiation with the calling party and initiating the video call to a called party; and after the called party makes a response, performing media negotiation with the called party and completing establishment of a video voice channel between the calling party and the called party; and when an indication of the calling party and/or the called party is received, processing a video according to the indication. With the method, device, or system of the present invention, the calling and called users can change video contents viewed by the other party or by himself/herself, which provides the capability of actively controlling the video contents for the calling and called users that conduct a video call and improves the users' interactive capability during the video call.

Abstract: Information may be provided to a telephony customer from an overlay service by causing a second stream of audio information to be overlayed on a primary stream of audio information. The overlayed information may be whispered to the user by causing the overlayed information to be compressed into a limited frequency spectrum which is inserted into a manufactured gap in the primary audio stream. This allows the user to continue to hear the main audio stream while also hearing the overlayed information. The overlay service may provide administrative information, advertisements, music, and/or Internet search results via the overlayed audio. Users may issue commands to the overlay service, to have particular information transmitted via overlayed audio. Where the user is also associated with a telephony application running on a computer platform, the overlayed information may be incorporated onto the information being shown to the user on a computer display.

Abstract: An approach for using a centralized conference call manager is provided for initiating conference calls. Users register with the centralized conference call manager to set up a conference call at a future date and time. When the designated time/date arrive, the centralized conference call manager contacts the various intended attendees using a variety of contact locations (e.g., office telephone, mobile telephone, home telephone, etc.). The system further allows an intended attendee to, when contacted by the centralized conference call manager, provide a different contact number that the conference call manager will use instead. An anticipated attendee can decline the call and provide a verbal reason for not being able to attend. The centralized conference call manager records the reason and, when requested, plays the reason back to the other attendees.

Abstract: A communication system that includes multiple conferencing devices connected in a daisy-chain configuration is communicably connected to far-end conference participants. Multiple conferencing devices provide improved sound quality to the far-end participants by reducing aural artifacts resulting from reverberation and echo. The daisy-chain communication system also reduces the processing and transmission time of the near-end audio signal by processing and transmitting the audio signal in frequency domain. Each conferencing device in the daisy chain performs signal conditioning on its audio signal before transmitting it in the frequency domain to a mixer. The output signal of the mixer is converted back to the time domain before being transmitted to the far-end. The daisy-chain configuration also provides a distributed bridge to external communication devices that can be connected to each conferencing device.

Abstract: A method, a computer program product, and a data processing system inform a teleconference participant that a person-of-interest has become active within the teleconference. A user is identified. A person-of-interest, an alert condition, and a notification method are received from the user. The identification, the alert condition, and the notification method are then stored in a person-of-interest data structure. When a user connects to a teleconference, the person-of-interest data structure is retrieved. The teleconference is then monitored for the occurrence of the alert condition. Upon the occurrence of the alert condition, the notification is delivered to the user according to the notification method.

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: A contact center media server for aurally positioning participants of a contact center transaction at aural positions designated by a contact center agent. The media server includes a communications interface coupled to a controller and adapted to interface with a plurality of voice paths. Each of the voice paths is associated with one of a plurality of participants in a contact center transaction. A three-dimensional (3D) spatializer engine is coupled to the controller and can receive incoming voice signals received over voice paths and corresponding aural position data. The 3D spatializer engine processes the incoming voice signals and generates outgoing voice signals that include signal characteristics that aurally position the first outgoing voice signals at an aural position with respect to the contact center agent indicated by the aural position data.