The present invention concerns an interconnect device and process for connecting a first network to a second network, comprising first connecting means for connecting the interconnect device to the first network, second connecting means for connecting the interconnect device to the second network, the second connecting means being connected to the first connecting means. The first connecting means and second connecting means are intended to permit a communication between a first terminal located on the first network and a second terminal located on the second network. The interconnect device detects a voice communication termination by the second terminal. If the voice communication has been initialized by the first terminal, it starts a delaying period for delaying the sending of an indication of the termination to the first terminal for allowing a terminal located on the second network to resume the voice communication before the end of the delaying period.

An IP telephony system obtains control over a group of telephone numbers that can be assigned to its customers' telephony devices under a contract with a first local exchange carrier that originally controlled the group of telephone numbers. The first local exchange carrier assigns one or more location routing numbers to the group of telephone numbers, and the one or more location routing numbers are for the exclusive use of the IP telephony system. The one or more location routing numbers are then recorded against the group of telephone numbers in a local number portability database. Other exchange carriers are then caused to route calls directed to the any telephony number associated with the one or more location routing numbers directly to the IP telephony system. As a result, a second local exchange carrier can terminate telephony communications directly to the IP telephony system, bypassing the first local exchange carrier.

A system, server, and method of DTMF detection in a VoIP network.

To enable internet softphone users to join to calls hosted at an audio conferencing bridge, a SIP Invite may be received at a bridge interface layer of an audio conferencing system. Parameters of the SIP Invite may be extracted, e.g. from one or more X-header fields and processed to enable the internet softphone to be joined to an audio conferencing bridge. The bridge interface layer may receive a SIP Refer message from the audio conferencing bridge that indicates a second audio conferencing bridge external to the system. The bridge interface layer may forward the SIP Invite to the second audio conferencing bridge and the RTP media may be established through the session border controller (SBC) of the audio conferencing system.

A method, system and computer program product are described for managing network communications to a Session Initiation Protocol (SIP) server capable of SIP processing using a SIP stack. A data packet is received from a network device. It is determined, from the data packet, whether the network device is a device recognized by the SIP server. Responsive to this determination, and before SIP processing using the SIP stack, it is determined whether the data packet conforms to a permitted configuration. The permitted configuration includes that data of the data packet indicates an unfragmented User Datagram Protocol (UDP) packet and that data indicative of SIP data in the received data packet matches a parsing rule. If the data packet conforms to the permitted configuration, it is passed to the SIP stack, if not it is discarded.

A system includes a first networking devices, a multiplexer, a second networking device, and a third networking device. The first networking device includes a pair of ports operational up to a first throughput. The multiplexer includes an input port connected to a port of the first networking device, and a pair of output ports. The second networking device includes an output port connected to a port of the first networking device and to an output port of the multiplexer, and operational up to a second throughput greater than the first throughput. The third networking device includes an output port connected to an output port of the multiplexer, and operational up to a third throughput no greater than the first throughput.

Implementations of the present disclosure involve an apparatus and/or method for a regional independent tandem switch of a telecommunications network. The tandem switch processes communications between a long distance carrier and a local exchange carrier through a voice over IP (VOIP) network. By utilizing the VOIP network, the regionally independent tandem may process long distance communications to and from any point in the network, regardless of the physical proximity of the tandem to the originating/destination communication device. The regionally independent nature of the tandem also allows for flexibility in communication routing through the network, load balancing between the network tandem switches and reduction of needed components of the network for proper processing of the long distance communications.

Systems and methods for data transport, including receiving one or more signals into a reconfigurable and flexible rate shared rate multi-transponder network architecture, wherein the network architecture includes one or more transponders with multiple line side interfaces and one or more client side interfaces. The transponders are configured to map one or more signals to multiple parallel Virtual Ethernet Links, remove idle characters from the one or more signals, buffer one or more blocks of characters using an intermediate block buffer, activate and deactivate one or more portions of input/output electrical lanes of an Ethernet module, multiplex and demultiplex the one or more signals to and from the input/output electrical lanes to enable sharing of a single optical transceiver by multiple independent signals, and insert blocks of idle characters to enable transmission over a lower rate transmission pipe.

A method for combining Internet protocols in a Differentiated Services model environment is described. The Session Initiation Protocol (SIP) and Common Open Policy Service (COPS) are combined together to provide methods of setting up a session and tearing down a session, while maintaining Authentication, Authorization, and Accounting (AAA) policies. The Open Settlement Policy (OSP) is also combined with SIP and COPS. This combination provides for an interchange of parameters between session setup, teardown, authorization, policy, Quality of Service (QoS), and usage reporting

A method of establishing a communication link between a mobile terminal of a wireless network and a subscriber of a network, such as an enterprise network, and/or a residential network.