Abstract: A method and system for testing voice applications, such as VoiceXML applications, is provided. The system provides a run-time simulation environment for voice applications that simulates and automates user interaction. A user simulation script is provided in a customized mark-up language. The voice application is processed to derive a nominal output of the voice application. The user simulation script is processed to generate a simulated output for the voice application corresponding to the nominal output. The user simulation script is also processed to generate a simulated input for the voice application corresponding to a pre-determined user input to the voice application. The simulated inputs and outputs are executed in conjunction with the voice application in real-time to test the application.

Abstract: A method for providing Interactive Voice Response functionality within a multi-tiered telephony environment can identify an enterprise archive file that contains an Interactive Voice Response (IVR) application. The enterprise archive file can contain multiple interpreters for different markup languages. The enterprise archive file can also include a servlet that maps the interpreters to telephony sessions. The enterprise archive file can be deployed within a J2EE compliant application server having a SIP interface. The deployed IVR application can be utilized to provide IVR functions for SIP based telephony sessions.

Abstract: In a telephony call management system, a method of distributing calls among a plurality of packet-based load balancing resources can include receiving over a telephony channel a call request specifying call information, comparing the call information with selection criteria to select one of the plurality of packet-based load balancing resources, and querying the selected packet-based load balancing resource to resolve the call request. A response from the selected packet-based load balancing resource can be received which indicates whether the call request can be resolved.

Abstract: 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.

Abstract: A method and system for testing voice applications, such as VoiceXML applications, is provided. The system provides a run-time simulation environment for voice applications that simulates and automates user interaction. A user simulation script is provided in a customized mark-up language. The voice application is processed to derive a nominal output of the voice application. The user simulation script is processed to generate a simulated output for the voice application corresponding to the nominal output. Conditional logic may be applied to the nominal output to generate a simulated input in response thereto. The user simulation script is specified in a customized mark-up language having a set of one or more conditional tags and an internal variable for the nominal output of the voice application.

Abstract: A method and system of compiling and caching voice application documents in order to provide more efficient retrieval of resources in a web-programming model such as in the VoiceXML domain. The method and system of the present invention requires that a voice application resource such as a VoiceXML document first be transformed into a representative object model before it is stored in the system's cache memory, and that this model becomes the voice application resource that is ultimately executed by the VoiceXML interpreter. The representative object model contains all of the data of the original source document but in a compressed and optimized format therefore occupying a smaller footprint in memory. The representative object model also contains the code necessary to execute the data. Further, the compressed representative object model does not need to be compiled again before execution.

Abstract: A method of debugging a telephone call can include, using Session Initiation Protocol (SIP) formatted messages, initiating a debug session between a debugger and a call server for a telephone call established over at least a portion of a packet-switched network. A debug function can be invoked within the call server as specified by a SIP formatted message originating from the debugger. The method further can include determining fault information pertaining to the telephone call using the debug function and sending a SIP formatted response from the call server to the debugger.

Abstract: A method (200) for providing debug services between two SIP compliant endpoints is presented. The method extends SIP compliance by including a SIP DEBUG message (370). The method (200) extends the SIP protocol for allowing SIP compliant endpoints to configure themselves for providing debug capabilities. The endpoints can be a SIP client (110), a SIP server (120), or a SIP proxy (115). The method can include the steps of sending a SIP INVITE request from a first device to a second device to initiate a SIP session, receiving a SIP INVITE response from the second device, sending a SIP DEBUG request from the first device to the second device to initiate a debug session, receiving a SIP DEBUG response from the second device, and providing encrypted debugging services during the SIP session between the first SIP compliant device and the second SIP compliant device.

Abstract: The present invention discloses a contact center system based upon open standards. The contact center system can include at least one agent node, a portal server, and an application server. An agent node can include a standard HTTP browser that communicates over a standard Internet Protocol network using standard protocols. The portal server can be configured to communicate with an agent node via an agent portal, which can consist of multiple agent portlets to present information. The application server can be configured to execute contact center applications that can collect and distribute information via the agent portlets and transfer calls to agents.

Abstract: A method and system of storing and accessing documents on a computer file system utilizing a file system-independent key for use in an index-less browser caching mechanism. The present invention is a method and system for storing a resource such as a document or group of documents on a computer file system using a Universal Resource Indicator (URI) as the file name by generating a unique system-independent key from the URI for use in an index-less browser caching system. The invention takes into account the various file entry length limitations enforced by each unique file system by converting the specified URI into a key containing hexadecimal values of the key's characteristics. In this fashion, an index-less browser caching system such as a Web browser of a VoiceXML browser is created that is file system-independent.

Abstract: A method and system for testing voice applications, such as VoiceXML applications, is provided. The system provides a run-time simulation environment for voice applications that simulates and automates user interaction. A user simulation script is provided in a customized mark-up language. The voice application is processed to derive a nominal output of the voice application. The user simulation script is processed to generate a simulated output for the voice application corresponding to the nominal output. Conditional logic may be applied to the nominal output to generate a simulated input in response thereto. The user simulation script is specified in a customized mark-up language having a set of one or more conditional tags and an internal variable for the nominal output of the voice application.

Abstract: A method and system for testing voice applications, such as VoiceXML applications, is provided. The system provides a run-time simulation environment for voice applications that simulates and automates user interaction. A user simulation script is provided in a customized mark-up language. The voice application is processed to derive a nominal output of the voice application. The user simulation script is processed to generate a simulated output for the voice application corresponding to the nominal output. Conditional logic may be applied to the nominal output to generate a simulated input in response thereto. The user simulation script is specified in a customized mark-up language having a set of one or more conditional tags and an internal variable for the nominal output of the voice application.

Abstract: In a telephony call management system, a method of distributing calls among a plurality of packet-based load balancing resources can include receiving over a telephony channel a call request specifying call information, comparing the call information with selection criteria to select one of the plurality of packet-based load balancing resources, and querying the selected packet-based load balancing resource to resolve the call request. A response from the selected packet-based load balancing resource can be received which indicates whether the call request can be resolved.

Abstract: A method and system for testing voice applications, such as VoiceXML applications, is provided. The system provides a run-time simulation environment for voice applications that simulates and automates user interaction. A user simulation script is provided in a customized mark-up language. The voice application is processed to derive a nominal output of the voice application. The user simulation script is processed to generate a simulated output for the voice application corresponding to the nominal output. The user simulation script is also processed to generate a simulated input for the voice application corresponding to a pre-determined user input to the voice application. The simulated inputs and outputs are executed in conjunction with the voice application in real-time to test the application. Execution time rates for text based input and output are calculated based on empirical speaking rates of human users.

Abstract: A method of providing streaming audio to a non-streaming telephony audio interface can include receiving audio segments of a complete audio message and storing the audio segments in memory. A size of the audio segments can be determined. A callback can be registered for each of the audio segments, wherein each registered callback can be provided a timer value associated with the size of the audio segment for which the callback was registered. The complete audio message can be played when an active one of the callback timers expires.

Abstract: In a telephony call management system, a method of distributing calls among a plurality of packet-based load balancing resources can include receiving over a telephony channel a call request specifying call information, comparing the call information with selection criteria to select one of the plurality of packet-based load balancing resources, and querying the selected packet-based load balancing resource to resolve the call request. A response from the selected packet-based load balancing resource can be received which indicates whether the call request can be resolved.

Abstract: A method of providing streaming audio to a non-streaming telephony audio interface can include receiving audio segments of a complete audio message and storing the audio segments in memory. A size of the audio segments can be determined. A callback can be registered for each of the audio segments, wherein each registered callback can be provided a timer value associated with the size of the audio segment for which the callback was registered. The complete audio message can be played when an active one of the callback timers expires.