Patent application number | Description | Published |
20080298349 | SYSTEM FOR TRANSMITTING HIGH QUALITY SPEECH SIGNALS ON A VOICE OVER INTERNET PROTOCOL NETWORK - The VoIP quality speech process is activated when a subscriber accesses a speech quality sensitive resource or in response to an activation of the feature by the subscriber, or when it is determined that the originating subscriber terminal device requires the transmission of high quality speech signals. A transmit buffer, associated with the port circuit that serves the originating device, stores a predetermined number of packets as they are transmitted from the originating device. In the case of lost or damaged packets, the VoIP quality speech system activates the transmit buffer to retransmit the missing or damaged packet to the destination device. Intelligent buffer management is provided, where the destination device can regulate the size of the transmit buffer as well as the size of its jitter buffer. | 12-04-2008 |
20090060170 | Method and apparatus for call control using motion and position information - A method and apparatus perform call control by obtaining initial biometric information of ears of a user of a handheld audio device; detecting a change in the biometric information of the ears of the user; determining a position and motion of the handheld audio device by analyzing the detected change in biometric information; and performing call control based on the determined position and motion. Further, the method and apparatus perform a first call control operation in response to the determined motion and position if the handheld audio device is communicating with a first endpoint and a second call control operation if communicating with a second endpoint. In addition, the method and apparatus change an internal call control operation of a handheld audio device by determining motion and a position of the handheld audio device. Also, the method and apparatus control internal operations of an endpoint by determining motion and position of the handheld audio device without terminating communication with the handheld audio device wherein the endpoint is one of a voice messaging system, conferencing system, or telephone operator. | 03-05-2009 |
20090060240 | Method and apparatus for configuring a handheld audio device using ear biometrics - A method and apparatus configure a handheld audio device to communicate audio information by identifying an ear being used with the handheld audio device and by configuring the handheld audio device in response to the ear identification to communicate audio information to the ear. The identification may use sonic or visual techniques to identify the ear. | 03-05-2009 |
20090060243 | Method and apparatus for communicating to a hearing aid using an aimed electro-magnetic field - A method and apparatus configure an audio device to communicate audio information to a hearing aid by receiving location information defining a location of a receiving inductive coil of the hearing aid; aiming a transmitting inductive coil of the audio device at the location; and controlling the transmitting inductive coil to generate an electro-magnetic field to communicate the audio information. | 03-05-2009 |
20090061819 | Method and apparatus for controlling access and presence information using ear biometrics - A method and apparatus control the operations of an audio device by obtaining by the audio device biometric information about an ear of a user; identifying the user by processing the biometric information; and controlling the operation of the audio device in response to the identification of the user. Also, a method and apparatus control the operations of an external database by receiving biometric information about the ear of a user from an audio device; identifying the user by processing the biometric information; and transmitting a control message to another system. Further, a method and apparatus control the operations of a telecommunication system by receiving biometric information about an ear of a user from a telecommunication set; identifying the user by processing the biometric information; and controlling the operation of the telecommunication system in response to the identification of the user. | 03-05-2009 |
20090204677 | CONTEXT BASED FILTER METHOD AND APPARATUS - A context sensitive filter method and apparatus is provided. In particular, information regarding the context in which a request for content is made is gathered, and is used to select filter parameters for application to content returned in response to the request. Context information can include information from a calendar application, location information, user preferences or other inputs. | 08-13-2009 |
20090248421 | Arrangement for Creating and Using a Phonetic-Alphabet Representation of a Name of a Party to a Call - A first party creates and edits a phonetic-alphabet representation of its name. The phonetic representation is conveyed to a second party as “caller-identification” information by messages that set up a call between the parties. The phonetic representation of the name is displayed to the second party, converted to speech, and/or converted to an alphabet of a language of the second party and then displayed to the second party. | 10-01-2009 |
20090282228 | Automated Selection of Computer Options - A user of a computer indicates a desired user interface behavior, and the computer automatically selects and sets options of programs and devices of the computer individually for each program to achieve that behavior. Alternatively, the user indicates a condition of the user, such as a specific motor or sensory disability, and the computer automatically adjusts its programs and devices to accommodate the user's needs. | 11-12-2009 |
20090316880 | Purposeful Degradation of Sidetone Audio for Providing Feedback about Transmit-Path Signal Quality - An enhanced sidetone system is disclosed which provides the user of a telecommunications terminal, while speaking, with immediate audio feedback that corresponds to what the far-end party is probably hearing. The sidetone system continuously samples the input speech signal from the user and also obtains signal quality statistics of the transmission path. These statistics can include descriptions of network quality-of-service characteristics (e.g., packet loss rate, etc.) and/or media quality characteristics (e.g., audio distortion due to echo cancellation, etc.). These statistics enable the disclosed technique to determine whether the transmitted signal quality is acceptable. When an unacceptable condition in transmit-path signal quality is detected, the technique modifies the traditional (main) sidetone signal. For example, a delayed sidetone signal can be transmitted back to the user's terminal, in addition to the main sidetone signal generated, so that the user perceives the combination of sidetone signals as a hollow-sounding, objectionable sound. | 12-24-2009 |
20100080374 | METHOD AND APPARATUS FOR IDENTIFYING AND ELIMINATING THE SOURCE OF BACKGROUND NOISE IN MULTI-PARTY TELECONFERENCES - A mechanism is provided that allows participants on the conference call to identify, and then mute or filter, a participant(s) responsible for introducing the noise, regardless of whether the noise is caused by transmission impairments or by the participant(s) being in a noisy location. For example, individual users could be able to press a “test” button that could block each of the participants one at a time. This would allow the source of the noise to be identified. This “test button” could be one or more of provided at the endpoint(s), be enabled through a web interface or, for example, through a dedicated conference call interface at the endpoint(s) or at the conference bridge. The blocking of each participant could occur through interaction with the main PBX using, for example, in-band signaling to the PBX. Once the source(s) of the noise is identified, noise mitigation can be applied as needed. | 04-01-2010 |
20100080375 | System and Method of Managing Conference Calls Through The Use of Filtered Lists of Participants - The system and method establish a conference call between a plurality of communication devices. Each communication device may have one or more participants. The number of participants on each communication device is determined. The system and method get a profile for each participant in the conference call. The profile contains at least one parameter. A filtered list of participants in the conference call is generated based on at least one parameter in the profiles. The filtered list of participants is then presented to various participants in the conference call. | 04-01-2010 |
20100100252 | Centralized Energy Management in Distributed Systems - A distributed system of appliances that comprises a centralized energy-use controller to coordinate the use of energy among a plurality of distributed and interconnected appliances is disclosed. For example, the centralized controller directs when each appliance refrains from using energy from its energy-storage device during intervals when the energy-storage device is capable of providing energy to the appliance but the centralized power supply is not, and when each appliance can consume energy from its energy-storage device during intervals when both the energy-storage device and the centralized power supply are capable of providing power to the appliance. | 04-22-2010 |
20100182930 | PACKET PRIORITIZATION AND ASSOCIATED BANDWIDTH AND BUFFER MANAGEMENT TECHNIQUES FOR AUDIO OVER IP - The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments. | 07-22-2010 |
20100188967 | System and Method for Providing a Replacement Packet - The system generates a first data stream which represents a data signal. The first data stream is encoded via a first encoding technique. The first data stream comprises one or more packets with a duration and timestamp. The data signal is encoded into a second data stream using a different encoding technique with a corresponding packet in the second data stream. A packet in the second data stream has the same duration and timestamp as the corresponding packet in the first data stream. | 07-29-2010 |
20100208728 | Multi-Route Transmission of Packets Within a Network - Two or more packets are generated from the same data stream (e.g., an audio signal) and are sent on a network. When the first of the two or more packets is received, a first optimal route for the first packet is determined, the first packet is sent on the first optimal route, and information about the first packet is stored in a packet list. When another packet is received, the process determines whether it is the second packet of the two or more packets by comparing at least one field in the second packet to the stored information about the first packet in the packet list. If there is a match, a second optimal route for the second packet is determined, and the second packet is sent on the second optimal route. | 08-19-2010 |
20100220172 | Automatic Video Switching for Multimedia Conferencing - After a video conference is established, a video conferencing system receives a video stream from the participants of the video conference. One of the received video streams is transmitted to the conference participants (e.g. the video stream of the person currently speaking). The video conferencing system monitors a second one (or typically all) of the received video streams to determine if a designated video event (e.g. someone raising his hand) has occurred. If the designated video event has occurred, the video conferencing system switches or supplements the currently transmitted video stream to or with the second one of the received video streams that contains the designated video event. The second video stream is then transmitted to participants in the video conference. The switching of the video stream and the switching of the audio stream may be independent. | 09-02-2010 |
20100239077 | MULTIMEDIA COMMUNICATION SESSION COORDINATION ACROSS HETEROGENEOUS TRANSPORT NETWORKS - The present invention, in one embodiment, is directed to the use of a communication pathway traversing a digital telephone network to handle a portion of the signaling traffic associated with a communication method performed over the Internet. | 09-23-2010 |
20100241432 | PROVIDING DESCRIPTIONS OF VISUALLY PRESENTED INFORMATION TO VIDEO TELECONFERENCE PARTICIPANTS WHO ARE NOT VIDEO-ENABLED - Descriptions of visually presented material are provided to one or more conference participants that do not have video capabilities. This presented material could be any one or more of a document, PowerPoint® presentation, spreadsheet, Webex® presentation, whiteboard, chalkboard, interactive whiteboard, description of a flowchart, picture, or in general, any information visually presented at a conference. For this visually presented information, descriptions thereof are assembled and forwarded via one or more of a message, SMS message, whisper channel, text information, non-video channel, MSRP, or the like, to one or more conference participant endpoints. These descriptions of visually presented information, such as a document, spreadsheet, spreadsheet presentation, multi-media presentation, or the like, can be assembled in cooperation with one or more of OCR recognition and text-to-speech conversion, human input, or the like. | 09-23-2010 |
20100253689 | PROVIDING DESCRIPTIONS OF NON-VERBAL COMMUNICATIONS TO VIDEO TELEPHONY PARTICIPANTS WHO ARE NOT VIDEO-ENABLED - The use of detected non-verbal communications cues, and summaries thereof, are used to provide audible, textual and/or graphical input to listeners who for any reason do not have the benefit of being able to see the non-verbal communications cues, or speakers about mannerisms or other non-verbal signals they are sending to other parties. This includes cues that are given while speaking or listening. The detection of one or more of an emotion and gesture could also trigger a dynamic behavior. For example, certain emotions and gestures could be characterized as “key emotions” or “key gestures” and a particular action associated with the detection of one of these “key emotions” or “key gestures.” | 10-07-2010 |
20100260326 | Short Impromptu Communications In Presence-Based Systems - An apparatus and methods are disclosed for enabling certain types of communications to occur, even when presence information might indicate that a particular user is unavailable. In the illustrative embodiment, a first user submits a request to communicate with a second user, where the request specifies a maximum time duration for the communication, and optionally: a minimum time duration, an expected time duration, a priority, a subject, and a type of communication. A presence server receives the request and decides whether the request should be granted based on the information specified in the request, and one or both of: presence information for the second user, and the contents of a calendar. The illustrative embodiment is also capable of detecting inconsistencies between calendars and presence information, as well as events that might affect a user's presence or indicate a departure from scheduled activities. | 10-14-2010 |
20100265834 | VARIABLE LATENCY JITTER BUFFER BASED UPON CONVERSATIONAL DYNAMICS - Methods, devices, and systems for balancing latency and voice quality during a communication session are provided. More specifically, mechanisms for monitoring parameters indicative of conversational dynamics and adjusting jitter buffer size based thereon are described. This allows latency to increase if the conversation is not highly interactive and decrease if a more interactive conversation is desired. | 10-21-2010 |
20100271944 | DYNAMIC BUFFERING AND SYNCHRONIZATION OF RELATED MEDIA STREAMS IN PACKET NETWORKS - The present invention is directed to the use of two or more buffers, at a common receiving node, to reduce the effects of jitter, packet loss, and/or packet latency and/or synchronize different types of packets. | 10-28-2010 |
20100290608 | SYSTEM AND METHOD FOR SENDING DATA USING CALLER ID - A communication system receives data that does not pertain to an attempted initial establishment of a communication. The communication system uses a new message format under an existing analog Caller ID standard. The communication system inserts the data (which can be in addition to existing Caller ID data) into a Caller ID message that uses the new message format. The communication system then sends the Caller ID message to a communication device/Private Branch Exchange (PBX)/contact center that can interpret the new message format. | 11-18-2010 |
20100303219 | METHOD FOR INCLUDING CALLER-PROVIDED SUBJECT INFORMATION IN THE CALLER-ID DISPLAY OF ENTERPRISE TELEPHONES - Say John on a PSTN needs to contact Chuck. Chuck's telephone is a SIP, H.323, DCP, or analog endpoint, connected to the PSTN via an enterprise network and gateway. John already knows Chuck's number. Before dialing Chuck's phone number, John navigates to a URL that includes something unique to Chuck in its name, such as www.xyzco.com/303-555-1212 or www.xyzco.com/chuck. The webpage of the URL contains two fields: number you will be calling from and subject. John enters his number and then enters the subject “Sale going through!” John then dials Chuck's number. The enterprise network receives the call and the associated Caller-ID via its PSTN gateway, maps the inbound Caller-ID to the information provided by John, routes the call to Chuck's phone, and causes the display on Chuck's phone to show John's Caller-ID and the subject of the call. Chuck thinks the subject may be important and therefore decides to answer. | 12-02-2010 |
20100303226 | BARTERING SYSTEM AND METHOD FOR CONTROLLING POSITION IN A WAIT QUEUE IN A CONTACT CENTER - A contact center establishes a communication with a user. The communication is placed into a position in a wait queue that has other position(s) with other communication(s) that are waiting to be serviced by contact center agents. An offer is made to the user to change an amount of time to wait in the wait queue before being connected to a contact center agent. The offer can be based on a commodity such as money, frequent flyer miles, willingness to listen to an advertisement while holding, completing a survey, and the like. In response to the user accepting the offer, the position of the communication in the wait queue is changed to a different position. This allows the user to interactively adjust their wait time. | 12-02-2010 |
20100322391 | PERSONAL IDENTIFICATION AND INTERACTIVE DEVICE FOR INTERNET-BASED TEXT AND VIDEO COMMUNICATION SERVICES - The present invention is directed to a method and device for assigning a telephone number to a browsing session and for providing physical location information to a public safety answering point in association with a web-based chat or relay session. | 12-23-2010 |
20100322395 | UNIFIED COMMUNICATIONS APPLIANCE - A unified communications appliance provides integration of various types of information, regardless of the modality, in a common, centralized interface where the various types of information are grouped based on what they are related to. For example, as is common with most modalities of information exchange, there exists a “subject” that is present in one of the fields of communication. The ability to associate all of the types of communication with a common “subject” (or conversation) and provide an interface that allows access to the various types of information, regardless of the modality is provided by the unified communications interface. | 12-23-2010 |
20100322397 | METHOD TO SET THE FLAG AS REPLIED OR FORWARDED TO ALL REPLIED OR FORWARDED VOICE MESSAGES - To assist with the usability of voice messaging systems, a technique is provided that provides audible information to a user about the status of one or more messages in the system. An audible message is played to the user indicating whether they have forwarded, replied, saved, or otherwise addressed a voice message. The audible indication can be triggered by the message recipient trying to reply, forward or otherwise access a message in the voice messaging system. This can be especially useful when the voice messaging system is accessed from a phone, such as an IP telephone or SIP phone. The technique at least allows users to know which voice messages they have addressed as well as to provide a summary of one or more messages in the voice messaging system such as 3 unread, 2 replied, 1 forwarded, and a total of 12 messages with message duration of 11 minutes. | 12-23-2010 |
20110026691 | STATE-BASED MANAGEMENT OF MESSAGING SYSTEM JITTER BUFFERS - Buffering is made more efficient by resizing a jitter buffer based, for example, on a user's location within a TUI. To illustrate how this might be implemented in a TUI-based system, assume that two jitter buffer sizes are available: a larger one for voice and a smaller one for DTMF. Assume that the ability to select the buffer size is software-controllable. By virtue of the TUI structure, the initial state for a communication session could be a buffer size appropriate for DTMF. Since the messaging system may provide an audible beep whenever it's appropriate for a user to speak, the same sub-routine within the TUI code that triggers the beep could also command the buffer management mechanism instructing it to size the buffer for voice. Any subsequent DTMF entry or other event indicating that voice input has been terminated could cause the buffer to resize appropriately for DTMF. | 02-03-2011 |
20110044440 | SENDING A USER ASSOCIATED TELECOMMUNICATION ADDRESS - One or more participants in a communication are authenticated using an authentication metric such as a face print or voice print. A single telecommunication address (or one for each participant) that is not associated with a communication device and is associated with at least one of the participants is determined. The telecommunication address (or addresses) is sent during the initiation of a communication session. | 02-24-2011 |
20110055555 | LICENSING AND CERTIFICATE DISTRIBUTION VIA SECONDARY OR DIVIDED SIGNALING COMMUNICATION PATHWAY - In one embodiment, the present invention is directed to the use of separate communication pathways over different types of networks to handle bearer and control signaling in connection with a license transaction. | 03-03-2011 |
20110069625 | PRIORITY-BASED, DYNAMIC OPTIMIZATION OF UTILIZED BANDWIDTH - Methods, devices, and systems are provided for performing priority-based codec conversions. Such conversions may implemented on calls sharing a link having limited bandwidth and may also be implemented after the call has been established but before the call has been terminated. The mechanisms provided herein maximize per-user call completion/quality and overall utilization of network bandwidth by dynamically adjusting encoding algorithms and transmission characteristics of calls using the constrained link. | 03-24-2011 |
20110071884 | Customer Loyalty, Product Demonstration, and Store/Contact Center/Internet Coupling System and Method - A system for storing information about searches and inquiries by a customer is provided. The system includes a customer service server that receives information from two or more sources, such as from a retail location sales agent, a website, a call center agent, etc. The information is associated and correlated to interrelate inquiries from the different sources. Further, when the user enters a retail location, a node or server at the retail location can push test application to a user's mobile device based on the past inquiries. These test applications are provided only when the customer is present in the retail location. As such, hacking the application is prevented. Further, with the customer using the application in the retail location, a sales agent is present to assist the customer. | 03-24-2011 |
20110075821 | AUTOMATIC CONFIGURATION OF SOFT PHONES THAT ARE USABLE IN CONJUNCTION WITH SPECIAL-PURPOSE ENDPOINTS - The present disclosure is directed, in some embodiments, to automatic switching of a telephony module between different operational modes in response to the identification of different types of incoming contacts. | 03-31-2011 |
20110091021 | SELECTION AND INITIATION OF IVR SCRIPTS BY CONTACT CENTER AGENTS - Methods, devices and systems for selecting IVR script and initiating that IVR script are provided. More specifically, playback of an IVR script to a user of a client endpoint can be initiated by an agent at a content sharing endpoint selecting, dragging, and dropping a representation of that IVR script onto a representation of the voice communication session. After initiation of the IVR script, the agent can drop off of the call, leaving the IVR system to interact with the client endpoint without requiring further agent involvement. | 04-21-2011 |
20110093548 | CONFERENCE-ENHANCING ANNOUNCEMENTS AND INFORMATION - A conference participant attempting to log into a conference that has been rescheduled is informed that the conference has been rescheduled to such-and-such date and/or time. If the login attempt is before or during the conference, the participant is offered information pertaining to the conference. If the login attempt is after the conference has ended, the participant is offered the information plus a recording of the conference. An identifier is associated with the conference and used by the host and the participants to uniquely identify the conference. | 04-21-2011 |
20110116505 | PACKET HEADERS AS A TRIGGER FOR AUTOMATIC ACTIVATION OF SPECIAL-PURPOSE SOFTPHONE APPLICATIONS - Methods, devices, and systems for automatically controlling the activation and/or deactivation of communication applications are provided. More specifically, methods, devices, and systems are provided such that the inspection of communication packet headers can be leveraged as a trigger for automatically activating and/or deactivating communication applications and the population of a corresponding user-interface to the application. | 05-19-2011 |
20110248927 | MULTI-MODE TOUCHSCREEN USER INTERFACE FOR A MULTI-STATE TOUCHSCREEN DEVICE - A simple exemplary embodiment can leverage the ability of the touchscreen or touchpad device to distinguish between a fingernail and a palm-side fingertip press as binary distinctions. This can be done via long nails or alternatively, by rotating a hand to invert the finger. The binary distinction can be used to perform different functions. For example, a fingertip press could be the functional equivalent of a left-click on a mouse, and a fingernail press could be the equivalent of a right-click. Another example using a simply binary distinction could be that a fingertip press while typing will result in lower case, and fingernail press while typing will result in upper case. Another example, using a binary distinction, would interpret a light touch for right-clicks or upper case characters and a heavy touch for left-clicks or lower case characters. | 10-13-2011 |
20110248946 | MULTI-MODE PROSTHETIC DEVICE TO FACILITATE MULTI-STATE TOUCH SCREEN DETECTION - Aspects are directed toward an active prosthetic that includes, for example, an LED, RF transponder, or comparable electrical, optical, and/or electromagnetic componentry that allows the characteristics of the prosthetic to be changed. These characteristics then can be correlated to different modes of operation when used with a corresponding input device. Other aspects are directed toward utilizing prosthetics with different shapes to affect different modes of behavior and input with an input device, such as a touchscreen or touchpad. Even further aspects are directed toward providing handicapped individuals with increased dexterity by providing a prosthetic that allows different modes of behavior when used with an associated input device. | 10-13-2011 |
20110282650 | AUTOMATIC NORMALIZATION OF SPOKEN SYLLABLE DURATION - A very common problem is when people speak a language other than the language which they are accustomed, syllables can be spoken for longer or shorter than the listener would regard as appropriate. An example of this can be observed when people who have a heavy Japanese accent speak English. Since Japanese words end with vowels, there is a tendency for native Japanese to add a vowel sound to the end of English words that should end with a consonant. Illustratively, native Japanese speakers often pronounce “orange” as “orenji.” An aspect provides an automatic speech-correcting process that would not necessarily need to know that fruit is being discussed; the system would only need to know that the speaker is accustomed to Japanese, that the listener is accustomed to English, that “orenji” is not a word in English, and that “orenji” is a typical Japanese mispronunciation of the English word “orange.” | 11-17-2011 |
20110282669 | Estimating a Listener's Ability To Understand a Speaker, Based on Comparisons of Their Styles of Speech - An automated telecommunication system adjunct is described that “listens” to one or more participants styles of speech, identifies specific characteristics that represent differences in their styles, notably the accent, but also one or more of pronunciation accuracy, speed, pitch, cadence, intonation, co-articulation, syllable emphasis, and syllable duration, and utilizes, for example, a mathematical model in which the independent measurable components of speech that can affect understandability by that specific listener are weighted appropriately and then combined into a single overall score that indicates the estimated ease with which the listener can understand what is being said, and presents real-time feedback to speakers based on the score. In addition, the system can provide recommendations to the speaker as to how improve understandability. | 11-17-2011 |
20120044156 | MULTI-FINGER SLIDING DETECTION USING FINGERPRINTS TO GENERATE DIFFERENT EVENTS - Fingerprint portions of two or more different fingers are detected on a detection surface, such as an optical surface, a touch pad, a touchscreen, or the like, and then a further detection made that the person has moved their finger(s), for example, apart, together or relative to one another. The movement can be detected based on identifying the fingerprint portion sliding across the screen. The combination of fingerprint information associated with a corresponding motion is correlatable to one or more actions or triggering events that are used to control one or more electronic devices. Further aspects are directed toward utilizing one or more of the techniques herein for a security application. For example, two users, each placing one or more fingers on a touch screen or touch pad, with the fingerprints thereafter being recognized, perform a certain movement with this triggering the unlocking, or locking, of an object. | 02-23-2012 |
20120063587 | MULTI-MICROPHONE SYSTEM TO SUPPORT BANDPASS FILTERING FOR ANALOG-TO-DIGITAL CONVERSIONS AT DIFFERENT DATA RATES - One exemplary problem addressed by the techniques disclosed herein is that the A-to-D converter of the broad-band audio codec that is being used more frequently in IP telephony, G.722, samples the analog audio source 16,000 times per second, rather than 8,000. Since all G.722-capable telephones must continue to be G.711-capable, one problem is that a microphone that provides appropriate bandpass filtering for G.722 encoding fails to provide adequate filtering for G.711. One exemplary aspect is therefore directed to telephones that must be able to switch back and forth between narrow-band digital audio encoding in which the A-to-D converter samples the audio stream 8,000 times per second, and a wide-band audio encoding in which the A-to-D converter samples the audio stream 16,000 times per second. This is accomplished using one or more of a plurality of switched microphones, a filter and a modification of the resonant frequencies of a handset. | 03-15-2012 |
20120075202 | EXTENDING THE TOUCHABLE AREA OF A TOUCH SCREEN BEYOND THE BORDERS OF THE SCREEN - A touchable area of a user interface (such as that displayed on a touch screen touchpad or trackpad), i.e., the locations where the user may place their finger in order to initiate an action, can be extended beyond the border of the screen or device through the use of remote sensors that can detect when, for example, a finger or object is present at a specific location in 3-D space relative to the device. Sensors exist to perform this detection and are typically based on one or more of optical detection (for example infrared), acoustic detection (for example, via high frequency echo location), ultrasonic detection, inductive detection, capacitive detection, and in general can be based on any type of opto, opto-electronic, electrical and/or electro-mechanical sensor technology. | 03-29-2012 |
20120163558 | LATENCY COMPENSATION ADJUNCT FOR TELECONFERENCE VOICE SWITCHES - When making a decision about who should have control of a voice channel in a teleconference, one exemplary aspect is directed toward a voice switch mechanism that determines and takes into account a latency (such as a round trip latency) of each participant's path to the switch. The switch uses this information to ensure that short-latency paths do not have an unfair advantage over long-latency paths when individuals competitively seek control of the voice channel. Illustratively, if an individual is participating in the path that has a round trip latency of 300 ms or greater than that of other participants, the voice switch creates a level playing field even if it detects voice energy from a short-latency user first by granting control of the channel to the long-latency user if voice energy is detected from that user within 300 ms of the short-latency detection. | 06-28-2012 |
20120209781 | Customer Loyalty, Product Demonstration, and Store/Contact Center/Internet Coupling System and Method - A system for storing information about searches and inquiries by a customer is provided. The system includes a customer service server that receives information from two or more sources, such as from a retail location sales agent, a website, a call center agent, etc. The information is associated and correlated to interrelate inquiries from the different sources. Further, when the user enters a retail location, a node or server at the retail location can push test application to a user's mobile device based on the past inquiries. These test applications are provided only when the customer is present in the retail location. As such, hacking the application is prevented. Further, with the customer using the application in the retail location, a sales agent is present to assist the customer. | 08-16-2012 |
20120218900 | AUTOMATIC MODIFICATION OF VOIP PACKET RETRANSMISSION LEVEL BASED ON THE PSYCHO-ACOUSTIC VALUE OF THE PACKET - An exemplary technique disclosed herein is that the transmitter of a VOIP stream can assess the psycho-acoustic importance of each packet, and then use a protocol that supports redundant transmission to retransmit only the packets that are judged to be important for voice quality and intelligibility. Illustratively, a packet containing a plosive might be retransmitted redundantly because of its disproportionate contribution to intelligibility, but a packet that occurs entirely within a long-duration fricative transmitted only once. An exemplary aspect may also support multiple levels of transmission redundancy based on multiple levels of packet importance, with the levels varying based on the relative psycho-acoustic importance of each packet and/or the degree of network congestion. | 08-30-2012 |
20120275337 | SYSTEM FOR TRANSMITTING HIGH QUALITY SPEECH SIGNALS ON A VOICE OVER INTERNET PROTOCOL NETWORK - The VolP quality speech process is activated when a subscriber accesses a speech quality sensitive resource or in response to an activation of the feature by the subscriber, or when it is determined that the originating subscriber terminal device requires the transmission of high quality speech signals. A transmit buffer, associated with the port circuit that serves the originating device, stores a predetermined number of packets as they are transmitted from the originating device. In the case of lost or damaged packets, the VolP quality speech system activates the transmit buffer to retransmit the missing or damaged packet to the destination device. Intelligent buffer management is provided, where the destination device can regulate the size of the transmit buffer as well as the size of its jitter buffer. | 11-01-2012 |
20120278727 | METHOD AND APPARATUS FOR ALLOWING DRAG-AND-DROP OPERATIONS ACROSS THE SHARED BORDERS OF ADJACENT TOUCH SCREEN-EQUIPPED DEVICES - A user interface(s) in which the displays of different devices become “synchronized” when the devices are brought into close proximity with one another. One exemplary embodiment permits drag-and-drop procedures that originate on one device to be terminated on the other. Illustratively, the solution could be handled in the following manner:
| 11-01-2012 |
20130007635 | TELECONFERENCING ADJUNCT AND USER INTERFACE TO SUPPORT TEMPORARY TOPIC-BASED EXCLUSIONS OF SPECIFIC PARTICIPANTS - An aspect associates people participating in a conference with one or more specific topics. This association can limit a particular participant's participation to specific topic(s) they are associated with, the one or more media streams for the other topics excluded from viewing and/or listening. For example, if a participant is to provide a presentation for Item 4, and that is the only item the participant should be participating in, that participant could be allowed to join the conference with a status message being provided to the participant as the conference progresses through Items 1-3. An “on-deck” message can be provided to the meeting participant indicating their agenda item is almost ready for discussion, and when Item-4 is selected to be discussed, the participant is provided with one or more appropriate media streams for the conference. The media for the other media streams blocked from the participant for the other items. | 01-03-2013 |
20130122851 | DETERMINATION BY PSAPS OF CALLER LOCATION BASED ON THE WIFI HOT SPOTS DETECTED AND REPORTED BY THE CALLER'S DEVICE(S) - Location information associated with a wireless access point is used to assist with emergency call routing. Additionally, the location information can be used to assist with determining where an emergency call is physically originating from. This location information is one or more of enterable, detectable and/or populated with the assistance of a location determining device, such as a GPS, associated with the wireless network. The location information can also be dynamic to account for mobile wireless access points, such as a mobile access point provided on public transportation. The location information is also associatiable with an outbound communication, such as an emergency communication, with this location information usable to route the communication to an appropriate entity(ies). | 05-16-2013 |
20130163747 | MULTI-MICROPHONE SYSTEM TO SUPPORT BANDPASS FILTERING FOR ANALOG-TO-DIGITAL CONVERSIONS AT DIFFERENT DATA RATES - One exemplary problem addressed by the techniques disclosed herein is that the A-to-D converter of the broad-band audio codec that is being used more frequently in IP telephony, G.722, samples the analog audio source 16,000 times per second, rather than 8,000. Since all G.722-capable telephones must continue to be G.711-capable, one problem is that a microphone that provides appropriate bandpass filtering for G.722 encoding fails to provide adequate filtering for G.711. One exemplary aspect is therefore directed to telephones that must be able to switch back and forth between narrow-band digital audio encoding in which the A-to-D converter samples the audio stream 8,000 times per second, and a wide-band audio encoding in which the A-to-D converter samples the audio stream 16,000 times per second. This is accomplished using one or more of a plurality of switched microphones, a filter and a modification of the resonant frequencies of a handset. | 06-27-2013 |
20130236002 | USING FACTOR ANALYSIS TO IMPROVE WORK ASSIGNMENT PERFORMANCE - In the next generation contact center, a plethora of attributes may be used to describe incoming work requests as well as agents able to handle the work. A work assignment engine may have to sort through hundreds of combinations of attributes in order to identify the optimal or a close-to-optimal solution. One of the problems is how to process this amount of information quickly, as discussed above, at times on systems that do not have the computational horsepower to analyze complex data in a timely manner. This can create a tremendous, unmanageable computational burden for the contact center. One exemplary embodiment reduces the computational burden, and provides additional benefits, by employing a contact center-optimized extension of factor analysis techniques. In general, factor analysis is a statistical method used to describe variability among observed, correlated variables, e.g., attributes, in terms of a potentially lower number of unobserved, uncorrelated variables called factors. | 09-12-2013 |
20130272565 | AGENT MATCHING BASED ON VIDEO ANALYSIS OF CUSTOMER PRESENTATION - Systems and methods for routing and/or servicing contacts using video analysis of one or more video streams are provided. The systems and methods are particularly applicable to a contact center. | 10-17-2013 |
20130293501 | DETECTION OF A ROLLING MOTION OR SLIDING MOTION OF A BODY PART ON A SURFACE - An optical scanner scans a first portion of a print of a body part such as a finger in a first area of an optical surface. The optical scanner detects a motion of the body part to a second area of the optical surface. This can be done in various ways. One way is for the optical scanner to detect a sliding motion of the body part to determine if most of the first portion of the print is in the second area. Another way is for the optical scanner to determine a rolling motion of the print based on a continuity of the print from the first area to the second area. A similar system and method is disclosed which detects a rolling motion of a body part by using a sleeve with multiple properties. | 11-07-2013 |
20130321133 | METHOD AND APPARATUS FOR IDENTIFYING A SPEAKER - Systems and methods for identifying a participant providing audible information during a communication session are disclosed. More particularly, speech localization is utilized to determine a location of the participant providing audible information. An identification device determined to be at a location corresponding to the location of the participant providing audible information is identified. The identity of the participant providing the audible information is then obtained by mapping the identification device to the participant. The information identifying the participant providing audible information can be provided to other endpoints of the communication session. | 12-05-2013 |
20130331053 | SPECIAL HANDLING OF CERTAIN TYPES OF COMMUNICATIONS - Systems and methods for providing special handling for certain types of communications are provided. More particularly, a normally secured access point can permit emergency communications to be transmitted in connection with unauthorized communication devices. Supplemental communications, for example to websites included in a predefined list by a communication device, can also be permitted once an emergency communication by that communication device has been allowed. Accordingly, emergency communications can be supported, while continuing to limit access with respect to non-emergency communications. | 12-12-2013 |
20130331149 | TACTILE INDICATION OF TRANSMISSION QUALITY IMPAIRMENTS - Systems and methods for providing a tactile alert or indication of a condition comprising a transmission quality impairment affecting a communication device are provided. More particularly, in response to detecting a transmission quality impairment, a tactile alert can be provided to the user of the communication device. The tactile alert can be varied in a recognizable way to indicate the type and/or degree of impairment. The generation of a tactile alert allows the user to be apprised of the condition, which may prevent the provision of speech or other input by the user from being clearly received by other communication endpoints, to be provided with certainty. Moreover, the condition can be communicated in compliance with communication device accessibility requirements. | 12-12-2013 |
20140029472 | PERSONAL IDENTIFICATION AND INTERACTIVE DEVICE FOR INTERNET-BASED TEXT AND VIDEO COMMUNICATION SERVICES - The present invention is directed to a method and device for assigning a telephone number to a browsing session and for providing physical location information to a public safety answering point in association with a web-based chat or relay session. | 01-30-2014 |
20140046656 | METHOD AND APPARATUS FOR AUTOMATIC COMMUNICATIONS SYSTEM INTELLIGIBILITY TESTING AND 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. | 02-13-2014 |
20140133647 | SPEAKER PHONE NOISE SUPPRESSION METHOD AND APPARATUS - Systems and methods for removing noise from an audible signal are provided. More particularly, a vibration sensor is used to obtain a vibration signal from an environment including a communication device. The signal from the vibration sensor is combined with a signal from a microphone associated with the communication device, to create a modified audible signal. More particularly, a filtering or subtraction process can be performed with respect to the audible signal, at a time corresponding to an event detected as part of the vibration signal. The resulting modified audible signal can have reduced noise as compared to the original audible signal. | 05-15-2014 |
20140169534 | METHOD, APPARATUS, AND SYSTEM FOR PROVIDING REAL-TIME PSAP CALL ANALYSIS - Methods, apparatus, and systems are provided such that a Public Safety Answering Point (PSAP) may utilize a new model to handle Open Line emergency calls, including audio optimization, automation, analysis, and presentation. Embodiments of the present disclosure assist with the difficult task of identifying background noise while trying to listen and talk to a caller, and give the best possible audio from the caller to the emergency call-taker or dispatcher. More particularly, an audio stream is split into at least two instances, with a first instance being optimized for speech intelligibility and provided to a call-taker or dispatcher and a second instance being provided for background sound analysis. Accordingly, the new PSAP Open Line model may allow for significantly more efficient emergency assessment, location, and management of resources. | 06-19-2014 |
20140200884 | TELECOMMUNICATIONS METHODS AND SYSTEMS PROVIDING USER SPECIFIC AUDIO OPTIMIZATION - 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. | 07-17-2014 |
20140241512 | SYSTEMS AND METHODS TO SUPPORT USING ANALOG TTY DEVICES WITH VOICE-ONLY PC SOFT CLIENTS - Methods and systems for allowing analog TTY devices to be operated in conjunction with voice-only soft client applications are provided. Supported communication styles include a text-only TTY mode, a mode in which the user of the soft client receives text and responds by voice, a mode in which the user receives voice and responds with text, and a voice-only mode. Mode selection may be manual or automatic. Communication between the TTY device and far-end user may be via a direct connection between the TTY and the associated PBX, or via a connection between the TTY and the PC that is hosting the soft phone application. | 08-28-2014 |
20140244235 | SYSTEM AND METHOD FOR TRANSMITTING MULTIPLE TEXT STREAMS OF A COMMUNICATION IN DIFFERENT LANGUAGES - A communication, such as a voice communication, is established between a communication device and a media application. The media application may be, for example, a voice conferencing system or a media server. Some or all of the communication is translated into a plurality of text streams that are in different languages. The plurality of text streams are transmitted to the communication device along with the stream of the communication. The communication device receives the text streams in the different languages and the stream of the communication. A user of the communication device can select a language for displaying one or more of the text streams in the different languages. The text stream associated with the selected language is then displayed to the user in conjunction with the stream of the communication. | 08-28-2014 |
20140270144 | PUBLIC SAFETY ANSWERING POINT LANGUAGE DETECTION - A Public Safety Answering Point (PSAP) is disclosed. The PSAP is configured to enable the detection of language preferences, capabilities, or inabilities and, based on such detection, assist the PSAP agent in assisting the PSAP caller. The PSAP may additionally or alternatively utilize the detection of language information to assist in the decision to route or re-route the PSAP caller to appropriate PSAP resources. | 09-18-2014 |
20140285474 | EVENT GENERATION BASED ON PRINT PORTION IDENTIFICATION - An optical scanner is configured to scan multiple print portions of a body part such as a finger. The optical scanner identifies a first one of the print portions in an area of an optical surface. An event such as launching an application is generated based on identifying the first print portion in the area of the optical surface. In addition, various events can be generated based on different combinations of print portions in different areas of the optical surface. | 09-25-2014 |
20140314212 | PROVIDING ADVISORY INFORMATION ASSOCIATED WITH DETECTED AUDITORY AND VISUAL SIGNS IN A PSAP ENVIRONMENT - A Public Safety Answering Point (PSAP) is configured to enable the detection of one or more clinical signs associated with a caller. The clinical signs may include both auditory and visual clinical signs and may be detected by analyzing a portion of the call information to determine one or more characteristics associated with the call information and comparing the one or more determined characteristics to known clinical sign characteristics. The PSAP may additionally or alternatively utilize the detection of the clinical signs to assist and/or provide an advisory recommendation in the decision of which, if any first responder resources should be dispatched; the recommendation may include which resources and at what priority the resources should be dispatched. | 10-23-2014 |
20140357215 | METHOD AND APPARATUS TO ALLOW A PSAP TO DERIVE USEFUL INFORMATION FROM ACCELEROMETER DATA TRANSMITTED BY A CALLER'S DEVICE - A Public Safety Answering Point (PSAP) is configured to receive a description of detected movement associated with one or more inertial sensors of a communication device. The description of the detected movement may then be analyzed to determine a likelihood of injury associated with the movement, whether or not the user associated with the communication device is experiencing abnormal neuromuscular activity, and/or whether the user is outside a range of the communication device such that the user is unlikely to hear audio information originating from the communication device. The PSAP may additionally or alternatively utilize the description of the movement information to assist and/or provide an advisory recommendation in the decision of which, if any first responder resources should be dispatched; the recommendation may include which resources and at what priority the resources should be dispatched. | 12-04-2014 |
20150030152 | METHOD AND SYSTEM FOR DETERMINING CUSTOMER'S SKILL, KNOWLEDGE LEVEL, AND/OR INTEREST - A microprocessor executable work assignment mechanism selects a work item associated with a customer, dynamically determines, for the customer, a customer proficiency level with respect to a contact center product and/or product area sold and/or serviced by a contact center, and uses the customer proficiency level in assigning the selected work item to a resource for servicing and/or in providing the determined customer proficiency level to an assigned resource for use in servicing the selected work item. | 01-29-2015 |
20150038102 | EMERGENCY REQUEST PRIOR INSIGHT DELIVERY - A communication endpoint is configured to record content which may be audio content, video content, and/or other content and send the recorded content to an endpoint, such as a contact center. In one instance, the communication endpoint initiates content recording upon the detection of an outbound user initiated request, such as an emergency request, and transmits the recorded content when the communication session has been established. As another example, the communication endpoint may record content in a circular manner such that a latest content corresponding to a predetermine amount of time may be transmitted when the communication session has been established. | 02-05-2015 |