Abstract: A method and apparatus are disclosed that seek to improve the quality of service that is experienced during the transmission of a stream of packets across one or more paths. In particular, a transmitting node encodes a source stream of data (e.g., audio, video, etc.) into one or more sub-streams, and distributes those sub-streams onto multiple network transmission paths. In accordance with the illustrative embodiment of the present invention, the transmitting node evaluates the quality of service of a first network path that fails to provide a quality-of-service guarantee. When the quality of service of the first network path becomes unsatisfactory, the coding of one or more sub-streams that are being transmitted on a second network path is adjusted. In other words, the coding on a second channel is adjusted in response to the changing conditions on a first channel.

Abstract: A method is disclosed for acoustic feedback attenuation at a telecommunications terminal. A speakerphone equipped with a loudspeaker and two microphones is featured. Signals from the two microphones are subjected to a calibration stage and then to a runtime stage. The purpose of the calibration stage is to match the microphones to each other by advantageously using both magnitude and phase equalization across the frequency spectrum of the microphones. During the runtime stage, the microphones monitor the ambient sounds received from sound sources, such as the speakerphone's users and the loudspeaker itself, during a conference call. The speakerphone applies the generated set of filter coefficients to the optimized microphone's signals. By combining the signal from the reference microphone with the filtered signal from the optimized microphone, the speakerphone is able to attenuate the sounds from the loudspeaker that would otherwise be transmitted back to other conference call participants.

Abstract: A call-center has agents using headsets, which are connected to a private business exchange (PBX). Noise cancellation occurs in a noise filter, within or connected to the PBX. Noise cancellation for a particular agent's conversation is achieved by receiving the voice signals from neighboring agents' headsets and by using adaptive noise cancellation in the noise filter to remove the other agents' conversations from the particular agent's conversation. Microphones may also be placed at other noise sources, such as HVAC equipment, so that offending noises are accurately received at the noise filter and removed from the agents' conversations.

Abstract: A method is disclosed that enables the monitoring, evaluation, and adjustment of a telecommunications network's audio-signal loss plan. The method can be implemented at a data-collection server, in which the server accumulates voice-quality measurement statistics from various nodes in the network. Such nodes include telecommunications endpoints, media gateways, private-branch exchanges, teleconference bridges, and so forth. The different types of statistics that can be acquired include voice activity detection, average speech level, average noise level, and so forth. The server accumulates the statistical data from the various nodes for multiple calls and over an extended period of time. The server is also able to compare the statistics against a theoretical model that is a function of the loss plan, at least in part. For example, the comparisons that the data-collection server performs can be used to determine why certain calls have been reported as having unsatisfactory quality.

Abstract: A method is disclosed that enables the managing of the overall sound level in an enterprise environment where telephones are used. A data-processing system such as a private branch exchange monitors whether one or more telephones are in use. Based on detecting when a first endpoint is in use and, therefore, when the endpoint's user is present, the private branch exchange controls one or more characteristics of the loudspeaker volume at a second endpoint. By accounting for other considerations such as the spatial closeness between the endpoints, which can be determined from office dimensions stored in a database, the private branch exchange of the illustrative embodiment is able to determine the degree of sound that is coupling over from one endpoint location to another. On a larger scale, the exchange is able to control the loudspeaker volumes of all of the endpoints in the workplace area. In doing so, the exchange manages the overall acoustic noise present.

Abstract: A technique is disclosed that enables the managing of environmental conditions within an enterprise workplace and, in doing so, provides an improvement in facilities cost management over some techniques in the prior art. A data-processing system such as a private-branch exchange monitors the workplace by using one or more telephones, or other “telecommunications endpoints” to which the exchange is connected, in the workplace area. The exchange determines whether people are present in the workplace area by monitoring which endpoints are in use. Additionally, the exchange monitors the sounds that are received by the microphones of the endpoints. Based on knowing which endpoints are in use, the exchange generates control signals for the purpose of controlling one or more environmental conditions such as temperature, lighting, and so forth. In some embodiments of the present invention, the exchange examines the audio content of the received signals and bases the control signals on the audio content analyzed.

Abstract: A user of a telecommunications terminal is made aware when the user's voice is disrupted by echo suppression, which notifies the user that he or she can adjust his or her speaking pattern, or move closer to the microphone in order to stop the clipping.

Abstract: A method enables a participant in a conference call to monitor, as he is speaking, whether his speech is getting through to the other participants. A teleconference bridge receives audio signals from a group of telecommunications endpoints that are involved in a conference call. The bridge generates audio signals to be transmitted, which are based on one or more of the received audio signals. During the ongoing process of minimizing the presence of acoustic echo, the bridge might exclude one or more of the received audio signals from the transmitted audio signals. When this occurs, particularly when an active talker is being excluded, the bridge transmits an indication to one or more of the endpoints as part of one of the transmitted audio signals. The indication can be audible such as a tone or a voice, visual such as a flashing light, or tactile such as vibration.

Abstract: A technique is disclosed that optimizes the background noise experienced by a party who is calling into a call center. Working as part of an overall call-assignment algorithm, the technique considers the acoustic noise that is present in the vicinities of multiple call-center agents who are otherwise satisfactory candidates to handle a call. The technique then selects an agent to handle the call who is associated with an optimal acoustic noise. Typically, the selected agent is associated with lowest background noise level. The background noise is monitored at each call agent's station by evaluating the signals that are present at the agent's microphone. Usually, this is done when a call agent is between calls and, as a result, is not using her headset at that moment. In other words, the background noise is actually measured, and the measurements are then used to assign a call agent to the incoming call.

Abstract: A technique applicable to a telecommunications system is disclosed that provides quality-related information in a receive path, in which the information serves as feedback about the call quality that is present in the corresponding transmit path. It is recognized that when a first party on a call hears degradation, he tends to change the way that he speaks or reacts to a second party. Based upon this recognition, the present invention addresses the problem of one-way degradation by providing a feedback signal to the first party, in the form of intentional degradation of the voice signals that he is receiving from the second party. The degradation that is introduced by the disclosed data-processing system is based on the current quality of the voice path from the first party to the second party (i.e., the transmit path), instead of the other way around (i.e., the receive path).

Abstract: A method is disclosed to determine the presence of one or more cellular phones in the same sound field as an endpoint that is dedicated for teleconferencing, such as a speakerphone. The teleconference bridge of the illustrative embodiment continually receives geo-location information about the cell phone. Based on the geo-location of the cell phone relative to the position of the speakerphone, the bridge determines whether to include or exclude signals that are received from the cell phone when preparing a signal for transmission to the speakerphone during a conference call. The bridge also determines whether to refrain from transmitting an audio signal to the cell phone, when the bridge infers that the cell phone is being used as a satellite microphone, such as when the cell phone is placed on a conference room table. As a result, each conference call participant is able to use his or her own cell phone as a personal satellite microphone, which can improve the sound quality of the conference call.

Abstract: A method is disclosed that breaks the “one line, one location” paradigm of teleconferencing in the prior art. The teleconference bridge in the illustrative embodiment is able to utilize more than one audio channel from each location. By having access to more than one audio channel from a teleconference location's sound field, the bridge is able to create a multi-channel effect during a conference call. When more than one microphone is present in the first sound field, they can be used to create a multi-channel effect in a second or other sound field involved in a conference call that has more than one loudspeaker. With the bridge mimicking the multi-channel imaging in one room with the multi-channel imaging in another, the conference call participants can experience audio depth during a conference call and can experience two-dimensional imaging, depending on the microphone or speaker separation that is present.

Abstract: A method is disclosed that breaks the “one line, one location” paradigm of teleconferencing in the prior art. The teleconference bridge in the illustrative embodiment is able to utilize more than one audio channel from each location, where there are multiple signal sources present in the room. As a result, the bridge is able to determine acoustically whether two are more endpoints are collocated with each another. During an initialization sequence, the bridge transmits special audio signals to one or more endpoints present in a particular sound field; those endpoints then play the signals out of their loudspeakers. Based on a characteristic (e.g., amount of correlation, signal strength, etc.) of the signals received at each microphone present in the same sound field, the bridge determines whether to include or exclude signals that are received from a first endpoint when preparing a signal for transmission to a second endpoint during a conference call.

Abstract: A user of a telecommunications terminal is made aware when the user's voice is disrupted by echo suppression, which notifies the user that he or she can adjust his or her speaking pattern, or move closer to the microphone in order to stop the clipping.

Abstract: The present invention utilizes pseudo-stereo for the communication of secondary information to the user of a telecommunications terminal, such as a speakerphone. In particular, the terminal utilizes a method for the presentation of secondary information to the user of the terminal in a teleconference call by adjusting the spatial properties of the monaural audio received at the user's terminal. An audio communication is modified so as to appear that the communicated audio is arriving from a particular direction in relation to the user's approximate position, wherein the direction that is assigned to the audio depends on one or more characteristics of the call participant who is originating the audio. Each characteristic of a call participant on a call can comprise, while not being limited to, the customer satisfaction of the call participant, the urgency of a need of the call participant, the group membership of the call participant, or the product ownership of the call participant.

Abstract: An apparatus and method are disclosed that enable a first telecommunications endpoint to ensure that a second endpoint is receiving the first endpoint's packet stream transmissions with a satisfactory waveform quality. When the second endpoint receives the packet stream, it decodes the media waveform from the stream, encodes the waveform back into a second packet stream, and transmits some or all of the packets in the second stream back to the first endpoint. The first endpoint then decodes the received waveform in the second stream and compares it to the original waveform transmitted to the second endpoint. Based on the comparison, the first endpoint adjusts the value of a quality indication, and provides the quality indication to its user and to the second endpoint. Advantageously, the user at the second endpoint is able to determine whether the received waveform is, in fact, close enough to the waveform that the first endpoint's user intended to be received and understood.

Abstract: A method is disclosed for acoustic feedback attenuation at a telecommunications terminal. A speakerphone equipped with a loudspeaker and two microphones is featured. Signals from the two microphones are subjected to a calibration stage and then to a runtime stage. The purpose of the calibration stage is to match the microphones to each other by advantageously using both magnitude and phase equalization across the frequency spectrum of the microphones. During the runtime stage, the microphones monitor the ambient sounds received from sound sources, such as the speakerphone's users and the loudspeaker itself, during a conference call. The speakerphone applies the generated set of filter coefficients to the optimized microphone's signals. By combining the signal from the reference microphone with the filtered signal from the optimized microphone, the speakerphone is able to attenuate the sounds from the loudspeaker that would otherwise be transmitted back to other conference call participants.

Abstract: Methods and apparatus are provided for reducing echo from a received signal. A suppression gain is applied to an output of an echo canceler that has processed the received signal. The suppression gain includes a region of sloping attenuation about a decision point. The echo canceler optionally estimates an echo path and subtracts then estimate from the received signal. The suppression gain includes a non-zero lower bound, gmin, on a maximum attenuation applied by a suppressor, that is based on operating conditions of the echo canceler. The region of sloping attenuation applies a variable amount of attenuation that depends on the size of an output error ?(n) of the echo canceler relative to the received signal, x(n).

Abstract: A method is disclosed that enables the monitoring, evaluation, and adjustment of a telecommunications network's audio-signal loss plan. The method can be implemented at a data-collection server, in which the server accumulates voice-quality measurement statistics from various nodes in the network. Such nodes include telecommunications endpoints, media gateways, private-branch exchanges, teleconference bridges, and so forth. The different types of statistics that can be acquired include voice activity detection, average speech level, average noise level, and so forth. The server accumulates the statistical data from the various nodes for multiple calls and over an extended period of time. The server is also able to compare the statistics against a theoretical model that is a function of the loss plan, at least in part. For example, the comparisons that the data-collection server performs can be used to determine why certain calls have been reported as having unsatisfactory quality.

Abstract: A method is disclosed that enables a participant in a conference call to monitor, as he is speaking, whether his speech is getting through to the other participants. A teleconference bridge receives audio signals from a group of telecommunications endpoints that are involved in a conference call. The bridge generates audio signals to be transmitted, which are based on one or more of the received audio signals. During the ongoing process of minimizing the presence of acoustic echo, the bridge might exclude one or more of the received audio signals from the transmitted audio signals. When this occurs, particularly when an active talker is being excluded, the bridge transmits an indication to one or more of the endpoints as part of one of the transmitted audio signals. The indication can be audible such as a tone or a voice, visual such as a flashing light, or tactile such as vibration.