摘要:
Event data messages can be provided by an interactive voice response (IVR) system to a complex events processor (CEP). The event data messages can include a Stream_ID and a series of textual elements. The Stream_ID can uniquely identify a call session between a caller and the IVR system. The series of textual elements can represent speech input provided by the caller. The CEP can create a text string from the series of textual elements of event data messages having the same Stream_ID. The text string can inherit the Stream_ID of the event data messages. The CEP can utilize user-defined business rules to process the text string. When the CEP issues an action message, the Stream_ID of the text string can be appended to the action message sent to the IVR system. The IVR system can modify the call session associated with the Stream_ID of the action message accordingly.
摘要:
The present invention discloses an open contact center formed from multiple contact center components that conform to open standards and that intercommunicate utilizing open standards. The open contact center can provide automated interactive communications with callers, can provide queue management for callers waiting to communicate with live agents, can provide skills based routing for assigning live agents to callers based upon skills of the live agents and skills needed by the callers, and can provide tooling for provisioning and monitoring the live agents. In one configuration, the contact center components can execute within a middleware solution, such as WEBSPHERE, that has IP Multimedia Subsystem capability. Additionally, the contact center components can be implemented as service oriented architecture (SOA) components that communicate over an enterprise service bus (ESB).
摘要:
The present invention discloses an open contact center formed from multiple contact center components that conform to open standards and that intercommunicate utilizing open standards. The open contact center can provide automated interactive communications with callers, can provide queue management for callers waiting to communicate with live agents, can provide skills based routing for assigning live agents to callers based upon skills of the live agents and skills needed by the callers, and can provide tooling for provisioning and monitoring the live agents. In one configuration, the contact center components can execute within a middleware solution, such as WEBSPHERE, that has IP Multimedia Subsystem capability. Additionally, the contact center components can be implemented as service oriented architecture (SOA) components that communicate over an enterprise service bus (ESB).
摘要:
Event data messages can be provided by an interactive voice response (IVR) system to a complex events processor (CEP). The event data messages can include a Stream_ID and a series of textual elements. The Stream_ID can uniquely identify a call session between a caller and the IVR system. The series of textual elements can represent speech input provided by the caller. The CEP can create a text string from the series of textual elements of event data messages having the same Stream_ID. The text string can inherit the Stream_ID of the event data messages. The CEP can utilize user-defined business rules to process the text string. When the CEP issues an action message, the Stream_ID of the text string can be appended to the action message sent to the IVR system. The IVR system can modify the call session associated with the Stream_ID of the action message accordingly.
摘要:
A method for providing voice telephony services can include the step of receiving a call via a telephone gateway. The telephone gateway can convey call identifying data to a resource connector. A media port can be responsively established within a media converter that is communicatively linked to the telephone gateway through a port associated with the call. A call description object can be constructed that includes the call identifying data and an identifier for the media port. The call description object can be conveyed to a telephony application server that provides at least one speech service for the call. The telephony application server can initiate at least one programmatic action of a communicatively linked speech engine. The speech engine can convey results of the programmatic action to the media converter through the media port. The media converter can stream speech signals for the call based upon the results.
摘要:
The present invention discloses a speech-enabled application that includes two or more linked markup documents that together form a speech-enabled application served by a Web 2.0 server. The linked markup documents can conform to an ATOM PUBLISHING PROTOCOL (APP) based protocol. Additionally, the linked markup documents can include an entry collection of documents and a resource collection of documents. The resource collection can include at least one speech resource associated with a speech engine disposed in a speech processing system remotely located from the Web 2.0 server. The speech resource can add a speech processing capability to the speech-enabled application. In one embodiment, end-users of the speech-enabled application can be permitted to introspect, customize, replace, add, re-order, and remove at least a portion of the linked markup documents.
摘要:
The present solution includes an automated response method. The method can receive user interactions entered through a real-time text exchange interface. These user interactions with the speech application can be dynamically and automatically converted as necessary into a format consumable by a voice server. A text input API of a voice server can be used to allow the voice server to directly accept text input. Further, automated interactions can be received from the voice server, which are dynamically and automatically converted into a format accepted by the text exchange interface. The text exchange interface can be an off-the-shelf unmodified interface. The speech application can be a VoiceXML based application that lacks an inherent text exchange capability.
摘要:
A method (200) and a system (100) for coordinated streaming use a single Real Time Protocol (RTP) producer (130) for handling multiple audio services (110). The method can include the steps of assigning (202) a RTP producer to handle multiple audio objects, and maintaining (204) a service for each object in accordance with a delivery schedule. RTP packets can be sent in accordance with the delivery schedule for complying with real-time requirements of a media rendering client thereby providing continuous real-time service delivery. The method can further include determining a wait time and updating the delivery schedule in view of the wait time. In one arrangement, the RTP producer can sleep for a pre-specified interval, and upon wake, prioritizes service delivery based on an audio object's wait time.
摘要:
A method for extending markup supported by a browser can include a step of identifying a browser that presents information written in a markup language. An extender can be identified that includes at least one extension to the markup language that the browser does not normally support. The extender can be loaded resulting in the markup language supported by the browser being extended to include the extension.
摘要:
Embodiments of the present invention provide a method and computer program product for the automated voice enablement of a Web page with free form input field support. In an embodiment of the invention, a method for voice enabling a Web page with free form input field support can be provided. The method can include receiving speech input for an input field in a Web page, parsing a core attribute for the input field and identifying an external statistical language model (SLM) referenced by the core attribute of the input field, posting the received speech input and the SLM to an automatic speech recognition (ASR) engine, and inserting a textual equivalent to the speech input provided by the ASR engine in conjunction with the SLM into the input field.