Abstract: Systems and methods for identifying suspected contraband mobile electronic devices within a confined community are provided. Details regarding said advertising opportunities are monitored from central advertising providers. A subset of the mobile electronic devices associated with advertising opportunities noting location measurements within a perimeter geofence of the confined community are identified and any such devices also associated with advertising opportunities noting location measurements outside of the perimeter are eliminated to arrive at a revised subset. The published details of the advertising opportunities for those mobile electronic devices in the revised subset are recorded.

Abstract: Systems and techniques for permitting, establishing, monitoring and controlling calls based on messages from outside parties, outside party availability information, institution's call restriction information, and/or account holder preferences. In one example, a message from an outside party app on an outside party computing device is receive by an inmate app on an inmate device. The inmate device displays the message at the inmate calling app including a request for the inmate to call the outside party device during a time period. When the inmate initiates a call to the outside party the call is permitted or not permitted based on determining whether the call is within the specified time period.

Abstract: An exemplary embodiment provides an equalizer adjustment method. The method includes: performing a handshake operation to establish a connection with a host system by a memory storage device; in the handshake operation, receiving a first signal from the host system and performing a first modulation on the first signal by the adaptive equalizer; after the handshake operation is ended, receiving a second signal from the host system and performing a second modulation on the second signal according to a modulation result of the first modulation by the adaptive equalizer to compensate the second signal; and adjusting the adaptive equalizer according to a modulation result of the second modulation.

Abstract: Embodiments relate to enhancing echo cancellation in a transceiver integrated circuit (IC) for full-duplex communication by providing a signal path connected to a dummy driver that replicates a signal path between a main driver and a counterpart transceiver IC to cause a duplicated signal generated by the dummy driver to more closely replicate a sending signal generated by the main driver. The signal path connected to the dummy driver includes circuit elements having transmission line parameters and RLC parameters that replicate transmission line parameters and RLC parameters of circuit elements in the signal path between the main driver and the counterpart transceiver IC.

Abstract: A system of at least one wireless access device in a local area network (LAN) having a plurality of transceivers. Each transceiver has a directional antenna positioned in a substantially circular array to communicate signals with a plurality of stations in a corresponding sector. Each sector defines a portion of a coverage area surrounding the wireless access device. The wireless access device has a network interface to a data network, and an array controller to control communication of data between the stations and the transceivers, and between the transceivers and the network interface. The array controller is configured to modify channel assignments so as to minimize channel interference dynamically.

Abstract: An apparatus and method for adaptively adjusting a target Received Signal Strength Indicator (RSSI) for Automatic Gain Control (AGC) in a full-duplex relay apparatus with an interference cancellation function. A Signal to Interference Ratio (SIR) estimation block estimates an SIR based on an interference signal estimated from a received signal and an interference-cancelled signal determined by cancelling the estimated interference signal from the received signal. A target RSSI determination block determines a target RSSI according to the estimated SIR. An AGC block adjusts a gain for the received signal based on the determined target RSSI.

Abstract: Combined echo and crosstalk cancellation is provided. Frequency domain adaptive filters are used to remove or reduce the effects of echo and crosstalk for a multi-channel and full-duplex communications system. Data from each transmit channel is buffered and converted to the frequency domain. The frequency domain data is multiplied by crosstalk coefficients to obtain a frequency domain correction signal for each channel. Adaptation of the crosstalk coefficients is based on correlations between the error signals and the data from each of the transmit channels. A single frequency domain transform engine, such as a Fast Fourier Transform engine, is employed for all calculations to save power and area.

Abstract: Selective audio combination for a conference. The conference may be initiated between a plurality of participants at respective participant locations. The conference may be performed using a plurality of conferencing endpoints at each of the participant locations. Audio may be received from each of the plurality of conferencing endpoints. Audio level information may also be received from each of the plurality of conferencing endpoints. The audio may be combined from a plural subset of the plurality of conferencing endpoints to produce conference audio. The plural subset is less than all of the plurality of conferencing endpoints. The audio may be combined based on the audio level information. The conference audio may be provided to the plurality of conferencing endpoints.

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: 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: Some embodiments provide an architecture for establishing multi-participant video conferences. This architecture has a central distributor that receives video images from two or more participants. From the received images, the central distributor generates composite images that the central distributor transmits back to the participants. Each composite image includes a set of sub images, where each sub image belongs to one participant. In some embodiments, the central distributor saves network bandwidth by removing each particular participant's image from the composite image that the central distributor sends to the particular participant. In some embodiments, images received from each participant are arranged in the composite in a non-interleaved manner. For instance, in some embodiments, the composite image includes at most one sub-image for each participant, and no two sub-images are interleaved.

Abstract: A new approach is proposed that contemplates systems and methods to support switching between computer audio and presenter audio during a conference call so that one and only one, not both, of the two audio streams is being transmitted to the rest of the conference at any time. More specifically, a switch engine is adopted, which separately takes one or more audio streams captured by a sound mixing component of the hosting device of a presenter and by a microphone attached to the hosting device as its two inputs, and outputs only one of the two inputs at any time. The selected audio output is then processed and transmitted to the rest of the conference with feedback from other audio sources canceled.

Abstract: Combined echo and crosstalk cancellation is provided. Frequency domain adaptive filters are used to remove or reduce the effects of echo and crosstalk for a multi-channel and full-duplex communications system. Data from each transmit channel is buffered and converted to the frequency domain. The frequency domain data is multiplied by crosstalk coefficients to obtain a frequency domain correction signal for each channel. Adaptation of the crosstalk coefficients is based on correlations between the error signals and the data from each of the transmit channels. A single frequency domain transform engine, such as a Fast Fourier Transform engine, is employed for all calculations to save power and area.

Abstract: A method and apparatus for providing echo cancellation are disclosed. For example, the method receives via an echo canceller a first audio signal directed towards an endpoint device, and adds an inserted signal to the first audio signal directed towards the endpoint device. The method determines if a second audio signal being received from the endpoint device comprises the inserted signal, and cancels at least a portion of the second audio signal that is associated with the first audio signal, if the inserted signal is detected.

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: 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: 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 a closed-loop wireless communication system, a codebook-based precoding feedback compression mechanism is provided to remove redundancy from the precoding feedback that is caused by channel correlation in time and frequency. Redundancy due to temporal correlation of the transmission channel is removed by sending precoding feedback only if there is a change in the precoder state for the channel to the receiver. Redundancy due to frequency correlation is removed by run length encoding the precoding feedback, thereby compressing the precoding feedback prior in the frequency domain. By compressing the precoding feedback, the average rate of precoder feedback is reduced.

Abstract: Disclosed are systems and methods which provide call processing for a plurality of telephones at a site for which calling services are provided. Call control functionality is distributed to remote locations using data networks according to embodiments. The aforementioned data networks and/or separate data networks are used to carry call content using voice over Internet protocols (VoIP). Embodiments utilize VoIP to carry calls from a location at which calling services are provided to a centralized call processing platform providing call processing functionality, such as calling party identification, call validation, call routing, and connection to the public switched telephone network (PSTN). PSTN interfacing by a call processing platform may be provided as a VoIP connection to the PSTN and/or as POTS trunking. Call processing platforms may provide for data sharing, aggregation, and/or analysis across multiple facilities served.

Abstract: Briefly, in accordance with one or more embodiments, a method and device capable of canceling self-jammer and one or more other interfering signals in an radio frequency identification system or the like is disclosed.

Abstract: Combined echo and crosstalk cancellation is provided. Frequency domain adaptive filters are used to remove or reduce the effects of echo and crosstalk for a multi-channel and full-duplex communications system. Data from each transmit channel is buffered and converted to the frequency domain. The frequency domain data is multiplied by crosstalk coefficients to obtain a frequency domain correction signal for each channel. Adaptation of the crosstalk coefficients is based on correlations between the error signals and the data from each of the transmit channels. A single frequency domain transform engine, such as a Fast Fourier Transform engine, is employed for all calculations to save power and area.

Abstract: A method of controlling teleconference signals includes receiving, at a teleconference bridge, endpoint-generated audio signals from each of a plurality of participating endpoints arranged at a plurality of locations. At least two of the participating endpoints are acoustically collocated at one of the locations. A bridge-generated audio signal is generated for each of the participating endpoints based on a set of signals. The set of signals includes all of the endpoint-generated signals received at the teleconference bridge, exclusive of the endpoint-generated signals transmitted to the bridge from each of the acoustically collocated endpoints.

Abstract: A method for determining at least one operational transmit power Snk over a communication channel (Ln) coupled to a disturbance causing transceiver (n) for at least one tone (k), comprises the steps of provision of at least one reference victim communication channel (Lref) for representing at least one reference victim for all victim communication channels which are degraded by cross-talk interferences by said communication channel (Ln), and maximizing the data rate (Rref) over said at least one reference victim communication channel (Lref) under the constraint that the data rate over said communication channel (Ln) achieves a target rate (Rntarget) without exceeding a power constraint (Pnmax).

Abstract: A single chip integrated circuit wireless data processor demodulates N separate data signals from M separate antennas simultaneously. The multi-antenna processor can be coupled to a baseband processor on the IC, so that it responds to changing channel conditions between two access points, and selectively kicks in if there is noise, interference, frequency fading, a need for an enhanced data rate, a need for an increased operating range, etc. to improve a performance of the baseband processor.

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. The invention relates equally to a corresponding server and to a corresponding terminal.

Abstract: A “Client-Side Echo Canceller” provides a unique system and method for reducing Multipoint Control Unit (MCU) computational overhead in a multi-point audio conference. In general, the local audio input signal of each client is transmitted in real-time to the MCU. The MCU then combines the audio input signals of all clients to create a single composite signal that is transmitted back to all clients in real-time. Each client then locally processes the composite signal to remove each client's local contribution to the composite signal prior to local playback in order to eliminate a local echo of each client's local audio input. In various embodiments, local cancellation of the local audio input from the composite signal is performed on either a time domain or a transform domain representation of the composite signal. Further, since each client receives the same signal, MCU transmission bandwidth can be reduced via multicast transmissions.

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

Abstract: A method and apparatus are disclosed for canceling acoustic echo generated during transmission of text information in a communication system supporting a TTY/TDD service. When a mobile-to-land call is established between a mobile subscriber and a PSTN subscriber, a decoder notifies an encoder of detection of TTY/TDD text in a BS vocoder. The encoder then transmits a packet having silence information, which prevents acoustic echo reproduction of the text in the TTY/TDD device.

Abstract: A transmitting and receiving arrangement is disclosed having a transmission path, a reception path, and a control device for suppression of interference signals which are produced by crosstalk from the transmission path to the reception path. The control device inputs correction signals that are derived from the transmission signals into the reception path at a point therein such that the interference signals are suppressed downstream from the point. The control device has a detector that detects the interference signals on the basis of characteristics of the transmission signals.

Abstract: A radio apparatus, to be used as a first ending node of a communication network, is connectable to a hands-free apparatus by a local radio link to form a hands-free system. A call made on the hands-free apparatus for an second ending node in the communication network is sent to the radio apparatus through a control channel on the local radio link. The radio apparatus and the hands-free apparatus arrange a voice channel to be established on the local radio link. The call is then sent to the second ending node. In a case where no voice channel is established within a predetermined time period after arranging the voice channel, the radio apparatus cancels the call to prevent a connection between the radio apparatus and the second ending node from being kept and charged.

Abstract: A virtual gateway bridges voice and related signaling traffic between multiple endpoints behind the same or different Network Address Translation devices or firewalls or neither. The virtual gateway also serves as a mixer when used in a teleconferencing mode, which handles a number of endpoints that are behind the same Network Address Translation device or firewall or not. In this mode, the virtual gateway mixes packets on its incoming side with other packets on its incoming side so that all outgoing packets include voice (or other) data from all sources. The embodiments herein may be used with Voice over IP (VoIP) connections and applications, as well as other applications in which two or more endpoints are attempting to establish a communications channel and their IP addresses are otherwise not suitable for the communications application.

Abstract: Systems and methods for reducing transmit echo in a transceiver are disclosed. A hybrid echo canceller includes a limited tap length FIR filter to cancel short-term echo, while an interpolated filter is used to cancel the long-tail echo. The limited tap length FIR filter adapts and calculates coefficients for each adapted tap. Taps of the interpolated filter, on the other hand, are adapted but coefficients are calculated for a subset of the taps. Various interpolation schemes may be applied to the calculated coefficients to associate a coefficient with each tap of the interpolated filter. The technique presented produces an effective filter length of N taps with a reduction in computation and signal processing resources. Preferred embodiments of the echo canceller may be construed as methods for reducing transmit echo.

Abstract: In a wireless code-division multiple access (CDMA) system (100), a talkgroup (101) of subscriber units is provided. A sub-talkgroup (102) of subscriber units, forming a part of the talkgroup is assigned at least one inbound channel (416-417). The talkgroup is assigned outbound channels (415). Members of the sub-talkgroup may simultaneously transmit voice information (410-411) using the at least one inbound channel, which voice information is summed (412) and re-transmitted to the talkgroup using the outbound channels. Voice information is summed so that an individual talker receives summed voice information without the individual subscriber's voice content. Subscriber units in the sub-listengroup are allowed to transmit voice information only after requesting, and receiving, an additional inbound channel.

Abstract: Computers, particularly personal computers, have become everyday communications devices, nearly as common as telephones and televisions. A big reason for this is the expansion of computer networks, such as the Internet, which allow sending and receiving information, including audio and video information, from one computer to one or more other computers. Communicating audio and video information between computers can be problematic because users have to wait for the information to be fully downloaded before viewing or hearing it and because different computers sometimes communicate using incompatible communications protocols, which prevents comprehension. Accordingly, the inventors devised new methods, software, and computer systems that support interactive data streaming (real-time exchange of multimedia data) and full-duplex communications, for example, network conferencing, using more than one communications protocol.

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

Abstract: A transmit processor that includes a transmit frequency domain equalizer (TX FEQ) that pre-compensates packets in the frequency domain to flatten transmit filter response to improve spectral mask and reduce PER. The TX FEQ taps may be selected so that the average power at output equals the average power applied to the input. The TX FEQ may be designed to yield 64 QAM packets with no quantization distortion at the input of an Inverse Fast Fourier Transform (IFFT) processor. The 64 QAM constellation map points and TX FEQ gain values are selected to avoid exceeding the bit resolution of the IFFT processor. The tones may be bound together into zones to reduce implementation complexity by reducing the number of TX FEQ taps.

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

Abstract: A distributed packet-based audio conferencing system, method for packet-based audio conferencing, and a transceiver for use in such conferencing are disclosed. The system uses a collection of transceivers, with each conference participant connected to a local transceiver. When a participant speaks, the local transceiver is responsible for relaying the speaker's voice over a packet network by multicast transmission to transceivers local to each other conference participant. If multiple participants speak simultaneously, a multicast talk stream may originate from each speaker's local transceiver. The total number of simultaneous speakers, however, is limited by an arbitration function resident in each transceiver. The system reduces the costs associated with an always-up communication system.

Abstract: A far-end modem transmits a "pilot tone" during the time that a near-end modem is training its echo canceler. Correspondingly, the near-end modem is modified to notch, or filter, out this pilot tone from the received signal. The filtered received signal is then used by the near-end modem to train its echo canceler.

Abstract: A fast response gain tracker operates to adjust the magnitude of an echo cancellation signal. The latter signal is developed from an echo canceler. The fast response gain tracker provides an adjusted echo cancellation signal that is subtracted from an echo-corrupted received signal to provide an echo-canceled signal. In particular, the fast response gain tracker correlates the echo-canceled signal with the echo cancellation signal from the echo canceler. If the gain of the gain tracker is correct the echo is canceled and there is zero correlation between the echo-canceled signal and the echo cancellation signal. However, if the echo level changes then a residual echo component appears in the echo-canceled signal, which now becomes correlated with the echo cancellation signal. In response, the magnitude of the gain tracker automatically adjusts the gain of the echo cancellation signal to match the change in echo level and thereby subsequently eliminates the residual echo.

Abstract: A noise reduction filter (28) is provided. The noise reduction filter (28) comprises a signal analysis block (30). The signal analysis block (30) is operable to receive an input audio signal and to determine a short term signal level. A noise floor block (32) is coupled to the signal analysis block (30). The noise floor block (32) is operable to receive the short term signal level from the signal analysis block (30) and to determine a noise floor level based upon the value of the short term signal level. A noise factor block (34) is coupled to the signal analysis block (30) and to the noise floor block (32). The noise factor block (34) is operable to receive the short term signal level and the noise floor level and operable to calculate a noise factor. A multiplier node (36) is coupled to the noise factor block (34) and receives the input audio signal.

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