A processing load at calling is reduced. An information processing apparatus including: a management unit configured to manage a control parameter related to call quality of a call previously made with a user terminal; and a transmission unit configured to relay packets of call voice and transmit the packets to an edge apparatus when a call is made with the user terminal, the edge apparatus being configured to perform processing based on the control parameter is provided.

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.

A conference system includes a conversation state determiner that determines whether or not the state of first and second users is a direct conversation state in which direct conversation is possible without using a speech system, and an output controller that controls whether or not to cause the speech system to output a first acquired voice from a second speaker, based on the determination result of the conversation state determiner.

A double-talk state detection method includes: calculating an energy ratio between a first energy of an error signal in each sub-band of M sub-bands and a second energy of a filtered signal in the same sub-band as the error signal, thereby obtaining M energy ratios, where the error signal is a difference between an input signal collected by a microphone and the filtered signal, the filtered signal is a signal obtained after performing filtering process on a reference signal, and M is a positive integer; performing a first smoothing processing on the M energy ratios to obtain M first energy smoothing ratios, and performing a second smoothing processing on the M first energy smoothing ratios to obtain M second energy smoothing ratios; performing double-talk state detection based on the M first energy smoothing ratios and the M second energy smoothing ratios to determine a state of the input signal.

To improve accuracy of an evaluation in an acoustic quality evaluation test performed by comparing an evaluation target sound and a reference sound. A data correction apparatus 3 compares, in a call performed between a near-end terminal 1 and a far-end terminal 2, an evaluation target sound in which a voice output from the near-end terminal 1 is recorded and a reference sound in which a voice spoken by a call partner using the far-end terminal 2 to correct test data used in a listening test for evaluating acoustic quality of the call. A correction target determination unit 31 determines, as a correction target section, a voiced section that does not include the voice of the call partner detected from an acoustic signal representing the reference sound. A correction execution unit 32 updates the correction target section of the acoustic signal representing the reference sound with a non-voice signal predetermined.

A method for providing telecommunications on a videoconferencing system can include: a telecommunication endpoint device connecting, via the internet, to a server in a telecommunication network, the server being configured to provide a multi-tenant service; associating the telecommunication endpoint device with a tenant in the multi-tenant service; at least one peripheral device connecting to the server via the internet; associating the at least one peripheral device with the tenant; and providing telecommunications with the telecommunication endpoint device, wherein the telecommunications comprise telecommunication data and at least a portion of the telecommunication data is provided by the at least one peripheral device.

A method and computer program product are provided for allowing a plurality of musicians who are in physically separate locations to create a single musical performance using a teleconferencing platform provided by a host server. The teleconferencing platform is electronically connected to musician participants who create the single musical performance via respective musician participant computers, and non-musician participants who experience the created single musical performance via respective non-musician participant computers. The single musical performance includes voice and/or instrument sounds contributed from each of the musicians. Host-sent audio is streamed by the host server to each of the participant computers. The host-sent audio includes a separable tempo and tone reference. Each of the musician participants provide their respective voice and/or instrument sounds to the host-sent audio. The separable tempo and tone reference is used to synchronize the single musical performances of the respective musician participants which are then combined to create the single musical performance. The host-sent audio, including the separable tempo and tone reference, is removed from the single combined musical performance, leaving only the single musical performance which may then be provided to the non-musician participants via their respective non-musician participant computers.

First and second computer systems and respective first and second microphones thereof receive respective portions of a same audio input signal. Audio buffers received respectively from the first and second computer system include data encoded from respective microphone inputs of the first and the second computer systems. The received audio buffers are synchronized and corrected for gain differences between the received audio buffers to produce corrected audio buffers. The corrected audio buffers are mixed into an output buffer. The synchronization reduces echo when the output buffer is played at a remote peer computer system.

A method and system for improving communications conferencing systems that experience continuous double-talk where the communication includes an intended continuous or intermittent soundtrack or other intended continuous sound. The technology as disclosed and claimed herein uses several techniques to mask the residual echo and make it less audible.

Method, apparatus, and program code embodied in computer-readable media, for providing enhanced echo suppression in a conferencing system having at least one microphone and at least one speaker. At least one microphone input signal is received, and at least one speaker input signal is provided. At least one processor has at least one primary echo-suppressor and at least one secondary echo-suppressor. The at least one primary echo-suppressor receives (i) the microphone input signal(s) and (ii) the speaker input signal(s). The at least one primary echo-suppressor provides at least one echo-suppressed microphone signal. The at least one secondary echo-suppressor receives the at least one echo-suppressed microphone signal and provides an output signal. The at least one processor provides the at least one echo-suppressed microphone signal to the at least one secondary echo-suppressor without providing the at least one speaker input signal directly to the at least one secondary echo-suppressor.