Avaya Inc.

Nortel Networks Corporation

Northern Telecom Limited

Avaya Management L.p.

John Cornelius Lynch

Chris McArthur

Peter Tarle

Marc Saunders

Paul Haig

Paul Roller Michaelis

Parameshwaran Krishnan

Jon Louis Bentley

Andre M Cyr

Brian Pick

Miguel Dearaujo

Glenn Eng

Heinz Teutsch

Rifaat Shekh-Yusef

Mark Fletcher

Mohamed Samy El-Hennawey

David Ward

Allan Lesperance

Pierre Couillaud

Michael Pothier

Anthony Peter Neville Brown

Tod Allen Rikley

Metodej Valentik

Chee Fai Lee

Nina Yunusova

Rayan Ahmed Mamun

Dustin John Laidsaar

David Cairns

Metodej E Valentik

Tracy Monk

Christopher Bruce Mcarthur

The technology disclosed herein enables ambient noise level determination at an endpoint using a secondary audio capture system for the endpoint. In a particular embodiment, a method includes during a communication session, receiving an audio signal representing sound captured by a secondary audio capture system for an endpoint on the communication session. A primary audio capture system for the endpoint captures voice communications from a user of the endpoint for the communication session. The method further determining an ambient noise level for the endpoint from the audio signal and, in response to determining that the ambient noise level satisfies a loudness criterion, presenting a notification to the user about the ambient noise level.

Ambient noise detection using a secondary audio receiver

Embodiments disclosed herein provide systems, methods, and computer readable media for synchronizing a media codec between network elements of a media communication session. In a particular embodiment, a method provides designating a first network element to be a static-clock network element during the media communication session and designating at least a second network element to be a dynamic-clock network element during the media communication session. The method further provides determining that a difference in clock speed exists between a second clock speed for the media codec at the second network element and a first clock speed for the media codec at the first network element. Also, the method provides adjusting the second clock speed to account for the difference in clock speed.

Synchronization of a media codec between network elements of a media communication session

Workplace safety is a principal concern in many environments. Protecting user ears from damage due to extended exposure to unacceptably high sound volume serves as an important component to workplace safety. Monitoring a device, such as a phone, utilized by a user often provides an incomplete picture of the sound level presented to the user. As provided herein, monitoring a user's sound exposure on one device may cause the sound level presented to the user from a second device to become limited. Additionally, over time the sound level limits may be adjusted based on the cumulative historic sound exposure. As a result, the user may avoid exposure to unacceptably high sound levels originating from more than one source and/or over an extended period of time.

Dynamic time-weighted systems and methods for management of acoustic exposure

The methods and systems of the present disclosure can set at least one threshold for properties of at least one sound, where the at least one threshold comprises a first threshold, and where the at least one sound comprises a first sound received at a first device; monitor changes in properties of the first sound over time at the first device; detect a first violation of the first threshold for the first sound; and provide a first notification based on the detection of the first violation.

Method and system for warning users of offensive behavior

Embodiments disclosed herein provide systems, methods, and computer readable media for probing alarm functionality at a premises. In a particular embodiment, a method provides, in response to activation of an audible alarm, obtaining audio information captured by a first communication device of a plurality of communication devices positioned throughout the premises. The method further provides determining whether the audio information indicates that the alarm is functioning properly and determining a first position of the premises where the first communication device is located. When the audio information indicates that the alarm is not functioning properly, the method provides indicating that the alarm is not functioning properly at the first position.

Probe of alarm functionality using communication devices

Embodiments disclosed herein provide systems, methods, and computer readable media for managing bandwidth based on audio power level. In a particular embodiment, a method provides, from a conference system, transferring at least one audio level criterion to a first user system in a communication session. At the conference system, when a level of first audio for the communication session satisfies the at least one audio level criterion at the first user system, the method provides receiving the first audio encoded using first audio codec parameters by the first user system. At the conference system, when the level of the first audio does not satisfy the at least one audio level criterion at the first user system, the method provides receiving the first audio encoded using second audio codec parameters by the first user system.

Audio level based management of communication resources

Embodiments disclosed herein provide systems, methods, and computer readable media for managing bandwidth based on audio power level. In a particular embodiment, a method includes receiving a media stream from a first client device, wherein the media stream includes an audio stream. During receipt of the media stream, the method provides determining whether an audio power level of the audio stream satisfies criteria for inclusion in an output media stream from the media server system along with one or more media streams received from one or more other client devices. Upon determining that the criteria is not satisfied, the method provides transferring a first notification to the first client device that includes a first instruction for the first client device to reduce an amount of bandwidth used for transferring the media stream.

Audio power based media bandwidth management

An indicator of a channel capacity of a wireless link between a wired telephone and a wireless hands free device is measured. Based on the indicator of the channel capacity, a call server decides whether to route a call to the wired telephone. The indicator of channel capacity may be is based on measurements of channel capacity between the telephone and the wireless hands free device taken at different times and at different locations. The measurements of channel capacity may be used to construct a map relating physical location and time to channel capacity available between one or more wired telephones and wireless hands free devices. The available channel capacity can also be used to cause indicators associated with low channel capacity to be displayed by the wired telephone or signaled to the wireless hands free device.

Wireless enterprise congestion management

Systems and methods for applying user specific acoustic adjustment parameters are provided. The intelligibility of speech for a particular user is determined and a set of acoustic adjustment parameters is determined. The set or template of acoustic adjustment parameters for the user is placed in central store, for example provided as or in association with a server. The template can be obtained from the server for application in connection with a communication involving the user by providing an identification of the template.

Telecommunications methods and systems providing user specific audio optimization

Systems and methods for automatic user specific, condition specific communication system intelligibility testing and optimization are provided. The intelligibility of speech for a particular user is determined using a test of intelligibility administered by an interactive voice response (IVR) application running on a communication server. The intelligibility test can be run for a particular user under different conditions. For each user and/or set of conditions, a set of speech signal adjustment parameters can be determined. A set of speech signal adjustment parameters that will enhance the intelligibility of a speech signal for a user are applied when that user is involved in a communication session. The particular set of speech signal adjustment parameters selected can depend on the communication equipment and/or environment associated with the communication session.

Method and apparatus for automatic communications system intelligibility testing and optimization

System and method to improve intelligibility of coded speech, the method including: receiving an encoded speech signal from a network; extracting an encoded media data stream and one or more control data packets from the encoded speech signal; decoding the encoded media data stream to produce a decoded speech signal; boosting an upper spectral portion of the decoded speech signal to produce a boosted speech signal; and outputting the boosted speech signal. In another embodiment, the method may include: receiving an uncoded speech signal; processing the uncoded speech signal, wherein the processing comprises generating an unencoded data stream from the uncoded speech signal; boosting an upper spectral portion of the unencoded data stream to produce a boosted speech signal; encoding the boosted speech signal to produce an encoded speech signal; and outputting the boosted speech signal.

System and method for method for improving speech intelligibility of voice calls using common speech codecs

A method of processing a frame received at a networked device having a port switch and a general-purpose processor. The method can include receiving frame information at the port switch, determining at least one port for the frame, and directing the received frame information based on the determined port(s).

Port switch

In a packet-based telephony network, a media gateway provides information on active media path connections to an active call server. For instance, this data may be provided to a backup call server to keep the backup call server in a warm standby mode. The backup call server polls a media gateway to retrieve a table of information on active media path connections. The polling may be done periodically or only during switchover from the active call server to the backup call server. After switchover, the backup call server has all necessary details about active media path connections to provide accurate billing information. Alternatively, after a temporary failure at the active call server, the active call server may send a request to the media gateway to retrieve a table of information on active media path connections.

Media gateway connection information recovery

A given DSP, when processing signals of a channel, stores channel instance data in a multiplexed memory in association with itself. If the channel processing task changes to one for which the given DSP is not optimised, the given DSP moves the channel to an appropriate new DSP. Additionally, the given DSP transfers the association of the channel instance data to an association with the new DSP. In this way, the new DSP is provided with the channel instance data without the requirement for physically transferring such data.

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Belleville, Canada

John Cornelius Lynch