Abstract: A method for adapting a shared organization device for different users of a plurality of users includes: identifying a first user of the plurality of users associated with the shared organization device that is interacting with the shared organization device based on a first user input; determining one or more user-associated display settings for the first user based on a physical ability and a cognitive ability of the first user; displaying an interactive user interface of the shared organization device based on the one or more user-associated display settings for the first user, comprising: displaying one or more interactive affordances at a first location of the touch screen display based on the physical ability of the first user, and displaying one or more symbolic representations of one or more tasks assigned to the first user based on the cognitive ability of the first user.

Abstract: An information processing system which enables more appropriate analysis on a state of communication participated by a plurality of participants is provided. The information processing apparatus includes an utterance judgement unit and an analysis section determination section. The utterance judgement unit judges an utterance section which is a time-series section of an utterance of each of a plurality of participants in the communication. The analysis section determination unit uses a reference analysis section to set boundaries of a plurality of actual analysis sections so that time points of boundaries of the plurality of actual analysis sections become times corresponding to the non-utterance times.

Abstract: An electronic apparatus and a controlling method thereof are provided.

Abstract: A method for estimating a bandwidth characteristic of a communication link between a transmitter node and receiver node includes (1) receiving, from the receiver node, a plurality of synchronization data packets sent by the transmitter node via the communication link, (2) estimating an offset time representing a time required for each synchronization data packet to traverse the communication link, (3) receiving, from the receiver node, a plurality of test data packets sent by the transmitter node via the communication link, and (4) estimating the bandwidth characteristic of the communication link at least partially based on the offset time and characteristics of one or more of the plurality of synchronization data packets and one or more of the plurality of test data packets.

Abstract: A technique is provided that can reduce degradation of the sound quality due to a tandem connection of paired coding and decoding, and can reduce the operation processing amount and the required memory amount of a multipoint control unit. In multipoint connection between terminals of a plurality of communication networks (for example, a fixed phone line and a mobile phone line) having different communication capacities, when a multichannel coding including a monaural coding scheme of a communication network having a smaller communication capacity is used in a communication network having a larger communication capacity to transmit sounds of a plurality points to the communication network terminal having the larger communication capacity, control is exercised such that monaural codes of the plurality points are output.

Abstract: Methods and systems for supporting efficient load balancing among a plurality of switches and a plurality of end nodes arranged in a tree topology in a network environment. The methods and systems can sort the plurality of end nodes, wherein the plurality of end nodes are sorted in a decreasing order of a receive weight. The method and system may further route, in the decreasing order of receive weights, the plurality of end nodes, wherein the routing comprises selecting at least one down-going port and at least one up-going port. Further, the method and system can increase an accumulated downward weight on each selected down-going port by the receive weight of the routed end node, and increase an accumulated upward weight on each selected up-going port by the receive weight of the routed end node.

Abstract: Methods and systems for supporting efficient load balancing among a plurality of switches and a plurality of end nodes arranged in a tree topology in a network environment. The methods and systems can sort the plurality of end nodes, wherein the plurality of end nodes are sorted in a decreasing order of a receive weight. The method and system may further route, in the decreasing order of receive weights, the plurality of end nodes, wherein the routing comprises selecting at least one down-going port and at least one up-going port. Further, the method and system can increase an accumulated downward weight on each selected down-going port by the receive weight of the routed end node, and increase an accumulated upward weight on each selected up-going port by the receive weight of the routed end node.

Abstract: A method and apparatus for providing a bot service is described herein. During operation a bot server will receive a request from a user. An action will be taken by the bot server that is dependent upon a talkgroup associated with the user. Because the action taken by the bot server is dependent upon talkgroup, the action taken can be tailored to specific individuals.

Abstract: Systems, methods, and apparatus for improving bus latency are described. Clock-cycle overhead associated with the transmission of trigger activation information may be reduced through the use of optimized datagram structures for register-configurable trigger activation mechanisms. A first mechanism defines a command code with a first Trigger-Activation datagram, and a second mechanism defines a command code with a second Trigger-Activation datagram that uses a 4-bit Magic-ID and eliminates 18 clock cycles from the conventional Extended Register Write datagram structure. A method performed at a device coupled to a serial bus includes generating a datagram that does not have an address field, populating a data payload of the datagram with trigger activation information directed to a plurality of slave devices coupled to a serial bus, and transmitting the datagram over the serial bus. Transmission of the datagram serves as a trigger that causes a configuration change in at least one slave device.

Abstract: One example method includes receiving, at an electronic computing device, a user query and identifying a time window in the past associated with the user query. The method further includes determining that the user query relates to a public safety incident. The method further includes generating a response to the user query. The response is generated as a function of the time window and information relating to the public safety incident. The method further includes storing the response such that information included in the response is added to one of (i) timeline information associated with a user of a communication device at a time commensurate with the time window and (ii) timeline information associated with the public safety incident at a time commensurate with the time window.

Abstract: With regard to searches and, more particularly, to searches performed on information repositories belonging to an enterprise, a centralized management system is used by the enterprise to manage the predictive search experience for users. A system offers a rich resolution experience to the end users based on user intent as determined from a variety of mechanisms, such as keywords, end user journey, clustered journey, etc. Also disclosed herein is a system that derives and offers various suggestions to end users to help them accomplish their objectives.

Abstract: Methods and systems are provided for automatically pairing headset to gaming audio source based on proximity. A headset that is configurable to process audio may pair with a first audio basestation configured for communicating audio between the headset and a game console, the pairing comprising establishing one or more wireless links for communicating the audio between the first audio basestation and the headset, and pair with a second audio basestation, maintaining pairing with the first audio basestation after said pairing with the second audio basestation. The pairing may occur automatically when the headset comes within a certain distance of the first audio basestation or the second audio basestation. The distance may correspond to physical contact between the headset and the audio basestation. The physical contact may correspond to a tapping of the headset onto the audio basestation.

Abstract: A conference server is configured to receive audio signals associated with active speakers at separate conference endpoints, and to mix the audio signals to form a mixed audio signal. The conference server is further configured to record a mixed audio track comprising the mixed audio signal, and to determine a relative loudness of each of the active speakers for given periods of time. The conference server is also configured to record a plurality of original audio tracks that each comprises the original voice of one or more of the active speakers before mixing, wherein the original voice recorded in each of the tracks at the given periods of time is based on the relative loudness of the active speakers.

Abstract: Techniques are provided for communicating with a plurality of endpoints participating in a conference session. A signal representing audio from each of the endpoints is received at a control unit. The control unit adjusts one or more target power levels for purposes of automatic gain control of the audio from one or more of the endpoints based on information derived in real-time from the audio of the plurality of endpoints during the conference session and predetermined information pertaining to a source of the audio from at least one of the endpoints.

Abstract: A method (250) of intercepting a VOIP communication is disclosed. The method (250) includes: detecting (260) whether a subscriber-caller to be monitored goes off-hook; intercepting (270) the subscriber-caller when off-hook, if lawful intercept is authorized; and sending (280) substantially all communications between the subscriber-caller and a subscriber-callee, to a subscriber-agent. Advantageously, this can provide an efficient, useful, repeatable and scalable way to intercept VOIP communications.

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 method is provided in one example embodiment and includes monitoring media streams associated with channels in a communication environment. The method also includes receiving an alert in one of the media streams and elevating a priority associated with the media stream receiving the alert. Elevating the priority can include increasing the volume of the media stream receiving the alert, changing a spatial direction for rendering the media stream receiving the alert, rendering the media stream receiving the alert to a priority spatial direction, identifying a primary channel and rendering the alert message to the primary channel, turning a volume down on media streams not receiving the alert, or a combination thereof.

Abstract: A method for channel estimation includes: loading a combination of signals sent on other lines over a line of a channel; measuring a signal-to-noise ratio (SNR) of the loaded line; and calculating crosstalk channels of the loaded line according to a coefficient of the combination of signals sent on other lines and the measured SNR. Accordingly, a device and system for channel estimation are provided. The technical solution of the present invention may be used for relevant systems such as an x Digital Subscriber Line (xDSL) system.

Abstract: Second images coded in a coded output video stream are generated for the visual display of image contents of first images coded in multiple coded input video streams, such that first coded image data of the coded input video streams are copied into second coded image data of the coded output video stream.

Abstract: The illustrative embodiments described herein provide improved systems and methods for conferencing enterprise and non-enterprise callers. In one embodiment, a method includes receiving a first call initiated by an enterprise caller at an enterprise, communicating with an enterprise media server to initiate a tributary conference on the enterprise media server, joining the enterprise caller to the tributary conference such that the enterprise caller is in data communication with the tributary conference, receiving a second call initiated by a non-enterprise caller via a public switched telephone network, interfacing the non-enterprise caller with a data center media server, creating a home conference on the data center media center, joining the second call to the home conference, and interfacing the home conference on the data center media server to the tributary conference on the enterprise media server to form a linked conference between the enterprise caller and the non-enterprise caller.

Abstract: The present invention discloses a solution for performing a sharing an ongoing data session between a first user and a group of at least one additional user in a telecommunication system. The basic idea of this invention is to provide a network integration node with a shared streaming session logic, replication logic and group membership management functions of a conference node, like for example the “Push to-talk over Cellular (PoC)” application. Using this functionality the conference system is to be included into the ongoing streaming session, shall invite additional members and forward the streaming traffic to them as well. This results into a shared streaming service using unicast communication resources.

Abstract: A system, method and computer program product for displaying a conversation summary is provided. The method may include providing a graphical user interface (GUI) configured to allow communication between a number of users. The method may also include displaying an invitation to join a conversation at the graphical user interface from at least one of the users and displaying a summary of the conversation in a tag cloud on a computing device. Numerous other variations and embodiments are also included within the scope of the present disclosure.

Abstract: A schedule for a desired call, obtained, for example as an SMS message sent from a user's mobile station, enables a server to automatically initiate set-up of a voice call for the user's mobile station at a scheduled time selected by the user and indicated in the message. The user can specify time and a telephone number for the call, in advance, in the call scheduling request message. However, at the time of the call, the user does not need to read a reminder or dial the call. The call scheduling and automatic set-up services described by way of examples herein may be very useful, for example, for attending important conference calls. This may also be very useful for setting a reminder to call a special person to wish him/her well on his/her birthday/anniversary etc.

Abstract: A method and apparatus for crosstalk channel estimation based on a measured signal-to-noise (SNR) of a loaded line. The method for channel estimation includes: loading, on a newly added line K, a combination of K?1 signals sent on lines 1 to K?1; obtaining a measured SNR of the line K loaded with the combination of the K?1 signals sent on the lines 1 to K?1; and calculating crosstalk channels of the line K according to K?1 coefficients of the K?1 signals sent on the lines 1 to K?1 and the measured SNR. Embodiments of the present invention may be used for relevant network communications systems such as xDSL systems.

Abstract: Conventional game consoles have a preference to connect to the Internet, which facilitates audio communications by gamers. However, when conventional game consoles are connected over a local area network, they somehow inhibit audio communications. A game communications exchange is coupled to game consoles and audio gear sets of gamers to facilitate audio communications by creating a common channel and distribute audio information from the common channel.

Abstract: In one aspect, audio streams are added to a mix until the mix is either complete (i.e., all audio streams have been added) or the mix is closed early (i.e., before the mix is complete). In another aspect, audio and video streams are synchronized by playing back the audio stream and then synchronizing display of the video frames to the playback of the audio stream.

Abstract: A method and apparatus for crosstalk channel estimation based on a measured signal-to-noise (SNR) of a loaded line. The method for channel estimation includes: loading, on a newly added line K, a combination of K-1 signals sent on lines 1 to K-1; obtaining a measured SNR of the line K loaded with the combination of the K-1 signals sent on the lines 1 to K-1; and calculating crosstalk channels of the line K according to K-1 coefficients of the K-1 signals sent on the lines 1 to K-1 and the measured SNR. Embodiments of the present invention may be used for relevant network communications systems such as xDSL systems.

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: A method for replaying a portion of a multi-party audio interaction to a participant of the audio interaction. A participant of a multi-party audio interaction, such as a conference call or a gaming session, can request a replay of a portion of the audio interaction. A conference processor can provide an outgoing audio stream to the participant that includes a replay audio stream that includes the requested replayed portion of the multi-party audio interaction and a real-time audio stream that includes the ongoing multi-party audio interaction. The replay audio stream can be aurally positioned at an aural position that is different from the real-time audio stream to enhance the participant's ability to distinguish between the two audio streams.

Abstract: Methods and systems consistent with the present invention provide efficient packet replication in a router in order to multicast a stream of data. Packets are replicated and processed in a multithreaded environment. Embodiments consistent with the present invention implement a two-stage process for packet replication. The first stage thread will recirculate the packet to multiple second-stage threads. These second-stage threads will then create one or more outgoing copies of the packet. In this way, the copies are handled by multiple threads running in parallel.

Abstract: A system and method for forming a cascaded teleconference between digital telephony devices is disclosed. The method comprises selecting a plurality of digital telephony devices for a teleconference. Each digital telephony device includes a microphone, a speaker, an input port, and at least one output port. A cascaded peer to peer network is formed between the selected digital telephony devices. A sum of the local microphone signal and at least one conference microphone signal is transmitted at the output of each of the plurality of digital telephony devices. The at least one conference microphone signal is broadcast over the speaker at each of the plurality of digital telephony devices to provide the cascaded teleconference.

Abstract: A virtual distributed multipoint control unit in a videoconference may be implemented using a master endpoint, one or more facilitator endpoints, and one or more leaf endpoints. The facilitator endpoint(s) may receive video and/or audio from the leaf endpoints (which may provide/receive videoconference audio and video to one or more local participants). The master endpoint may receive video and/or audio from one or more facilitator endpoints (which may include cumulative video stream from multiple leaf endpoints) and/or directly from one or more leaf endpoints. The master endpoint may arrange received video and/or audio and provide cumulative video and/or audio streams to the leaf endpoints (e.g., directly and/or through the facilitator endpoints). The virtual distributed multipoint control unit may support a conference with more leaf endpoints than the master endpoint has input ports and/or decoders.

Abstract: During a conference, a multipoint control unit (MCU) designates priority and non-priority endpoints. The MCU forms priority audio from the priority endpoint and sends that audio to the other endpoints at a normal level. However, the MCU forms non-priority audio from the non-priority endpoint and handles that audio based on whether the input audio from the priority endpoint is from speaking or not. If the priority endpoint's audio indicates a participant at that endpoint is speaking, then the MCU sends the non-priority audio to the other endpoints at a reduced level. Designation of which endpoint has priority can be based on which endpoint has a current duration of audio indicative of speech that is longer than other endpoints. Alternatively, the designation can be based on which endpoint is currently presenting content during the conference or based on a mix of speech audio and content presentation.

Abstract: A method and corresponding device and system to provide a real-time call announcement to a called device. According to the method, a called device begins receiving call announcement media without having first engaged in session setup signaling to set up a session through which to receive the call announcement media. In normal operation, the device would disregard such incoming media, since no session has been set up for receiving the media. As disclosed, however, the device will also receive at about the same time a call setup request that seeks setup of the underlying call, and the device will treat its receipt of that call setup request as a key to allow playout of the call announcement media even though no session has been set up for receipt of the call announcement media. The called device will then play out the call announcement media as an announcement for the call.

Abstract: A method for realizing forward channel sharing in time division communication system, which completes group configuration at the cluster control service center server and modifies the attributes of all the users to enable the users to support multi-user packet MUP service, including: (a) the cluster user initiates a group call; the access network allocates a unique group identifier MACIndex for the group call after receiving the call request from the calling cluster user and informs all of the users within the group about this MACIndex (201); (b) after the communication starts (202), the access network packets the user data of one or more groups with multi-user packet MUP, packets the user data of each group into one sub-packet and does scheduling, then transmits the packets through forward channel (203); (c) after receiving the MUP packets from the forward channel, the cluster user looks into the MUP packets for the sub-packet that matches the MACIndex of its group, thereby realizing group communication (204

Abstract: A method, a system, and a device of video communication are provided. A sub-picture in a synthetic picture corresponding to each audio stream received is marked. Position information of each audio stream is obtained according to a position of each sub-picture in the synthetic picture. The audio stream and the corresponding position information are sent to a terminal. The terminal processes an audio signal according to the received position information of the audio stream, so that the audio stream carries the position information. During the video communication, the terminal receives the audio stream and the position information, and performs position information sense processing to the audio stream according to the position information. This reduces the frequent communication between the terminal and a device for processing position information and improves the flexibility of the system.

Abstract: A transceiver includes an analog front end (AFE) device and first, second, third, and fourth transmitters. A digital signal processor (DSP) receives digital receive signal and digital transmit signals and generates a shared error control signal and first, second, third, and fourth individual error control signals. An echo canceller system generates an estimated echo signal. First, second, and third NEXT canceller systems generate first, second, and third estimated NEXT signals, respectively, each based on the second analog receive signal and respective ones of the analog transmit signals from the second, third, and fourth transmitters during the first mode and each based on the shared error control signal, respective ones of the second, third, and fourth individual error control signals, and the respective ones of the analog transmit signals from the second, third, and fourth transmitters during the second mode.

Abstract: Methods and apparatus for automatically connecting an emergency caller with appropriate responders within an interoperability and collaboration system. According to one aspect of the present invention, a method includes obtaining a request for a response, and identifying a first location from which the request originated. The request is obtained from a source that utilizes a phone and is located at the first location. The method also includes identifying at least one potential responder that may be capable or appropriate for providing the response. Finally, the method also includes creating a virtual talk group that includes the source and the potential responder. The potential responder may listen substantially directly to communications from the source within the virtual talk group using a land mobile radio.

Abstract: A system and method related for performing lookup operations for objects distributed among different nodes in a peer-to-peer network are disclosed. Various nodes in the peer-to-peer network may store objects. Objects stored on a given node may be accessed by other nodes in the peer-to-peer network. To access an object, a node may first perform a lookup operation to determine where the object is stored, i.e., to determine which node in the peer-to-peer network stores the object. The peer-to-peer network may utilize a method to improve the performance of object lookup operations. In one embodiment, the method may allow object lookup operations to be performed with a latency on the order of one hop.

Abstract: In a computer system having a memory, a processor, and a network interface, a method for listening on multiple conferencing interfaces having the steps of loading a set of transport components into the memory; initializing each transport components of the set of transport components to listen on a particular conferencing interface using the network interface, each transport component of the set of transport components listening to a different conferencing interface; 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.

Abstract: A method is provided for forming a multi-media communication path between at least first, second and third communication devices coupled to a multi-media provider system during post answer call redirecting and/or teleconferencing. The method includes receiving and processing a first call request at a circuit-based portion of the multi-media provider system for forming a first communication link between the first and second communication devices. Thereafter, predetermined attributes of the first communication link may be sent to an IP-based portion of the multi-media provider system for configuring the IP-based portion of the multi-media provider system to process a subsequent request to execute post answer call redirecting and/or teleconferencing.

Abstract: A system and method for optimizing call setup latency for geographically dense groups including updating client location information using at least one application server while the client moves from one sector to another to provide a location update; identifying and updating parameters including location update information uniquely identifying a sector from which a client is receiving communication service; and determining at least one geographically dense call group; and choosing at least one designated responder to the at least call group. A client per cluster is selected as a designated responder for that cluster and when the server needs to set up a geographically dense group call, a random number is included corresponding to the designated responder for that group in a call setup message.

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: A voice conferencing system assigns voice conferences across multiple media processors. The voice conferencing system may thereby allow voice conferences to proceed, even when any single media processor in the conferencing system does not have the resources needed to handle the voice conference. The voice conferencing system may enhance communication capabilities, without significantly increasing cost or equipment requirements.

Abstract: Disclosed is a method for dynamically allocating channel resources in a wireless communication system. A request for a channel resource by a new talkgroup call is received. Whether the new talkgroup call is associated to an associated communication based upon an operational relationship of the new talkgroup call to the associated communication is determined. The channel resource for the new talkgroup call based upon the operational relationship is allocated.

Abstract: In one embodiment, a method includes correlating, for each intermediate device associated with a Real-time Transport Protocol (RTP) application that connects endpoints through a plurality of intermediate devices across a network, one or more first RTP segments input into the intermediate device with one or more second RTP segments output from the intermediate device. The correlation results are then graphically displayed results of the correlating to obtain a full, end-to-end picture of RTP quality from a source endpoint to a destination endpoint of the RTP application. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure.

Abstract: Embodiments according to the present invention allow multiple parties equipped with telecommunications devices (endpoint devices) of similar or differing audio capabilities to join in a conference call and enjoy the fidelity associated with their equipment. Rather than operating with the frequency and fidelity limits of traditional telecommunications circuits (for example, about 300-3500 Hz), at least a portion of a conference bridge according to the present invention operates at higher frequencies. Conference participants using high-fidelity endpoint devices hear each other with greater clarity and reproduction accuracy. Conference participants using conventional interfaces send and receive lower-fidelity audio, but it affects only their own participation and not that of the entire conference. Participants can connect via for example, wireline, wireless Public Switched Telephone Network (PSTN), or via networks such as the Internet. Multiple voice encoding schemes are accommodated.

Abstract: The present invention is directed to a contact center that includes a switching fabric 110, 130 operable to configure a communication session between a first customer and a first resource 138 for servicing of a first contact by the first resource and a contact tracking agent 232 operable, during the servicing of the first contact by the first resource, to (a) monitor the first contact center endpoint for at least one of (i) a change in contact state, (ii) the connection of the first resource and the first customer through a second (new) communication channel, and (iii) the addition of a party to and/or removal of a party from the communication session and (b), when the at least one of (i)-(iii) occurs, terminate a first contact part and create a second contact part. The first and second contact parts are associated with the communication session.

Abstract: An apparatus for mixing audio signals in a voice-over-IP teleconferencing environment comprises a preprocessor, a mixing controller, and a mixing processor. The preprocessor is divided into a media parameter estimator and a media preprocessor. The media parameter estimator estimates signal parameters such as signal-to-noise ratios, energy levels, and voice activity (i.e., the presence or absence of voice in the signal), which are used to control how different channels are mixed. The media preprocessor employs signal processing algorithms such as silence suppression, automatic gain control, and noise reduction, so that the quality of the incoming voice streams is optimized. Based on a function of the estimated signal parameters, the mixing controller specifies a particular mixing strategy and the mixing processor mixes the preprocessed voice streams according the strategy provided by the controller.

Abstract: A method for managing conference resources includes receiving from an endpoint of a user a call to join a multipoint conference between a plurality of participants. Each participant communicates on the multipoint conference through a respective endpoint. The method includes determining that an insufficient amount of resources exist to allow the user to communicate on the multipoint conference and determining participation instructions to enable the user to communicate on the multipoint conference. The method also includes communicating the participation instructions. Location information associated with the user and location information associated with at least one of the plurality of participants may be used to determine the participation instructions to enable the user to communicate on the multipoint conference.