A system includes participant nodes that unicast a voice stream to a master node and receive a multicast voice stream from the master node. The multicast voice stream may be a mixed voice stream mixing the voice streams of the participants. In this case, each participant performs echo cancellation on the multicast voice stream. Alternatively, the multicast voice stream may be a multiplexed voice stream multiplexing the voice streams of all participants. In this case, each participant de-multiplexes the multicast voice stream and generates a mixed voice stream that does not include that participant. Alternatively, the multicast voice stream received by each participant may be a multiplexed digital voice stream multiplexing multiple mixed streams including one mixed stream that does not include the voice stream of that participant. In this case, each participant de-multiplexes the multicast voice stream to obtain the mixed stream that does not include that participant.

A system includes participant nodes that unicast a voice stream to a master node and receive a multicast voice stream from the master node. The multicast voice stream may be a mixed voice stream mixing the voice streams of the participants. In this case, each participant performs echo cancellation on the multicast voice stream. Alternatively, the multicast voice stream may be a multiplexed voice stream multiplexing the voice streams of all participants. In this case, each participant de-multiplexes the multicast voice stream and generates a mixed voice stream that does not include that participant. Alternatively, the multicast voice stream received by each participant may be a multiplexed digital voice stream multiplexing multiple mixed streams including one mixed stream that does not include the voice stream of that participant. In this case, each participant de-multiplexes the multicast voice stream to obtain the mixed stream that does not include that participant.

Systems, methods, and computer program products for provisioning a temporary disposable number are described. A user can be provided with a pool of available temporary disposable numbers that have a limited shelf life. The user can select one of the available temporary disposable numbers while submitting a permanent phone number associated with a communications device (e.g., mobile phone, home phone, business phone, etc.). Prior to activating the selected temporary disposable number, the temporary disposable number is linked to the permanent phone number. After activation, when an incoming call to the temporary disposable number is received, the permanent phone number is identified to be associated with the temporary disposable number being called. The incoming call is then forwarded to the communications device on which the permanent phone number is established.

Acoustic echo cancelling includes receiving a source signal and a sink signal; providing a first error signal representative of an echo-free residual signal based on a first set of coefficients based on the source signal and the sink signal, the first error signal forming an output signal of the controller; providing a second error signal based on a second set of coefficients based on the source signal and the sink signal; detecting a room change if the evaluated first second error signal is greater than a sum or product of the evaluated second first error signal and a first threshold; copying one of sets of reference coefficients stored in a memory to the second acoustic echo canceller; and copying the first set of coefficients from the first acoustic echo canceller as a set of reference coefficients into at least one of the second acoustic echo canceller and the memory.

An apparatus and/or method discloses a video conference with enhanced audio quality using high-fidelity audio sharing (HAS). In one embodiment, a network connection between a first user equipment (UE) and a second UE is established via a communication network for providing an interactive real-time meeting. After sending a first calibration audio signal from the first UE to the second UE, a second calibration audio signal is retuned from the second UE to the first UE according to the first calibration audio signal. Upon identifying a far end audio (FEA) delay based on the first calibration audio signal and the second calibration audio signal, a first mixed audio data containing the first shared audio data and first FEA data is fetched from an audio buffer. The first FEA data is subsequently removed or extracted from the mixed audio data in response to the FEA delay.

One embodiment provides a method, including: identifying, using an information handling device, a proximate sound source; determining, using a processor, that noise produced by the proximate sound source is duplicated on a conference bridge that a user is connected to and is within an audible range of the user; and preventing, based on the determining, the noise on the conference bridge from reaching the user. Other aspects are described and claimed.

A method for initialization of a group of customer premises equipment devices (CPEs) during a training that registers capabilities of the CPEs is disclosed, wherein at least one CPE registers late to the training and cannot be registered. The method includes determining capabilities of the CPEs during a joining phase of the training, wherein it is determined whether a CPE device is capable of employing vectoring, and placing in a hold status the at least one CPE that registers late by keeping a line active that is coupled to the at least one CPE. The method further includes providing another joining phase after the joining phase to register the at least one CPE that registers late.

Systems, methods, and devices are disclosed for detecting an active speaker in a two-way conference. Real time audio in one or more sub band domains are analyzed according to an echo cancellor model. Based on the analyzed real time audio, one or more audio metrics are determined from output from an acoustic echo cancellation linear filter. The one or more audio metrics are weighted based on a priority, and a speaker status is determined based on the weighted one or more audio metrics being analyzed according to an active speaker detection model. For an active speaker status, one or more residual echo or noise is removed from the real time audio based on the one or more audio metrics.

In accordance with an aspect of the disclosure, an electronic device comprises a communication circuitry configured to establish a voice call with an external electronic device; a microphone; a memory configured to store a first sound quality enhancement parameter; and a processor, wherein the processor is configured to: obtain an audio signal associated with speech through the microphone, during the voice call; transmit, to a server, voice data based on the audio signal when the ratio is within a first range; transmit, to the server, noise data based on the audio signal, when the ratio is within a second range; receive an updated sound quality enhancement parameter from the server with the communication circuit during the voice call; and adjust the first sound quality enhancement parameter stored in the memory, based on the updated sound quality enhancement parameter received from the server.

A voice call method is provided. The method includes: obtaining a voice call state of a terminal system at a historical moment, the terminal system being provided with at least two audio acquisition devices; obtaining first voice signals acquired by the at least two audio acquisition devices at a current moment, and determining signal energy of the first voice signals; and determining a target audio acquisition device at the current moment from the at least two audio acquisition devices based on the voice call state at the historical moment and the signal energy of the first voice signals.