Abstract: A framework in a cloud network that may allow for debugging at multiple vantage points at different layers (e.g., layer 2, layer 3, etc.). The methods may provide tracer or measurement services that filter, capture, or forward flows that may include packets, calls, or protocols to look for particular signatures.

Abstract: A system, method, and computer program product are provided for generating event tests associated with a testing project. In use, an indication of one or more event types is received for creating one or more events on which to perform event testing associated with at least one testing project. Additionally, an indication of one or more parameters associated with the one or more events on which to perform the event testing is received. Further, one or more event files are generated for testing the one or more events based on the one or more event types and the one or more parameters. In addition, the one or more event files are sent to one or more event processing systems for testing the one or more events, the testing of the one or more events including storing information associated with all faults resulting from the testing of the one or more events in a risky events repository that is capable of being utilized to generate testing rules for additional testing projects.

Abstract: The determination of user perceived delay in audio and video conferencing systems/services is described. Example embodiments consistent with the present invention measure delays in audio and/or video conferencing systems and services, by treating such systems and services as black boxes, thereby providing end-to-end delays as they would be perceived by users. For example, the user perceived round-trip delay in a video conference system/service may include time delays from each of video capture, video encoding, video transmission, video decoding and video rendering.

Abstract: A speech analytics system (“SAS”) monitors speech from a remote party and agent to determine when text-based guidance information (“TBGI”) and a script control icon is to be provided to the agent. The agent may respond to a particular context with certain information, either by verbally informing the remote party or causing a prerecorded script to be played to the remote party. In one embodiment, the SAS monitors the agent's speech to ascertain if an expected speech response is provided by the agent within a first time period. If not, the SAS may cause another TBGI and script control icon to be presented to the agent, and again monitor the agent's speech to ascertain if an expected speech response is provided by the agent within a second time period. If the agent again fails to respond properly, then data may be written indicating a compliance failure.

Abstract: A call calculator and method thereof used to automate testing of a voice self service platform. The call calculator may have functional logic to calculate an estimated call load. The type of load may be least one of cyclic, peak, burst, choppy, and sustained. The call calculator may use a set of input parameters of an algorithm to calculate the estimated call load, wherein the algorithm may be based on an integral calculus model.

Abstract: A device may receive an indication to test a call flow associated with an interactive voice response (IVR) system. The device may identify an input associated with the call flow, and may identify a first response associated with the call flow. The first response may be determined based on providing the input to the IVR system on a previous call. The device may initiate a test call with the IVR system. The test call may be initiated after the previous call, and may cause the input to be provided to the IVR system and a second response to be received from the IVR system. The device may identify the second response, and may compare first audio information, associated with the first response, and second audio information associated with the second response. The device may determine a test result based on the comparison, and may provide the test result.

Abstract: A call calculator and method thereof used to automate testing of a voice self service platform. The call calculator may have functional logic to calculate an estimated call load. The type of load may be least one of cyclic, peak, burst, choppy, and sustained. The call calculator may use a set of input parameters of an algorithm to calculate the estimated call load, wherein the algorithm may be based on an integral calculus model.

Abstract: A method for monitoring voice quality in a communication network. In a first embodiment, the method comprises the steps of: establishing a voice communication pathway (VCPW) between first and second telephone terminals; establishing speech cadence between first and second terminals, wherein the terminals alternate between opposing send and listen/receive modes such that when one of the terminals is sending a voice test message the other terminal is either listening for or receiving the voice test message and vice versa; and calculating a voice quality score for one or more received voice test messages. In a second embodiment, synchronous cadence is established between a plurality of telephone terminals and a voice recognition server (VRS).

Abstract: A device simulates an emergency services test call based on an automatic number identification (ANI) to be tested, and generates a test notification associated with the emergency services test call. The device also automatically initiates the emergency services test call, receives results of the emergency services test call, and generates an emergency services call information report based on the results of the emergency services test call.

Abstract: A device includes a traffic load profile containing information for regulating rates of routing calls and a controller. The controller may be configured to access the traffic load profile, obtain network calls from a traffic load producer, and route the network calls to a system under test based on the traffic load profile.

Abstract: A device may automatically place a call to a call routing device based on one of a list of numbers or addresses. The call may be forwarded by the call routing device to a call agent device. The device may also establish a session with the call agent device to which the call is forwarded by the call routing device and automatically receive, from the call agent device, a message that includes an identifier for a virtual call agent. The device may extract the identifier from the received message, compare the extracted identifier to an identifier that is associated with the call, and determine whether the call routing device correctly forwarded the call based on the comparison.

Abstract: The invention provides a test phone whereby the test phone comprises means to communicate to a server, means to generate SIP signaling towards the server, whereby the test phone by means for SIP signaling may request the server to set-up a test call using specific resources for routing said test call.

Abstract: A system for remote control of equipment such as office equipment is provided. The system involving a telephone interface and a control system. A method is provided for remote equipment control having steps of installing and configuring equipment. A method is provided controlling equipment remotely having steps relating to using elements of a control system.

Abstract: A system and method that may include placing a first telephone call from a data network to a target phone number using a trusted carrier; recording the first telephone call in a reference audio data file; placing a second telephone call from the data network to the target phone number using a suspect carrier; recording the second telephone call in a test audio data file; and comparing the test audio data file to the reference audio data file.

Abstract: A device includes a traffic load profile containing information for regulating rates of routing calls and a controller. The controller may be configured to access the traffic load profile, obtain network calls from a traffic load producer, and route the network calls to a system under test based on the traffic load profile.

Abstract: A call routing test and verification system for an interactive voice response (IVR) system including an automated test driver (ATD) for connecting to a IVR system over a public switched telephone network (PSTN) configured to provide signals representative of content of utterances appropriate for responding to prompt signals of an IVR menu and a test module to enable a test mode of the IVR system in response to the connection by ATD, wherein the test mode allows the IVR system to transmit a call routing string to the ATD prior to transfer of the connection to one of a plurality of agent terminals in response to reaching a call routing point of the IVR menu, where the ATD is further configured to compare at least a portion of the transmitted call routing string to an expected routing string for the call routing point.

Abstract: Systems and methods for calls per second network testing are disclosed. The methods may be performed by a network testing system or other computing device. The method may include receiving user selection to create a network test. A user specification of a bulk network calls test type and the call transmission criteria for the network test are received. The call transmission criteria may include a number of calls per second and one of either a talk time or a number of channels. When the talk time is specified, the number of channels is calculated. Similarly, when the number of channels is specified, the talk time is calculated. User selection to begin the network test is received, and packets are transmitted for the network test according to the call transmission criteria. During transmission, the call transmission criteria are recalculated at a system defined interval to avoid bursts.

Abstract: A call flow guidance system helps a call center agent navigate the maze of complex call flows that apply to an enormous spectrum of caller issues, and further improves adoption rates and training timelines for the call center agents. Through the system, call center agents easily navigate multiple call flows and view information tailored to improve average handle time (AHT) and first call resolution (FCR). The system provides dynamic content attached to call flows and improve call flow navigation by implementing links that bridge nodes between display sections of a particular call flow and links that cross from one call flow to another.

Abstract: A method and system for testing a Telephony User Interface is disclosed. Voice prompts of the Telephony User Interface are converted into tone prompts that are representative thereof. Each tone prompt can have a predetermined frequency and/or duration, so that it is readily recognizable by a Telephony User Interface tester. Thus, automation of the testing of Telephony User Interface is enhanced.

Abstract: Methods and apparatus to handle support contacts in a call center are disclosed. An example method for handling a first contact to a support center includes receiving the first contact from a first person, receiving a first identifier associated with the first person, analyzing contact details for a second contact associated with a second identifier associated with a second person different from the first person to determine if a problem associated with the first identifier exists, and indicating that a problem associated with the first identifier exists if the contact details for the second contact associated with the second identifier indicate that a problem associated with the first identifier exists.

Abstract: A method for testing an assigned Location Routing Number (LRN) that is assigned to a first switch is provided according to the invention. The method includes in the first switch generating a LRN test telephone call to a second switch. The LRN test telephone call is targeted to the LRN assigned to the first switch. The method further includes indicating that the LRN is being incorrectly routed in the second switch if the first switch does not receive the LRN test telephone call back from the second switch. The method further includes indicating that the LRN is being correctly routed in the second switch if the first switch receives the LRN test telephone call back from the second switch.

Abstract: A method and apparatus are disclosed that provide a technique for testing a data-processing system that enables communications between two or more telecommunications endpoints. An example of such a data-processing system is a Session-Initiation Protocol (SIP) proxy server. In particular, a test-execution system, in accordance with the illustrative embodiment of the present invention, transmits test-related signals to and receives test-related signals from one or more endpoints, such as SIP-enabled telephones. Some of the transmitted test-related signals direct an endpoint to transmit a message, such as a session-initiation request, to the data-processing system under test. Some of the received test-related signals comprise information from an endpoint, such as whether or not a message has been received from the data-processing system under test.

Abstract: Methods, apparatus, and computer program products are provided for testing a Voice Over Internet Protocol (VoIP) communication system. A VoIP test phone call is generated through an Internet communication link to a VoIP service provider for a called telephone number. Success of the VoIP test phone call is determined. The success of the VoIP test phone call may be determined based on whether it is established with communication equipment at the called telephone number within a threshold time, based on whether it is established within a threshold number of attempts, and/or based on a measured quality of service in the VoIP test phone call when it is established. Further VoIP test phone calls to test the VoIP communication system can be scheduled.

Abstract: A dial testing system and a dial testing method using the dial testing system are provided. The dial testing system includes a dial testing agent device adapted to perform an automatic dial testing through a network element and control an automatic dial testing result reported by the network element; and a dial testing server adapted to collect the automatic dial testing result reported by the dial testing agent device. The dial testing system and the dial testing method provided by the present invention may assure allsidedness of a dial testing, enhance the speed and quality of the dial testing operation, and save a large amount of manpower and material resources.

Abstract: A test emergency response location (ERL) is used to exercise an emergency response (ER) service. The test ERL does not have any connections to the real external PSAP or OSAN and only routes the call to synthetic phones or agents. A 911 call is intercepted and routed to an ER application to make a decision on which emergency location identification number (ELIN) to use. The ER application recognizes the call originator as a synthetic phone or agent and routes it through the test ERL. In one embodiment, the test ERL is configured so that a second synthetic phone or agent is registered as the PSAP endpoint and a third synthetic phone or agent is registered as the OSAN endpoint. In a properly functioning system, the ER application forwards the call to the second and third agents. When the second and third agents receive the calls, the system is considered to be functioning properly.

Abstract: A method and system for verifying the connection-related communications data registered by a network node, where the connection-related communications data are used for the calculation of charges includes establishing at least one predetermined test communication connection. By clearing the predetermined test communication connection via the network node and registering a time of an event, the system and method registers the time of at least one additional event for calculating charges at a predetermined measuring point lying outside of the network node. A systematic temporal measuring error is determined. A reference data record containing time-related data and a connection-related communications data record generated is generated. The method corrects the time-related data of the reference data by the ascertained systematic temporal measuring error and compares the reference data record to the connection-related communications data record to determine if a difference lies within a predetermined range.

Abstract: Performance parameters of a switch may be ascertained by observing its behavior during a brief interval of high call volume. This may be achieved by accumulating call messages on a queue before allowing them to be processed together as a group. Without increasing the rate of incident calls, an accumulation of messages is accomplished by altering the amount of time that elapses between transferring the accumulation to a processing queue. In one implementation, test calls are accumulated in call input queue. Ordinarily, these calls are transferred to a second queue for processing every few milliseconds (the transfer period). By increasing the transfer period, an increased number of calls are stored in the call input queue before being transferred to the second queue. This allows a larger number of calls to be processed by the telephone switch in a given cycle and, hence, allows the switch's operation under increased call volumes to be observed.

Abstract: The invention includes a system and a method for testing a communication network. The inventive method transmits a first signal from a first point to a second point of the communication network, where the first and the second points are remotely located. The inventive method further includes recording a first time value of the transmitting of the first signal using a first clock, receiving a second signal at the second point of the communication network, and recording a second time value of the receiving using a second click. The first clock and the second clock operate from a substantially similar reference. The method may further include comparing the first signal and the second signal as a function of the first and second time values. Also, the method may determine at least one performance characteristic of the communication network based on the comparison.

Abstract: After implementation of an interactive voice response (IVR) system providing voice prompts (i.e., utterances) to aid a telephone caller, it is desirable to provide call-flow verification to validate system accuracy, particularly under high-volume or saturation calling conditions. An IVR system has a call-flow verification (CFV) mode which can be activated for this purpose. In the CFV mode, the IVR system provides prompt signals which include coded signals representing the content of utterances. An automated call generator (ACG) unit places simulated user calls to the IVR system. The content of utterances is represented by coded signals included in prompt signals sent by the IVR system during the course of the simulated call and stored by the ACG unit. Verification can be applied for all possible paths an application might take (to check all possible go-right and error paths).

Abstract: The invention includes a system and a method for testing a communication network. The inventive method transmits a first signal from a first point to a second point of the communication network, where the first and the second points are remotely located. The inventive method further includes recording a first time value of the transmitting of the first signal using a first clock, receiving a second signal at the second point of the communication network, and recording a second time value of the receiving using a second clock. The first clock and the second clock operate from a substantially similar reference. The method may further include comparing the first signal and the second signal as a function of the first and second time values. Also, the method may determine at least one performance characteristic of the communication network based on the comparison.

Abstract: The invention is a method and system for remotely testing network paths diagnosing problems in establishing a communications link between a call-routing component and a target component in a telecommunications network environment. The test system permits a call to be originated at any end office from a remote location. Originating a call at a customer's end office accurately recreates the customer's experience. A test method accurately reveals problems with a specific communications path. The last mile of a customer's connection can also be tested.

Abstract: A method for generating and distributing telephonic traffic. The method includes receiving a call profile from a user interface, generating a traffic distribution from the call profile. The method then creates a traffic profile from the traffic distributions and receives at least one call hold time from a user interface, and then transmits the traffic profile and the at least one call hold time to a call generator. Also included is a traffic distribution generator. The traffic distribution generator includes a traffic graphical user interface, user input controls on the traffic graphical user interface and a processor to generate traffic distributions and the distribution and create a traffic profile from the distributions, and a port operable to transmit the traffic profile to a bulk call generator.

Abstract: A simple means is provided to test the cross connect wiring of a communications system which uses a POTS splitter to separate broadband and narrow band signals. A suitable frequency test signal such as 1 KHz tone is fed at sufficient amplitude to the broadband port of the POTS splitter, so that even after attenuation by both the POTS splitter capacitance and inductance filters, it is detectable by suitably sensitive detectors connected to the narrow band port of the POTS splitter. The test signal may be applied intermittently to enable its detection even in low signal to noise situations.

Abstract: A method for placing a test call in a telecommunications network. The method includes retrieving an indication of a translation change for a telecommunications switch from storage and placing a test call to test whether the translation change was successful.

Abstract: A bulk call generator is configured for measuring the response time of a telephony-based interactive response system by detecting the presence of prescribed signaling messages and in-band prompts, where the in-band prompts include tones (such as DTMF tones) recognizable by the bulk call generator. In particular, the tones enable the bulk call generator to uniquely identify a corresponding message, for example a test greeting generated by the telephony-based interactive response system, or a test message retrieved for playback by the telephony-based interactive response system. Hence, the bulk call generator can measure the response time across all channels of a telephony-based interactive response system in an economic and scalable manner, for verification that prescribed prompts and/or messages are played by the telephony-based interactive response system.

Abstract: A system and method for measuring latency in voice communications is presented. In an exemplary embodiment, latency is measured by establishing a call between a first and a second telephony device, and measuring a latency between a signal originating at the first telephony device and the signal as it arrives at the second telephony device.

Abstract: A method and associated apparatus for testing telephone system operation. The method comprising connecting a call agent simulator to a network. Call actions or events are applied from the call agent simulator to the network expected to produce expected results in the network. The telephone system operation is verified by comparing the actual results produced to the expected results. In another aspect, an IP network or a consumer premises equipment is verified using the call agent simulator. In another aspect, the telephone system is a Voice over Internet (VoIP) system.

Abstract: A telephone call simulator generates telephone calls for large sets of subscribers in a PSTN-IP telephone service for purposes of validating various telephone services, e.g. call waiting, call forwarding, etc. available to subscribers of the telephone service. via the IP network. The telephone numbers of subscribers are grouped in different call processing actions and/or subscriber call behavior based on the options offered by the service. The telephone numbers and test logic scripts are stored in a test input data file. Test logic scripts are generated for each group of related telephone number and stored in a file, typically the test input data file. The test input data file is read by test assignment apparatus which assigns the test logic scripts to process within a call processing pool. The test logic scripts record test result data to a test output data file.

Abstract: A prepaid service assurance system provides service assurance for a prepaid telecommunication system by generating a call to the prepaid telecommunication system. In response to the call, the prepaid service assurance system then receives an actual response from the prepaid telecommunication system and automatically compares the actual response with an expected response for the prepaid telecommunication system. In other embodiments, the prepaid service assurance system determines whether a modification action is needed based on the comparison of the actual response and the expected response. If the modification action is needed, the prepaid service assurance system identifies the modification action based on the determination that the modification is needed. Once the modification action is identified, the prepaid service assurance system transmits an instruction to provide the modification action for the prepaid telecommunication system.

Abstract: A method and apparatus for measuring call setup time is presented. A first device in a network sends a signal to a second device. The first device monitors for a response from the second device and adjusts an answer tone setting for the second device based on the response. After which, the first device calculates an elapsed time between sending the signal and a receipt of the response.

Abstract: A prepaid service assurance system provides service assurance for a prepaid telecommunication system by generating a call to the prepaid telecommunication system. In response to the call, the prepaid service assurance system then receives an actual response from the prepaid telecommunication system and automatically compares the actual response with an expected response for the prepaid telecommunication system. In other embodiments, the prepaid service assurance system determines whether a modification action is needed based on the comparison of the actual response and the expected response. If the modification action is needed, the prepaid service assurance system identifies the modification action based on the determination that the modification is needed. Once the modification action is identified, the prepaid service assurance system transmits an instruction to provide the modification action for the prepaid telecommunication system.