摘要:
The spatial sound conference system enables participants in a teleconference to distinguish between speakers even during periods of interruption and overtalk, identify speakers based on spatial location cues, understand low volume speech, and block out background noise using spatial sound information. Spatial sound information may be captured using microphones positioned at the ear locations of a dummy head at a conference table, or spatial sound information may be added to a participant's monaural audio signal using head-related transfer functions. Head-related transfer functions simulate the frequency response of audio signals across the head from one ear to the other ear to create a spatial location for a sound. Spatial sound is transmitted across a communication channel, such as ISDN, and reproduced using spatially disposed loudspeakers positioned at the ears of a participant. By inserting a spatial sound component in a teleconference, a speaker other than the loudest speaker may be heard during periods of interruption and overtalk. Additionally, speakers may be more readily identified when they have a spatial sound position, and the perception of background noise is reduced.
摘要:
A voice response unit (VRU) includes a silent prompt feature in the form of an intentional delay inserted after a message is given to a caller, during which delay the caller may invoke alternative processing or interrupt current processing. If appropriate alternative or interrupt commands are not received during the delay period, then processing continues as provided in accordance with the previously played message. The duration of this silent prompt delay is carefully selected to provide sufficient response time for the caller to request alternative processing, while avoiding a perceptible or objectionable delay to the average caller not requiring alternative processing. Optimally, this delay period should be within a range of one to two and one-half seconds and, preferably, within a range of 1.2 to 2.3 seconds, an optimal time being 1.8 seconds.
摘要:
The spatial sound conference system enables participants in a teleconference to distinguish between speakers even during periods of interruption and overtalk, identify speakers based on spatial location cues, understand low volume speech, and block out background noise using spatial sound information. Spatial sound information may be captured using microphones positioned at the ear locations of a dummy head at a conference table, or spatial sound information may be added to a participant's monaural audio signal using head-related transfer functions. Head-related transfer functions simulate the frequency response of audio signals across the head from one ear to the other ear to create a spatial location for a sound. Spatial sound is transmitted across a communication channel, such as ISDN, and reproduced using spatially disposed loudspeakers positioned at the ears of a participant. By inserting a spatial sound component in a teleconference, a speaker other than the loudest speaker may be heard during periods of interruption and overtalk. Additionally, speakers may be more readily identified when they have a spatial sound position, and the perception of background noise is reduced.
摘要:
A voice response unit (VRU) includes a library of content equivalent messages and prompts which may be substituted for one another to vary the presentation of messages provided to a user and thereby more closely simulate a human operator. Groups of content equivalent messages and prompts include multiple audio files, each with a slightly different wording or phraseology, but conveying substantially the same information. After a particular message content is selected, the corresponding group of messages is identified and a random number is generated and used to select one of the audio files of the group for playback. The VRU may be included as part of an automated dialer or auto attendant. In such a system, a calling party is greeted by the VRU and is prompted by a randomly selected prompt to speak the name of the called party. The system accesses a telephone directory, attempts to identify a name corresponding to the name spoken, and dials the number. The caller may interrupt or request alternative processing during a predetermined time period after the system has selected and read back a closest matching name or its corresponding telephone number. If processing is halted by the caller indicating that the name or telephone number selected by the system is incorrect, the system will attempt to identify a second closest guess, or if none is available, to ask the caller to reinput the name of the called party. Alternative processing includes hearing the telephone number without having it dialed, and diverting a call to voice mail.
摘要:
An intelligent telephone network provides personalized communication services based on subscriber prescribed double voice identification of the calling and answering parties on a subscriber line. Specifically, when a person requests a service, the network executes a double speaker identification/verification procedure to identify first and second subscriber identified voices on a call. One or more switching offices of the network utilize profile data associated with the identified subscriber to control services over a predetermined communication link. For example, on a call over a telephone line, the speaker identification/verification process provides a virtual office equipment number corresponding to one of the subscriber designated voices. The central office switch that is servicing the call receives the virtual office equipment number and uses that number to retrieve profile data associated with the subscriber including the subscriber prescribed processing routine. The central office switch provides a personalized form of service to the subscriber by processing the call using the retrieved profile data. As part of this service, the switch and the network provide individualized double voice identification to tailor call processing and subscribed designated external action on the basis of the double voice identification rather than the line or station that carries the call. The network can provide the personalized services to several subscribers sharing a common line or for some subscriber selected period of time.
摘要:
A telephone communications system Advanced Intelligent Network (AIN) platform provides a voice activated call dialing functionality through speaker independent phoneme speech recognition having a minimum volume of storage without requiring user template training. Speaker independent phoneme recognition identifies phoneme strings of caller spoken utterances which are then compared to phoneme string representations that previously have been stored in respective caller processing records (CPRs) for those subscribers listed in the ISCP database, or stored in an equivalent peripheral database with which the ISCP can communicate. Each stored phoneme string representation is associated in the CPR with a destination telephone number that may then be extracted to route a call.
摘要:
A mechanized directory assistance system for use in a telecommunications network includes multiple speech recognition devices comprising a word recognition device, a phoneme recognition device, and an alphabet recognition device. Also provided is a voice processing unit and a computer operating under stored program control. A database is utilized which may comprise the same database as used for operator directory assistance. The system operates as follows: A directory assistance caller is prompted to speak the city or location desired. The response is digitized and simultaneously inputted to the word and phoneme recognition devices which each output a translation signal plus a probability level signal. These are compared and the highest probability level translation is selected. The caller is prompted to speak the name of the sought party. The response is processed in the same manner as the location word. In the event that the probability level fails to meet a predetermined standard the caller is prompted to spell all or part of the location and/or name. The resulting signal is inputted to the alphabet device. When translations are obtained having a satisfactory probability level the database is accessed. If plural listings are located these are articulated and the caller is prompted to respond affirmatively or negatively as to each. When a single directory number has been located a signal is transmitted to the caller to articulate this number. The system also includes provision for DTMF keyboard input in aid of the spelling procedure.
摘要:
A voice response unit (VRU) includes a silent prompt feature in the form of an intentional delay inserted after a message is given to a caller, during which delay the caller may invoke alternative processing or interrupt current processing. If appropriate alternative or interrupt commands are not received during the delay period, then processing continues as provided in accordance with the previously played message. The duration of this silent prompt delay is carefully selected to provide sufficient response time for the caller to request alternative processing, while avoiding a perceptible or objectionable delay to the average caller not requiring alternative processing.
摘要:
A voice response unit (VRU) includes a silent prompt feature in the form of an intentional delay inserted after a message is given to a caller, during which delay the caller may invoke alternative processing or interrupt current processing. If appropriate alternative or interrupt commands are not received during the delay period, then processing continues as provided in accordance with the previously played message. The duration of this silent prompt delay is carefully selected to provide sufficient response time for the caller to request alternative processing, while avoiding a perceptible or objectionable delay to the average caller not requiring alternative processing. Optimally, this delay period should be within a range of one to two and one-half seconds and, preferably, within a range of 1.2 to 2.3 seconds, an optimal time being 1.8 seconds. The VRU may be included as part of a voice activated dialing system which recognizes a name of a party to be called, identifies the associated telephone number from a telephone directory, and reads the name found back to the caller. The system then uses a silent prompt, waiting for the described 1.8 seconds to allow the user to invoke alternative processing, such as requesting the listing, voice mail options, or a next closest match. If no alternative processing is requested during the 1.8 seconds, processing continues and the named party is called.
摘要:
An intelligent telephone network provides personalized communication services based on a voice identification of the subscriber. The network executes a speech processing operation to identify the person or a party that the person is calling as a known subscriber. The network can provide personalized services to several subscribers sharing a common line. For incoming calls to such a line, the network executes an interactive procedure to determine from the caller which subscriber is being called. If the line is free, the switch applies distinctive ringing. If the line is in use, the switching office uses the profile of the identified subscriber to provide a distinctive call waiting tone or a voice message over the line. The tone or voice message indicates to the party using the line that there is a call waiting and specifically identifies the called subscriber.