Abstract: A system and method for securing the mobile device applies the rules to determine if an event associated with an application is a secure event. If the event is a secure event, the system applies the rules to determine if the event is authenticated. If the event is authenticated, the event is authorized and the system updates rule data associated with the event and/or other associated events. Updating the rule data allows other associated events to be authenticated. If the event is not authenticated, the system requests authentication from a user. If the authentication is valid, the event is authorized and the system updates the rule data associated with the event and/or other associated events. If the authentication is not valid, the system secures the mobile device. Authorizing the event enables a user to access the application and/or data associated with the application.

Abstract: The system and method described herein allows for full monitoring and recording of SIP calls by using standard SIP messages. During the call set up between a first SIP device and a second SIP device, information is derived from a first SIP INVITE message from a first SIP device. Information is then derived from a response message from the second SIP device. The monitor/recorder receives one or more SIP INVITE messages to set up a first media stream from the first SIP device and a second media stream from the second SIP device to the monitor/recorder. The monitor/recorder receives the first and/or second media stream. The monitor/recorder then records the first and/or second media stream. In addition, the system and method provide for recording on demand, required recording, and pause-and-resume recording of SIP calls.

Abstract: A distributed interactive voice processing system is disclosed that distributes IVR capabilities to one or more agent terminals associated with a call center. One or more agent terminals support the traditional functions of a live agent, as well as additional IVR capabilities to support the functions of a virtual or automated agent A call management system manages the distributed IVR resources by monitoring the availability status of the live and automated IVR agents in the distributed interactive voice processing system and distributing telephone calls and other types of communications to the appropriate agents. The availability status of the automated IVR agents is based on estimated available CPU cycles on the agent terminal A received call is routed to the extension on the agent terminal associated with the selected IVR channel.

Abstract: A method and apparatus are disclosed for delivering a content stream to a terminal only when the terminal has a transducer (e.g., speaker, video display, etc.) that is (i) capable of outputting the content stream's media type, and (ii) there are no software applications running at the terminal that use the transducer. For example, if a terminal has both a speaker and a video display, and the terminal is running an application that sends output to the video display, an audio stream, but not a video stream, is delivered to the terminal. The illustrative embodiment is disclosed in the context of an interactive voice response system, but is applicable in a wide variety of other telecommunications systems and applications.

Abstract: A method and apparatus are disclosed for delivering a content stream to a terminal only if: (i) the terminal has a transducer (e.g., speaker, video display, etc.) that is capable of outputting the content stream's media type, and (ii) the battery of the terminal has sufficient energy to handle receiving, storing, processing, and outputting the content stream. In accordance with the illustrative embodiment, a telecommunications terminal periodically transmits a message that indicates the energy level of its battery, or the media types (e.g., audio, video, etc.) that are welcome at the terminal, or both. The illustrative embodiment is disclosed in the context of an interactive voice response system, but is applicable in a wide variety of other telecommunications systems and applications.

Abstract: An apparatus and method are disclosed that enable an interactive voice response (IVR) system to select, tailor, and deliver a “filler” content stream to a calling telecommunications terminal during a delay in a call (e.g., when performing automated speech recognition, retrieving other content, etc.). The delivery of the filler content can reduce the chance that the caller terminates the call prematurely, and can also be used to provide information to the caller, advertise new products, etc. The filler content can be based on one or more of the following: the date and time, the identity of the caller, a prior call to the IVR system, the type of the calling telecommunications terminal, and a content stream that the IVR system is in the process of retrieving.

Abstract: A method and apparatus are disclosed for enabling an interactive voice response (IVR) system to deliver a video stream to a telecommunications terminal and handle events related to the video stream. In particular, the illustrative embodiment overcomes two disadvantages of the prior art. First, the illustrative embodiment enables events to be generated and caught by a script of an IVR system at any time during a call. Second, the illustrative embodiment enables events to be handled in parallel with the execution of a script of the IVR system, thereby enabling the performing of actions response to a stream-related event without halting or interfering with other tasks of the IVR script.

Abstract: A method and apparatus are disclosed that enable the synchronization of user profile information for M users who are sharing K telecommunications terminals, wherein M and K are not necessarily equal to each other. The disclosed technique addresses the problem that occurs when multiple users share multiple terminals, in which a user that selects a particular terminal to use has to contend with the contact lists, user preferences, and call log entries that are left over from the previous user or users of the selected terminal. In accordance with the illustrative embodiment of the present invention, the user profile information is associated with and stored for each user in a group of users, in contrast to being associated with and stored for the telecommunications terminal. The user profile information comprises (i) contact information, (ii) call or message activity, and (iii) user terminal preferences.

Abstract: The present invention enables service agents in a call center to monitor portions of calls that are especially error-prone (e.g., automated speech recognition of an address, etc.), while portions of calls that are not error-prone occur without any human monitoring. An interactive voice response (IVR) system script (e.g., a VXML script, etc.) is divided into a plurality of independent dialog sequences, each of which is assigned a human-monitoring requirement that indicates whether (and optionally, to what degree) the dialog sequence requires monitoring by a service agent. In the first illustrative embodiment, a partial ordering enables some dialog sequences to appear in a different order in that of the IVR script when it is advantageous to do so. In the second illustrative embodiment, each dialog sequence that requires human monitoring is recorded and packaged into an agent-review task that is subsequently sent to an available service agent for review.

Abstract: Apparatus and methods are disclosed that enable an interactive voice response (IVR) system to deliver content streams of various media types (e.g., video, audio, etc.) to telecommunications terminals via the addition of extensions to the Voice extensible Markup Language (VXML). The IVR system will deliver a particular content stream to a terminal only if: (i) the terminal has a transducer (e.g., speaker, video display, etc.) that is capable of outputting the content stream's media type, and (ii) that transducer is currently enabled.

Abstract: A method and apparatus are disclosed that enable an interactive voice response (IVR) system to generate video content in addition to audio content (e.g., synthesized speech, etc.). The video content is based on the state of the IVR system, the video display capacity of the calling telecommunications terminal, and information supplied by the user via a telecommunications terminal. The illustrative embodiment generates video content based on the text from which the audio content is generated. In particular, the video content comprises an abridged version of this text that is suitable for display at the telecommunications terminal. In the illustrative embodiment, the abridged version of the text is generated via syntactic and semantic processing. In addition, an abridged version of user-supplied information is generated and incorporated into the video content.

Abstract: A method and apparatus are disclosed for intelligently delivering components of a multimedia content stream (e.g., audio component, video component, etc.) to a telecommunications terminal based on the quality of service (QoS) for transmissions received at the terminal. The illustrative embodiment is disclosed in the context of an interactive voice response system, but is applicable in a wide variety of other telecommunications systems and applications.

Abstract: A method and apparatus are disclosed that enable an interactive voice response (IVR) system to deliver content streams of various media types (e.g., video, audio, etc.) to telecommunications terminals. The illustrative embodiment provides extensions to the Voice extensible Markup Language (VXML) standard that enable an IVR system script to specify the playback order, timing, and coordination of multiple content streams (e.g., whether an audio stream and a video stream should be played back concurrently or serially; whether a particular content stream should finish before playback of another content stream commences; whether a content stream that is currently playing should be stopped and supplanted with another content stream, etc.).

Abstract: Methods and apparatus are provided for presenting default content to a caller when a media stream of the specified type is not available. If it is determined that a destination endpoint is not providing a media stream of a first media type to an originating endpoint, a transmission of default media of the first media type to the originating endpoint is initiated.

Abstract: A method is disclosed that enables the delivery of relevant content to a telecommunications user engaged in a call. In particular, in the illustrative embodiments a telecommunications terminal that is to receive content is selected based on (i) a telecommunications terminal involved in a call and (ii) the mode of communication (e.g., voice, video, text, etc.) of the call. In addition, in the illustrative embodiments the second terminal to which content is to be delivered might also be based on one or more of the following: the identity of the user; the identity of other users involved in the call; the telecommunications terminal employed by the user for the call; other telecommunications terminals involved in the call; the date and time; the location of the user; and the location of other users involved in the call.

Abstract: An apparatus and methods are disclosed that enable a user of a telecommunications terminal to dynamically supplant the video content of an outgoing media stream (e.g., an outgoing videoconference stream, etc.) with video from a document (e.g., a PowerPoint® file, a Windows Media Video [WMV] file, etc.) via the terminal's graphical user interface (GUI). When a user drag-and-drops a graphical object that is associated with a document onto a graphical object that is associated with the outgoing media stream, the video content of the outgoing video stream is supplanted with video content from the document. Subsequently, a user can drag-and-drop a document icon away from the second graphical object to restore the video content of the outgoing media stream to its prior source.

Abstract: A method and apparatus that enable the delivery of relevant content to a telecommunications user engaged in a call are disclosed. In the illustrative embodiment content is selected based on the state of the conversation, which is determined by any of the following: the interaction of a service agent or interactive voice response system with a software application; the menu state in an interactive voice response system; the current point in a script that is followed by the service agent; the current point in a workflow that is related to the call; information associated with the caller; and information associated with the call. Content might also be based on one or more of the following: the identity of the user, the telecommunications terminal employed by the user, the date and time, and the location of the user.

Abstract: A method is disclosed that coordinates the presentation of speech signals and screen images to a receiver of a call, without some of the costs, disadvantages, and limitations of techniques in the prior art. The illustrative embodiment of the present invention builds upon the infrastructure of the Session Initiation Protocol (or “SIP”) to distribute and transfer calls to agents who are associated with a call center or contact center system. A server-based system is used in conjunction with call distribution based on SIP to replace some desktop and agent computer/telephony integration arrangements, thereby providing first-party call control. Customer-related information is embedded in the call transfer itself; as a result, the display of the customer-related information occurs at the agent's telecommunications terminal at the same time the call arrives.

Abstract: A method and apparatus that enable the delivery of relevant content to a telecommunications user engaged in a call are disclosed. In the illustrative embodiment content is selected based on dialog of the call (e.g., speech text, etc.), and optionally, one or both of: (i) the state of the call (e.g., on-hold, transferring to another line, engaged in conversation, etc.), and (ii) the state of the conversation (e.g., greeting, data entry [such as keying in a PIN], adjournment, etc.). Content might also be based on one or more of the following: the identity of the user; the identity of other users involved in the call; the telecommunications terminal employed by the user for the call; other telecommunications terminals involved in the call; the date and time; the location of the user; and the location of other users involved in the call.

Abstract: Methods and apparatus are provided for classifying a spoken utterance into at least one of a plurality of categories. A spoken utterance is translated into text and a confidence score is provided for one or more terms in the translation. The spoken utterance is classified into at least one category, based upon (i) a closeness measure between terms in the translation of the spoken utterance and terms in the at least one category and (ii) the confidence score. The closeness measure may be, for example, a measure of a cosine similarity between a query vector representation of said spoken utterance and each of said plurality of categories. A score is optionally generated for each of the plurality of categories and the score is used to classify the spoken utterance into at least one category. The confidence score for a multi-word term can be computed, for example, as a geometric mean of the confidence score for each individual word in the multi-word term.