37results about "Virtual private networks" patented technology

A method and system for secure Voice over Internet Protocol (IP) (VoIP) communications. The method and system provide secure VoIP voice calls, video, Instant Messaging (IM), Short Message Services (SMS), or Peer-to-Peer (P2P) communications while maintaining privacy over the Internet and other communications networks such as the pubic switched telephone network (PSTN) to and from any network device through a virtual private network infrastructure interconnecting private VoIP network devices. The method and system allow a network device to function as an IP private branch exchange (PBX) or a private VoIP gateway and provide and control VoIP voice communications without using other public or private VoIP gateways or VoIP servers or devices on a communications network such as the PSTN or the Internet.

A collaboration services suite is adapted to support a plurality of integrated telecommunications services accessed by geographically dispersed team members using a virtual team environment (VTE) client that generates a graphical user interface (GUI) for each of the respective team members. Communications sessions are automatically set up by the collaboration services suite in response to request messages generated by the VTE client when a team member initiates a communications session request using the GUI. Team members require no knowledge of another team member's communications device address in order to initiate a communications session. The collaboration services suite includes a VTE server that communicates with the VTE clients, a presence engine that collects and maintains a status of communications devices specified in a current profile of the team member; and, a call server for handling setup and control of a voice component of each communications session completed.

A network architecture includes a communication network that supports one or more network-based Virtual Private Networks (VPNs). The communication network includes a plurality of boundary routers that are connected by access links to CPE edge routers belonging to the one or more VPNs. To prevent traffic from outside a customer's VPN (e.g., traffic from other VPNs or the Internet at large) from degrading the QoS provided to traffic from within the customer's VPN, the present invention gives precedence to intra-VPN traffic over extra-VPN traffic on each customer's access link through access link prioritization or access link capacity allocation, such that extra-VPN traffic cannot interfere with inter-VPN traffic. Granting precedence to intra-VPN traffic over extra-VPN traffic in this manner entails partitioning between intra-VPN and extra-VPN traffic on the physical access link using layer 2 multiplexing and configuration of routing protocols to achieve logical traffic separation between intra-VPN traffic and extra-VPN traffic at the VPN boundary routers and CPE edge routers. By configuring the access networks, the VPN boundary routers and CPE edge routers, and the routing protocols of the edge and boundary routers in this manner, the high-level service of DoS attack prevention is achieved.

A collaboration services suite is adapted to support a plurality of integrated telecommunications services accessed by geographically dispersed team members using a virtual team environment (VTE) client that generates a graphical user interface (GUI) for each of the respective team members. Communications sessions are automatically set up by the collaboration services suite in response to request messages generated by the VTE client when a team member initiates a communications session request using the GUI. Team members require no knowledge of another team member's communications device address in order to initiate a communications session. The collaboration services suite includes a VTE server that communicates with the VTE clients, a presence engine that collects and maintains a status of communications devices specified in a current profile of the team member; and, a call server for handling setup and control of a voice component of each communications session completed.

A virtual private network includes an internet protocol (IP) network and a public switched telephone network (PSTN). An enterprise gateway is operably connected to the IP network. The enterprise gateway is operably connected to a switch of the PSTN through a direct access line (DAL). The set-up signaling for virtual private network calls and the calls themselves are transported across the internet protocol network and the public switched telephone network through the direct access line.

A method, system and apparatus are provided for call routing in a telephony network. The telephony network includes at least one voice gateway and is associated with a plurality of Virtual Private Networks (VPNs). The telephony network provider provides conventional telephony network access to all its VPN customers, while providing VPN customers with the freedom to manage call routing/processing in their networks. This is done by co-locating the voice gateway with the Provider Edge (PE) router of the provider network and registering the voice gateway with the routing entity in each VPN.

A point-of-presence (POP) call center system capable of answering, servicing, queuing and routing of calls at local points of presence to reduce communications costs and enhance operational efficiency for toll-free inbound call centers. The POP call center system of the includes a set of point-of-presence call center gateways distributed at points of presence close to the point of call origination that are connected by a virtual private network to premises call center gateways at business locations where the call centers reside.

This disclosure concerns systems and methods for network communications. In one example, the method includes: providing subscriber terminals of each network with a respective publicly-known telephone number for allowing communications to be addressed to each subscriber terminal; associating selected subscriber terminals of each network together in a group such that those subscriber terminals can address communications to one another using a respective code allocated to each of the selected subscriber terminals; and, maintaining a database of the codes and corresponding publicly-known telephone numbers of all of the networks and providing to one of the networks a publicly-known telephone number corresponding to a code of a subscriber terminal of another one of the networks to allow communication from a subscriber terminal of one of the networks to be addressed to a subscriber terminal of the other of the networks using the code of the latter subscriber terminal.

A communication platform for providing computer/telephony integration services to remote subscribers comprises: a hosted call switching unit in communication with an external telephone network or interconnected networks through a communications trunk; for each subscriber, a subscriber telephony component, such as an intelligent agent, executed by processing means belonging to the communication platform and connected to an external subscriber's information system through a permanent private secure data channel, whereby said subscriber telephony component can communicate in a private manner with other information system components of said subscriber so as to be logically part of said information system, each subscriber component being capable of controlling said switching unit according to subscriber data; resources available to each subscriber telephony component in association with call processing or routing; means for allocating resources to each telephone call handled by a subscriber telephony component in response to data communication with said component through a secure interface.

A method which allows standard telephone users to audio conference with video conferencing participants over IP networks in a private secure environment. A dial-out is performed from one or more conference client terminals bridging audio between the Internet and the PSTN networks. The process uses a mixed mode hybrid network architecture for call set-up, initialization and teardown including the method to mix audio at the desktop terminal instead of in a general purpose server as in the prior art. The method conferences video and audio between multiple clients and include audio from a standard telephone network within the conference. A virtual private network connects all of the IP clients together including the voice over IP server used to transcode the proprietary audio into the H.323 standard for transport into the telephony network.

A system and method to provide secure access across the untrusted public switched telephone network is described. The system and method can be initiated by a security policy defining actions to be taken based upon at least one attribute of the call.

Setup of a Voice over Internet Protocol (VoIP) call is initiated and an authentication token is received for the VoIP call that is set up, that indicates that the VoIP call is authorized. The authentication token is inserted into packets for the VoIP call. The packets, including the authentication token therein, are transmitted into an IP network. The authentication token may be placed in an IP version 6 (IPv6) flowID field.

The invention discloses an enterprise telephone exchange platform and a call processing method. The enterprise telephone exchange platform comprises an enterprise telephone exchange service control point, an enterprise telephone exchange service exchange point and an intelligent peripheral, wherein the enterprise telephone exchange service control point comprises a basic call processing module and a service capability enhancement module; the basic call processing module is used for judging whether current network exchange equipment can support a call triggered by an exchange network service exchange point or not and transmitting the call to the service capability enhancement module when judging that the current network exchange equipment cannot support the call triggered by the exchange network service exchange point; and the service capability enhancement module is used for controlling the intelligent peripheral and the enterprise telephone exchange service exchange point to perform the service associated with the call. Through the technical scheme, compared with an enterprise telephone exchange system based on an intelligent platform technique, the enterprise telephone exchange platform provides larger system capacity and reduces network weaving expenditure.

An illustrative intelligent network and method for providing voice telephony over Asynchronous Transfer Mode (“ATM”) and private address translation are provided that can provide significant advantages. The method includes generating an input ATM setup message at the calling party CPE that includes a VToA designator and a called party phone number, extracting information from the input ATM setup message such as the VToA designator and the called party phone number, analyzing the information, designating an ATM address of a called party CPE to be stored in the first parameter of an output ATM setup message, determining if private address translation is needed, designating the ATM address of the called party CPE to be stored in a first instance of the second parameter of the output ATM setup message, designating an ATM address of an egress ATM edge switch to be stored in the first parameter of the output ATM setup message, and generating an output ATM setup message. The method also includes extracting information from the output ATM setup message such as the ATM address of the called party CPE, designating the ATM address of the called party CPE that was stored in the first instance of the second parameter of the output ATM setup message to be stored in the first parameter of a destination ATM setup message, and generating a destination ATM setup message that includes the ATM address of the called party CPE stored in the first parameter and the called party phone number value stored in the second parameter. An illustrative intelligent network for providing VToA and private address translation is also provided.

The invention relates to an IP concentration user exchanging communication system which is used for establishing communication connection for a group of concentration users. Wherein, the group comprises a mobile wireless user and a mobile station of the mobile wireless network, and the system comprises a network device which is arranged in the grouping exchanging network, and comprises a computer program product for providing the concentration user exchanging function for the concentration users. Wherein, the network device is connected with the mobile exchanging center of the mobile wireless network through a first signaling route guided by a first network boundary element and a second signaling route guided by a second network boundary element. The mobile exchanging center generates the first signaling information when establishing the communication connection by the mobile station and transmits to the network device through the first signaling route. The computer program produce generates the second signaling information after obtaining the first signaling information and transmits to the mobile exchanging center through the first signaling route, thereby affecting the route of the communication connection.