Abstract: According to various embodiments of the present invention, systems and methods for routing telephone calls from host switch to remote switch in a Voice Over Internet Protocol (“VoIP”) network. A host switch in the VoIP network may be communicably coupled with a plurality of remote switches and with a public switched telephone network (“PSTN”); the host switch may receive a call from PSTN having signaling that includes a called party number field and/or a location routing number field. Based on the called party number field and/or the location routing number field, host switch identifies a target remote switch from among the plurality of remote switches that is associated with the called party number field by, for example, querying a database of associations between location routing numbers and/or telephone numbers and remote switches. Host switch then sends the call to identified target remote switch.

Abstract: A calling network capable of accepting voice and data information includes a voice distribution server, wherein the voice distribution server is communicably coupled to an integrated access device, wherein the voice distribution server is associated with a computer readable medium, and wherein the computer readable medium includes a customer profile; and wherein the customer profile includes at least one registered access number associated with the integrated access device, and at least two quasi-registered access numbers associated with the integrated access device.

Abstract: sessions conducted over a communications network, such as a voice over Internet protocol (VoIP) network, that employs a session initiation protocol (SIP). The method includes receiving a subscription request from a third party subscriber, the subscription request identifying a targeted user to be monitored, monitoring communication sessions in which the targeted user is a participant, to detect SIP events corresponding to state transitions associated with the communication sessions.

Abstract: An objective, service-level specific voice call listening quality test scheme for a Voice Over IP (VOIP) network is presented. Test probes are deployed along the border of the VOIP network. Each test probe is capable of placing calls over the VOIP network to the other test probes at different levels of service and measuring call quality using an objective measurement algorithm such as PAMS or PSQM. The measurement results are collected on an ongoing basis to obtain information on the VOIP network's voice call quality. The information is compared to thresholds to measure performance against Service Level Agreement guarantees.

Abstract: A network (100) includes multiple gateways (135, 140) and a router (105) connected to at least one of the multiple gateways (135, 140). The router (105) is configured to receive packets that include multiple first virtual circuit identifiers associated with the multiple gateways in the network (100), assign second virtual circuit identifiers to the at least one connected gateway, and initiate transmission of a message to the at least one connected gateway informing the gateway of the first virtual circuit identifiers.

Abstract: A system for automated installation of a communication line using an optimal route between a source location and a destination location is disclosed. The system includes a routing engine providing an automated design process for rendering the optimal route. The routing engine utilizes a routing algorithm to select the optimal route from a graph of capacity links defining a plurality of possible routes between the source and destination locations. If, at any time during the design process, the optimal route or capacity links defining the optimal route are detected as unavailable for any reason, the routing engine re-initiates the design process and thereafter selects a new optimal route based upon a new capacity graph built without the previously unavailable capacity link. Once designed, the available optimal route is provided to a command and control engine, which, in turn, manages the installation of the communication line using the optimal route.

Abstract: A method for determining network topology of a provider network includes selecting a first network element, selecting a first port on the first network element, and iteratively performing connectivity validation tests using the first port, wherein each connectivity validation test is associated with a type of network element and yields a result that indicates whether a second port on a second network element of the associated type is connected to the first port. A system for discovering topology of a network, the system comprising a topology discovery engine in operable communication with a near network element and operable to identify a first port of a far network element that is connected to a second port of the near network element by remotely altering operation of the near network element to cause the second network element to respond in a manner that identifies the first port.

Abstract: Embodiments of a network architecture include a backbone node having a plurality of independent routers or switches connected in a matrix, wherein the matrix includes a plurality of stages of routers or switches, to form a node having a node switching capacity that is greater than the node switching capacity of the individual routers or switches. A method includes assigning one of a plurality of backbone networks to a destination network address, associating a next hop loopback address with the destination network address, and advertising the destination network address in combination with the next hop loopback address through the selected backbone network address.

Abstract: An open architecture platform bypasses data from the facilities of a telecommunications carrier, e.g. an incumbent local exchange carrier, by distinguishing between voice and data traffic, and handling voice and data traffic separately. An SS7 gateway receives and transmits SS7 signaling messages with the platform. When signaling for a call arrives, the SS7 gateway informs a control server on the platform. The control server manages the platform resources, including the SS7 gateway, tandem network access servers (NASs) and modem NASs. A tandem NAS receives the call over bearer channels. The control server determines whether the incoming call is voice traffic or data traffic, by the dialed number, and instructs the tandem NAS how to handle the call. Voiced traffic is transmitted to a switch for transmission from the platform. Data traffic is terminated at a modem NAS, where it is converted into a form suitable for a data network, such as a private data network or an Internet services provider (ISP).

Abstract: An embodiment of a method for use in performing interoperability testing between two network architectures includes forming multiple transaction signatures that each characterizes a communication transaction conducted over the two networks and analyzing each of the transaction signatures to determine if the transaction signature matches a previously classified pattern. The method may further include generating a report identifying each of the communication transactions and indicating that each of the communication transactions is classified with a classification associated with a previously classified pattern or are of unknown classification. Another method includes receiving a transaction signature composed of multiple transaction elements, and determining whether the received transaction signature corresponds to a previously identified transaction signature pattern.

Abstract: An objective, service-level specific voice call listening quality test scheme for a Voice Over IP (VOIP) network is presented. Test probes are deployed along the border of the VOIP network. Each test probe is capable of placing calls over the VOIP network to the other test probes at different levels of service and measuring call quality using an objective measurement algorithm such as PAMS or PSQM. The measurement results are collected on an ongoing basis to obtain information on the VOIP network's voice call quality. The information is compared to thresholds to measure performance against Service Level Agreement guarantees.

Abstract: A method and system is provided for performing inbound call screening in a packet-based network, such as an H.323 Voice over IP (VoIP) network. The inbound gateways on the network are registered with inbound gatekeepers, and standard messages are used between an inbound gateway, an inbound gatekeeper and an inbound screening database to decide: whether an inbound call to a particular called number (DID) is to be allowed into the network; whether the called number should be translated into a different called number; and whether a routing index should be included in the called number to indicate the destination of the call.

Abstract: A system for determining packet quality on an Internet protocol-based (IP-based) network includes a packet test device that traces multiple paths on the IP-based network to obtain node identifiers corresponding to nodes on each path, and an Internet quality monitor (IQM) that associates a quality metric with each path, and associates an identified link between nodes in a path with one of the quality metrics. A method for monitoring quality of data on an Internet protocol (IP)-based network includes deriving a network topology corresponding to a portion of the IP-based network, determining a quality metric associated with a path in the network topology, deriving a link in the path, and determining a quality metric associated with the link based on the quality metric associated with the path.

Abstract: A graphical user interface and method for customer centric network management. The graphical user interface includes a first view showing text input by a user indicative of customer identification data corresponding to a customer in a network. This data is received by a server connected to the user. The server retrieves stored information corresponding to the customer identification data from a database and provides the stored information and actual circuit path information corresponding to a customer service based on the stored information to the graphical user interface. A second view of the graphical user interface shows the stored information, and a third view shows a graphical representation of a customer path corresponding to the actual circuit path information.

Abstract: A system for testing a software client or hardware module in a voice-over-IP (VoIP) network path includes a VoIP client, a test tool, and an analyzer. The VoIP client, connected to the VoIP network, makes a series of calls to a destination. The calls activate the software client or hardware module under test. The test tool, which has access to the network, plays a series of audio files over the VoIP during the calls. The test tool also polls the software client or hardware module over the network for status information after the calls are completed. While the audio files are being played, the analyzer records the audio files at the destination for measurement the quality of the calls. Accordingly, the quality of voice connection on the VoIP network can be measured, and specific components on the network can be tested.

Abstract: A computer system includes an input device (210) and at least one processor (205). The processor receives (600) an electronic document that includes data representing a telephone number. The processor further receives input via the input device selecting (615) a telephone number from the electronic document. The processor additionally signals (725), via a packet-switched network, a telecommunication system to connect a call between the telephone number and a calling party telephone number in response to the selection of the telephone number.

Abstract: Ethernet-based networks for routing Internet Protocol (IP) traffic between source and destination sites. One embodiment includes a plurality of discrete data transmission backbones between the source and destination sites. The source site includes control means for distributing IP traffic at the source site to the plurality of backbones for transmission to the destination site.

Abstract: According to various embodiments of the present invention, systems and methods for collecting, analyzing, and/or displaying signaling information for a call through a network, including: receiving a request for signaling information, wherein the request identifies a particular call handled by at least one network element within a voice over internet protocol network, querying the at least one network element for signaling information related to the particular call, and displaying the signaling information for the particular call. According to some embodiments of the present invention, systems and methods may further include storing signaling information on a network element, generating a call detail record based on the signaling information, wherein the call detail record identifies the network element and the particular call, sending the call detail record to a database, and searching the database based on the one or more criteria to identify the at least one network element.

Abstract: According to various embodiments of the present invention, systems and methods for verifying the routing of a call through a network, including: generating an outbound call from a test tool in a first network, the outbound call including a test key and a destination number, the destination number being associated with a communication device accessible to first network, routing the outbound call to second network, receiving outbound call from second network in the form of inbound call to first network, the inbound call including the test key and the destination number, accessing the test key, and based at least in part on the test key, routing the inbound call to the test tool. In some instances, call tracking information may be collected, analyzed, and/or displayed.

Abstract: A method and system is provided for performing inbound call screening in a packet-based network, such as an H.323 Voice over IP (VoIP) network. The inbound gateways on the network are registered with inbound gatekeepers, and standard messages are used between an inbound gateway, an inbound gatekeeper and an inbound screening database to decide: whether an inbound call to a particular called number (DID) is to be allowed into the network; whether the called number should be translated into a different called number; and whether a routing index should be included in the called number to indicate the destination of the call.