Abstract: A bridge for multi-network communications is provided. A voice bridge includes an open settlement protocol (OSP) compliant interface for establishing inter-domain voice connections. The voice bridge facilitates connections between OSP compliant and non-OSP compliant networks and interfaces with an OSP server. A call set-up component processes requests from OSP compliant and non-OSP compliant networks. An OSP interface facilitates OSP compliant reporting to the OSP server. Other embodiments comprising additional features, such as address masking, are also provided.

Abstract: An apparatus and method for increasing available ports on a voice router is provided. A first gateway and a second gateway are assigned a single port number for a data stream, the direction of packet flow is identified to determine a destination gateway. The destination gateway is one of the first and second gateways, depending on the direction of the packet flow. The packets are then forwarded to the destination gateway. The voice router can further consolidate RTCP streams from a plurality of gateways into a single port on the voice router.

Abstract: A software phone (“softphone”) application enables a user to make a phone call from a computing device user Voice over Internet Protocol (VoIP). While the functionality of the softphone remains the same, its appearance is defined by a skin that can be changed. A server stores a profile for users of the softphone. Based on the user profile, the server selects a skin and pushes the skin to the user's softphone, thereby changing the appearance of the softphone. The skin can include advertisement and can include several layers that can be displayed simultaneously. Some of the layers are static, and some of the layers are animated.

Abstract: An apparatus and method for increasing available ports on a voice router is provided. A first gateway and a second gateway are assigned a single port number for a data stream, the direction of packet flow is identified to determine a destination gateway. The destination gateway is one of the first and second gateways, depending on the direction of the packet flow. The packets are then forwarded to the destination gateway. The voice router can further consolidate RTCP streams from a plurality of gateways into a single port on the voice router.

Abstract: A network address translation apparatus and method within a Voice over Internet Protocol (VoIP) router is described. This apparatus and method operates within networking devices such as routers, switches, bridges, etc. These devices masks both source and destination addresses and may either replace or complement a firewall in providing security. According to one embodiment, network type information, corresponding to the gateways within connection, is embedded in a port number within a packet header. This network type information is extracted and analyzed to identify the network types of both gateways within the connection. According to another embodiment, a port number is assigned according to the network types of both gateways within a connection. Thus, this network type information may be identified by processing the port number on which a packet is received.

Abstract: A system and method for establishing a communication link between an internet telephony device and a PSTN telephone during a user registration period comprises receiving data from a user indicating a plurality of frequently called countries, accessing a gateway provider table comprising data on a plurality of gateway providers that provide service to the plurality of frequently called countries, determining the lowest cost gateway provider for the plurality of frequently called countries, and displaying to the user information on the lowest cost gateway provider for the plurality of frequently called countries.

Abstract: A system and method for managing the transfer of data from a server to a client over an integrated voice and data network limits the bandwidth used for sending non-voice data, such as advertising data, from a server to the client. The limit on the transmission of non-voice data ensures that enough bandwidth is available for the transmission of voice data. The system and method comprise at least one server and at least one client which are communicatively coupled over an integrated voice and data network. A transfer data rate input is sent by a client to a server. The server determines the transfer data rate for non-voice data based on the transfer data rate input. The server then sends the data, preferably non-voice data, to the client at the transfer data rate. The present invention may also comprise an intercept layer and a bandwidth management control module.