Abstract: The present invention provides methods, devices, and systems that provide the ability to playback multiple VoIP audio streams simultaneously with a single RTP session and further provides the ability to conference all streams together prior to transmission. More specifically, the present invention focuses on the transport layer and sample-based multiplexing employed to provide this capability.

Abstract: A method and apparatus prevent eavesdropping via a computer by detecting and alerting if more than one authorized driver is controlling a interface circuit that is providing audio or video input information. Further, prevention is performed by detecting and alerting if more than one authorized application programming interface is receiving audio or video input information from an authorized driver. Also, prevention is performed by detecting and alerting if more than one authorized software application is receiving audio or video input information from an authorized driver. In addition, prevention is performed by detecting and alerting upon first receipt of audio or visual information by an authorized software application via an authorized application programming interface and authorized driver.

Abstract: The present invention provides a method, device, and system for taking voice memos during a communication session. More specifically, voice memos may be taken using the same communication device that is used in the communication session. This can be accomplished by automatically muting the communication device and establishing a communication channel with another endpoint, such as a recording endpoint, to record audio signals transmitted by the communication device.

Abstract: The present invention provides methods, devices, and systems that provide the ability to playback multiple VoIP audio streams simultaneously with a single RTP session and further provides the ability to conference all streams together prior to transmission. More specifically, the present invention focuses on the transport layer and sample-based multiplexing employed to provide this capability.

Abstract: The present invention provides methods, devices, and systems that provide the ability to playback multiple VoIP audio streams simultaneously with a single RTP session and further provides the ability to conference all streams together prior to transmission. More specifically, the present invention focuses on the transport layer and sample-based multiplexing employed to provide this capability.

Abstract: The present invention provides methods and systems for managing bandwidth use in a VoIP network. More specifically, a user or enterprise may monitor the amount of voice bandwidth that is utilized throughout a period along with the total amount of bandwidth utilization. Based on a determined likelihood of exceeding a predetermined bandwidth allocation for a given period, the amount of voice bandwidth utilized on a per call basis is adjusted.

Abstract: A voice communications terminal is provided that includes: a microphone 208 operable to receive an audio stream; and a voice activity detector 220 operable to (a) determine whether the audio stream comprises voice signals and (b) when the audio stream comprises voice signals, at least one of (i) notify a communications server 110 and (ii) provide at least a portion of the audio stream to the communications server 110.

Abstract: An encoder and associated vector estimation method and system (1) for processing a sequence of input vectors (y0 to yT) each comprising a plurality of elements. The vector estimation system (1) has a digital filter (2) with a filter vector input (3) for receiving said sequence of input vectors (y0 to yT) and a predictor gain input (4) for controlling characteristics of the filter (2). The filter (2) is a Kalman filter and has both a current slowly evolving filter estimate output (6) and a previous slowly evolving filter estimate output (20). The current slowly evolving filter estimate output (6) provides a current filtered estimate value of a slowly evolving component of said sequence of input vectors (y0 to yT) and the previous slowly evolving filter estimate output (20) provides a previous filtered estimate value of the slowly evolving component of said sequence of input vectors (y0 to yT).

Abstract: A gain control method for reducing or eliminating the undesirable effects of acoustic echo coupling between a speaker and a microphone while providing a full-duplex connection. Weighted normalized far-end and near-end powers are used to first calculate a suppression value which is used to determine the attenuation factor between an upper and a lower limit, thus improving reliability in noisy environments. To further improve the quality of the full-duplex connection, a smoothing factor is applied to the attenuation calculation to provide a low power and constant sounding echo without annoying transient-like sounds.

Abstract: An encoder and associated vector estimation method and system (1) for processing a sequence of input vectors (y0 to yT) each comprising a plurality of elements. The vector estimation system (1) has a digital filter (2) with a filter vector input (3) for receiving said sequence of input vectors (y0 to yT) and a predictor gain input (4) for controlling characteristics of the filter (2). The filter (2) is a Kalman filter and has both a current slowly evolving filter estimate output (6) and a previous slowly evolving filter estimate output (20). The current slowly evolving filter estimate output (6) provides a current filtered estimate value of a slowly evolving component of said sequence of input vectors (y0 to yT) and the previous slowly evolving filter estimate output (20) provides a previous filtered estimate value of the slowly evolving component of said sequence of input vectors (y0 to yT).

Abstract: A gain control method for reducing or eliminating the undesirable effects of acoustic echo coupling between a speaker and a microphone while providing a full-duplex connection. Weighted normalized far-end and near-end powers are used to first calculate a suppression value which is used to determine the attenuation factor between an upper and a lower limit, thus improving reliability in noisy environments. To further improve the quality of the full-duplex connection, a smoothing factor is applied to the attenuation calculation to provide a low power and constant sounding echo without annoying transient-like sounds.