Broadcom Corporation

Verizon Patent and Licensing Inc.

Avaya Inc.

Prakash C Khanduri

Nelson Ray Sollenberger

Ronish Patel

Emil F Stefanacci

John J Jetzt

Mark A Crandall

Michael D Lange

Hanks Zeng

Anwarul Hannan

Ken Kasiske

Yuk Lun Li

Huaiyu Zeng

David Bien

Baris Yildiz

Bosen Zhao

Charles Lane

A system, method, and computer-readable medium provide for determining that a user device does not include a subscriber identity module (SIM) profile activated to provision a wireless service of a first mobile network operator (MNO); obtaining location data for the user device; generating multiple mobile network codes (MNCs) and corresponding public land mobile network (PLMN) identifiers (IDs) for multiple MNOs that service an area associated with the location data, wherein the multiple MNOs include the first MNO; configuring, using the PLMN IDs, a radio frequency (RF) channel associated with each of the MNOs using each PLMN ID; generating temporary device IDs for the user device; transmitting a discovery message including each device ID via the RF channel associated with each MNO; receiving from the first MNO storing a reserved SIM profile for the device ID, a network address; and downloading the reserved SIM profile via the network address.

Systems and methods for scalable remote SIM provisioning

A method, device, and computer-readable medium provide for identifying a user of a first end device based on first user identification information; registering the first end device in a provider network based on first user credentials, the first end device having a SIM profile associated with an MDN; generating a network session ID to access a first application service from the server device via a first network connection; initiating a first pairing mode with a second end device; identifying the user based on second user identification information obtained from the user via the second end device; determining that the second end device has a higher hosting priority than the first end device; sending a request to the server device to transfer use of the MDN to the second end device; and establishing a second network connection to the server device and releasing the first network connection based on the transfer.

Systems and methods for service enablement and end device activation

A method, a device, and a non-transitory storage medium are described in which a full duplex system may use information from a user of the device as a basis to configure parameters of a dynamic range controller system. The parameters may include high pass filter values, compression values, and masking values. The full duplex system may use a pre-distortion filter for generating an estimated echo feedback reference signal. For generating the estimated echo feedback reference signal, the pre-distortion filter uses an echo coupled reference signal via microphone path as a reference signal, and an echo feedback reference signal via speaker feedback path as the input. An echo canceller of the full duplex system may use the estimated echo feedback reference signal, which takes echo path distortion into consideration, for cancelling both linear and non-linear echo.

Method and system for full duplex enhanced audio

A wearable device, such as a smartwatch, may be used to enable switching of network services between endpoints. The switching of network services may include transferring the telephone number associated with a user's account to a particular active device, selected from a number of possible devices, of the user. The switching may be based on a number of factors and coordinated through the wearable device. The factors may include, for example, the location of the user, the location of the user relative to the user's endpoints, user provided priority information, and/or other user-specified profile information relating to endpoint preferences.

Wearable device as service enabler

A device may provide a baseline uplink audio signal and a baseline downlink audio signal to create a double talk communication for a user device. The device may receive an audio signal, transmitted by the user device, based on providing the baseline uplink audio signal and the baseline downlink audio signal. The device may determine a score based on the audio signal. The score may indicate an effectiveness of the user device with regard to processing audio associated with the double talk communication. The device may provide information that identifies the score to indicate the effectiveness of the user device with regard to processing audio associated with the double talk communication.

Testing audio quality associated with a user device during a double talk communication

Methods and apparatus for customizing the audio parameters of a networked voice communications device (NVCD) are presented. One method includes receiving a settings file, determining a group parameter, extracting at least one audio control parameter from the settings file based upon the group parameter, deriving audio processing parameters based upon a value selected from the at least one audio control parameter, and controlling the audio characteristics of the networked voice communications device using the audio processing parameters. An apparatus having audio characteristics configured by a settings file is also presented, which includes a processor, and memory which is functionally coupled to the processor, where the memory stores a group parameter and executable instructions, and the executable instructions cause the processor to control the audio characteristics of the networked voice communications device based upon the group parameter.

Methods and apparatus for customizing the audio characteristics of networked voice communications devices

A communication device may determine total gain for a downlink processing path via the communication device. The total gain may be determined based on determination of audio requirements for each of a plurality of audio processing devices supported via the communication device, during the downlink processing path. The communication device is operable to determine a default mix signal gain, a downlink gain adjustment and a calibration gain based on the determined audio requirements, and to calculate the total gain for the downlink processing path based on the determined default mix signal gain, the downlink gain adjustment and the calibration gain. The determination of the total gain may be based on determination of a telephony state in the communication device. The calibration gain may be determined during calibration of the communication device, whereas the downlink gain adjustment may be calculated based on an adjustment step-size and/or a mode based gain index.

Method and system for audio system volume control

A telecommunication system including a fall duplex speakerphone, comprising a first microphone to generate a coupled signal including uplink information and non-linear distortion sensed by the first microphone in a speaker phone mode, a second microphone to generate a reference signal including downlink information and the non-linear distortion sensed by the second microphone in the speaker phone mode, and an acoustic echo canceller (AEC) to receive the coupled signal from the first microphone, to receive the reference signal from the second microphone, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the reference signal.

Full duplex speakerphone design using acoustically compensated speaker distortion

A telecommunication system including a speakerphone provides a coupled signal path, including a microphone, configured to sense an incoming audio signal and an echo signal and generate a coupled signal. The echo signal includes non-linear distortion generated by a speaker of the speakerphone. An echo signal path, including an amplifier, is configured to sense the echo signal and generate an echo reference signal. The echo reference signal includes the non-linear distortion. An acoustic echo canceller is configured to receive the coupled signal from the coupled signal path, to receive the echo reference signal from the echo signal path, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the echo reference signal.

System for reducing speakerphone echo

A VoIP access point may be operable to provide VoIP servicing to a plurality of wireless audio endpoint devices. The VoIP access point may extract VoIP audio data received via IP backbone and communicating the extracted audio data as non-VoIP formatted data to the wireless audio endpoint devices. In the uplink direction, the VoIP access point may receive non-VoIP formatted audio data from the wireless audio endpoint devices and pack the received data into IP packets for VoIP communication. The VoIP access point may also be operable to perform PCM encoding/decoding operations during VoIP servicing operations. The wireless audio endpoint devices may perform audio processing during VoIP communications via the VoIP access point. One or more intermediary communication devices may be utilized to route messaging between the VoIP access point and at least some of the wireless audio endpoint devices.

Method and system for endpoint based architecture for VoIP access points

Methods and systems for a dual echo canceller (EC) are disclosed and may include cancelling echo in utilizing a dual echo canceller, wherein said dual echo canceller includes an active echo canceller and an adaptive echo canceller. Filter coefficients may be copied from the adaptive echo canceller to the active echo canceller for the cancellation, based on whether said adaptive echo canceller has converged. The coefficients may be copied utilizing copy logic, which may comprise divergence detection and/or echo path change detection. The coefficients may be reset to default settings utilizing the copy logic. The coefficients may be calculated utilizing normalized block least mean squares (NBLMS), and may be calculated when the NBLMS is enabled by update logic. The coefficients may be calculated utilizing linear predictive coefficient (LPC) filtered uplink and downlink signals.

Method and system for a dual echo canceller

Methods and systems for dynamic range control in an audio processing system are disclosed and may include controlling a dynamic range of an audio signal by expanding the dynamic range utilizing a dynamic expander, and dividing the audio signal into a plurality of frequency bands. Each of the bands may be individually compressed utilizing a multi-band compressor. A sum of the individually compressed frequency bands may be compressed utilizing a full-band compressor. The audio signal may be filtered utilizing a pre-emphasis filter, such as an infinite impulse response filter and may be divided into frequency bands utilizing one or more finite impulse response filters and/or delay modules. The dynamic expander may include adaptive thresholds and an envelope detector. Each of the frequency bands may be compressed utilizing syllabic compression in the multi-band compressor. The compressed sum of compressed plurality of bands may be processed utilizing an audio CODEC.

Method and system for dynamic range control in an audio processing system

In a method and apparatus for measuring voice quality of an endpoint in a Voice-over-IP (VoIP) network, a plurality of network quality parameters are determined for the endpoint. A voice quality factor for the endpoint is determined in real time based on the network quality parameters.

Method and apparatus for measuring voice quality on a VoIP network

Methods and apparatus for controlling the audio characteristics of a networked voice communications device (NVCD) are presented. One method presented includes receiving a settings file, extracting at least one audio control parameter from the settings file, deriving audio processing parameters based upon a value selected from the at least one audio control parameter, and controlling the audio characteristics of the networked voice communications device using the audio processing parameters and the at least one audio control parameter. A method for providing audio parameters to an NVCD is also presented which includes establishing a settings file, which includes at least one audio control parameter, receiving a request to send the settings file, and sending the settings file over a network to the networked voice communications device.

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Freehold, NJ, United States of America

Prakash C Khanduri