摘要:
A real-time interface between the public switched telephone network (PSTN) and an Internet Protocol (IP) network provides voice to data and data to voice conversion between the PSTN and the IP network in a seamless process. The interface, a central communication network, performs Class 5 switching between the PSTN and the IP network, besides providing enhanced services. Receiving a call, the central communication network simultaneously routes the call to a plurality of preprogrammed numbers on the PSTN and on the IP network. The central communication network provides call screening, takes voice messages and converts them to e-mail messages, takes e-mail or facsimile messages and converts them to voice messages. Communication between a PSTN phone on a local PSTN, a computer hooked up to the IP network, a phone hooked up to the IP network by a gateway, a private branch exchange (PBX) on a local PSTN, a wireless communication system with pagers and/or cell phones hooked up to a local PSTN, and facsimile machines on a local PSTN, for example, is provided by the central communication network. Through the central communication network, a computer hooked up to the IP network can exchange voice messages and facsimile messages with a PSTN connected device and conduct conference calling with a plurality of PSTN devices.
摘要:
Audible and signaling information sent through a multi-line hunt group may be transmitted over long distances using packet or IP-enabled communication networks. At least one remote site is connected to the communication network. The remote site includes a switch interconnecting a plurality of subscribers. At least one multi-line hunt group is connected to the switch. The communication system also includes at least one service site connected to the communication network. The service site includes a switch connected to a service platform. At least one multi-line hunt group is connected to the switch. A gateway interfaces each multi-line hunt group and the communication network. Each gateway converts audible information received over communication lines and signaling data received over each signaling line into a data format acceptable by the communication network.
摘要:
A hybrid telephony system with packet switching as well as circuit switching optimizes utilization of transport networks, and is accessible from any conventional telephone set. A call originating from a circuit-switched network is passed through a gateway computer to a backbone packet-switched network, and then through a second gateway computer to a second circuit-switched network where it terminates. The voice of both the originating party and the terminating party is converted to data packets by the near-end gateway computer and then converted back to voice by the far-end gateway computer. In an alternative scenario, the originating party uses a computer on the packet-switched network, which replaces the originating circuit-switched network, and the originating computer. Powered by CPUs, DSPs, ASICs disks, telephony interfaces, and packet network interfaces, the gateway computers may have media conversion modules, speech processing modules and routing resolution modules, and are capable of translating telephony call signaling as well as voice between circuit-switched and packet-switched networks. Optionally, the gateway computers may also have analog trunking modules, MF and DTMF digit modules and special services modules, in order to support analog circuit-switched networks and secure telephone calls.
摘要:
An access architecture for real-time communications is described. The architecture includes an inter-architecture network utilizing a single protocol, a plurality of border elements in communication with the inter-architecture network and with an external network, and one or more call control elements in communication with said inter-architecture network. The external network utilizes any of a variety of known networking technologies and protocols. The inter-architecture network utilizes a single protocol such as SIP. The present architecture provides a single common infrastructure for offering real-time communications services independent of call control protocols and networking technologies.
摘要:
The present invention describes a system and method for communicating voice and data over a packet-switched network that is adapted to coexist and communicate with a legacy PSTN. The system permits packet switching of voice calls and data calls through a data network from and to any of a LEC, a customer facility or a direct IP connection on the data network. The system includes soft switch sites, gateway sites, a data network, a provisioning component, a network event component and a network management component. The system interfaces with customer facilities (e.g., a PBX), carrier facilities (e.g., a LEC) and legacy signaling networks (e.g., SS7) to handle calls between any combination of on-network and off-network callers. The soft switch sites provide the core call processing for the voice network architecture. The soft switch sites manage the gateway sites in a preferred embodiment, using a protocol such as the Internet Protocol Device Control (IPDC) protocol to request the set-up and tear-down of calls. The gateway sites originate and terminate calls between calling parties and called parties through the data network. The gateway sites include network access devices to provide access to network resources. The data network connects one or more of the soft switch sites to one or more of the gateway sites. The provisioning and network event component collects call events recorded at the soft switch sites. The network management component includes a network operations center (NOC) for centralized network management.
摘要:
A unified messaging system, method and user interface is provided for a handheld mobile communication device. The method may include the following steps: (a) providing a unified event listing display comprising: a plurality of summary descriptors, each summary descriptor having one or more information elements to identify the event associated with the summary descriptor; (b) selecting one of the plurality of summary descriptors; (c) displaying a voice mail interface associated with the selected summary descriptor, wherein a plurality of voice mail actions are displayed on the interface; (d) selecting one of the plurality of voice mail actions; (e) communicating a voice mail command associated with the selected voice mail action to a unified messaging server via a wireless data communication network; (f) receiving the voice mail command at the unified messaging server; (g) executing an action associated with the received voice mail command at the unified messaging server; (h) establishing a circuit-switched communication connection to the mobile device from a PBX system associated with the unified messaging server; and (i) performing an action corresponding to the received voice mail command through the established circuit-switched communication connection.
摘要:
A method for reducing latency of VoIP communications while efficiently using network resources and maintaining voice quality. This is achieved by managing packet size on a per-call basis, using factors such as distance between gateways, current backbone network status, service requested or access mechanism for a given call is disclosed. Packet size is selected on a per-call basis based on the distance between endpoints in the call. If the endpoints are far apart, the selected packet size is small. If the endpoints are close together, the selected packet size is large.
摘要:
In an Internet telephone connection method, a communication bandwidth is managed by using a bandwidth controller, gateway devices and voice relay routers to monitor communication quality under bandwidth reservation. A communication path under bandwidth reservation is preferentially selected by using gate keepers and the voice relay routers, and the connection of the Internet telephone is performed. A problematic device is prohibited from being selected as a connection-destination device at the time of a call setup and the problem restoration is monitored by using the gate keepers and the gateway devices. Further, a device having invalid attribute information is prohibited from being selected as a connection-destination device at the time of call-setup, and the restoration of the attribute information is notified to the overall network. Accordingly, the reservation of the communication bandwidth can be performed under the control of the overall network, fixed communication quality can be maintained and reliability can be enhanced.
摘要:
A method for automatically initiating a call, according to which the existing connections from a first user of a telecommunication network to other users are detected and are evaluated regarding the statistical regularities thereof while calls are initiated to the other users based on the identified statistical regularities is provided. Particularly service-based companies to inform their clients in a targeted manner is allowed.
摘要:
A method and system are provided for converging time division multiplexing (TDM) communication networks and packet based networks. Signaling may be monitored between TDM network elements by a Convergence Resource Manager (CRM) to identify calls that may be re-directed over a packet network to an end node with another associated CRM; the re-directed call bypassing much of the TDM network. Installation of a CRPd typically does not require any configuration changes to the TDM network such as point codes. A self learning CRM may also be provided so that a routing database may be automatically built. An originating SS7 message may be modified by a self learning CRM to add a Tag which identifies the originating node. As the modified message traverses the SS7 network, any other self learning CRMS in the network report to the originating self learning CRM that the message has been seen. The last reporting self learning CRM is then associated with the dialed number for future routing of originating calls over the packet network, directly to the last reporting self learning CRM and end network node, thus flattening the TDM network.