Abstract: A method for managing services for a network service provider includes the steps of receiving a Feature Request Message from a switching node associated with a mobile station within a non-circuit switched network at a registration node. The Feature Request Message includes a code for activating a feature within the registration node enabling connections to the non-circuit switched network. A service request is received for providing a network service to the mobile station at the registration node. The service request is routed to the switching node associated with the non-circuit switched network responsive to the feature within the registration node enabling connection to the non-circuit switched network being activated.

Abstract: A cellular communication system comprises a core network, a first access network not supporting circuit switched services and a second access network supporting circuit switched services. A user equipment detects that an application using circuit switched communication is initiated and accordingly initiates a reselection from the first to the second access network. The user equipment transmits a location update message to a circuit switch mobile switching center of the core network which in response notifies a mobility management entity for the first access network that the access network reselection has been initiated. In response to the notification the mobility management entity initiates a change of a routing association from being with the first access network to being with the second access network. The invention may e.g. allow improved circuit switched fallback for a 3rd Generation Partnership Project cellular communication system comprising an access network supporting only packet switched services.

Abstract: A method (and corresponding equipment) for use by a device (10) in establishing a network connection on behalf of an application (14a-d) hosted by the device (10), including: a step (23b) of obtaining information about currently active cellular network systems, which may include information about connections currently in use; and a step (24) of deciding whether to allow establishing the connection based on factors including the information about currently active cellular network systems. The information about connections currently in use may include identifiers for applications using the connections currently in use. The factors can also include the identity of the application making the request and the identities of the applications using connections currently in use, so as to make the decision as to whether or not to allow establishing the connection after taking into account the importance and type of the different applications contending for the network resources (connections).

Abstract: A system, a method and computer-readable media for initiating the hosting of email for an Internet domain name. A secure communication channel is established with an entity requesting email hosting services. After verifying that the requesting entity is authorized to control the domain name, information describing email accounts with the domain name is received, and email services are provided for the email accounts.

Abstract: A method for communication in a communication system includes predefining a plurality of service delivery modes having different signaling and media path configurations in circuit-switched and packet-switched networks. A request to place a call is accepted in the circuit-switched network, and a call service to be provided to the call by a service platform in the packet-switched network is identified responsively to the request. A service delivery mode is selected from the plurality responsively to an attribute of the call indicated by the request. Signaling and media paths are established for the call via the circuit-switched and packet-switched networks in accordance with the signaling and media path configurations of the selected service delivery mode. The call service is provided from the service platform to the call using the established signaling and media paths.

Abstract: Systems and methods are described for implementing an application-level routing protocol for multiparty audio-video conferencing. In one implementation, application-level per-stream routing techniques separately control audio data and video data between conference members hosted on a network. Different audio application-level multicast (ALM) trees are generated by each member, are dynamically updated according to shortest-path-first selection of data delivery paths, and are used to send audio data to the other members of the videoconference. Likewise, different video ALM trees are generated by each member, are dynamically updated according to broadest-path-first selection of data delivery paths, and are used to send video data to the other members of the videoconference. Separate audio and video ALM trees for each member can utilize IP multicast in segments of the network in which IP multicast is enabled.

Abstract: A communication system comprising a home server, an access gateway, and a policy server, wherein the policy server transmits, to the first access gateway, a link creation request for associating a communication path of the first terminal and a communication path of the second terminal; and the access gateway sets a fifth communication path which connects the first communication path and the third communication path upon receiving the link creation request, records information on the set fifth communication path in the communication path management information, and transmits the data from the first terminal via the first communication path to the second terminal via the fifth communication path and the third communication path based on the communication path management information.

Abstract: The present invention relates to a method in a communication network for connecting at least one application server (AS) (213) of an internet protocol multimedia subsystem (IMS) (200) to an intelligent network (IN) (100) through an interface unit (301). The IMS (200) comprises a call server control function (CSCF) unit (205) connected to the at least one AS (213) arranged for processing call signaling. In the method the interface unit (301) directs the call signaling to the at least one AS (213) through the CSCF unit (205) and the at least one AS (213) processes the call signaling and sends the processed call signaling through the CSCF unit (205) back to the interface unit (301). Based on the information received in the call signaling processed by the at least one AS (213), the interface unit (301) then directs the call to a destination.

Abstract: The present invention provides a technique for providing a forced hold service such as is used for an emergency services call, which is supported at least in part over a packet network. The forced hold service acts to effectively hold a connection for the call with a called party, even when the caller takes an action that would normally end a call, such as going on hook, pressing end, or the like. When the caller takes an action that would normally end the call, the forced hold service allows the caller to automatically reconnect to the emergency services provider over the held connection upon going offhook, pressing send, or the like. Alternatively, the emergency services provider can effectively re-engage the call wherein the caller is reconnected over the held connection upon going offhook, pressing send, or the like.

Abstract: A voice over internet protocol telephone system is described in which services for handling calls invoking account codes are distributed to voicemail servers in the system. If the voicemail server associated with a particular user does not have services for handling account code calls, then the request is forwarded to another voicemail server in accordance with a predefined procedure.

Abstract: An embodiment of the present invention is a method for server side integration of communication devices and the general purpose PC of the same user through a computer network wherein no physical connection is required between the PC and the communication device. The user registers with PnC (phone and computer) server for subscribing to one or more PnC services such as drop-to-call, conference-call-dropping service, webpage sharing, caller kaleidoscope etc., via user interface of communications device and/or PC. Various features for subscribing and unsubscribing to services are provided along with authenticating the user using the name and the phone number of the user while registering with the server.

Abstract: In one embodiment, an apparatus may verify an identity of a destination Voice-over-Internet-Protocol (VoIP) call agent for a destination telephone number based on demonstrated knowledge of at least one public switched telephone network (PSTN) call initiated to the destination telephone number. The apparatus may also receive the identity of the destination VoIP call agent based on the demonstrated knowledge of the at least one PSTN call initiated to the destination telephone number.

Abstract: In one embodiment, an apparatus may receive at least one call attribute of a public switched telephone network (PSTN) call initiated to a destination telephone number. The apparatus may verify a destination Voice-over-Internet-Protocol (VoIP) call agent for the destination telephone number based on demonstrated knowledge of the PSTN call. The apparatus may transmit an indication the destination VoIP call agent is verified for the destination telephone number.

Abstract: A system for verifying VoIP call routing information. The system may include an apparatus integrated with a private branch exchange (PBX). The apparatus may store at least one call attribute of a public switched telephone network (PSTN) call initiated to a destination telephone number. The apparatus may verify a destination Voice-over-Internet-Protocol (VoIP) call agent for the destination telephone number based on demonstrated knowledge of the PSTN call. The apparatus may route a new call either over a VoIP network to the destination VoIP call agent or over a circuit switched network based on whether the destination VoIP call agent is verified for the destination telephone number.

Abstract: A subsystem selection entity and method of controlling a subsystem selection entity is described. The subsystem selection entity is arranged to conduct a subsystem selection-procedure and a control procedure which comprises at least one control operation for controlling how one or more subsystems conduct a subscriber-not-reachable service.

Abstract: A system may provide trust relationship information for a telephone number, where the trust relationship information may indicate whether one or more nodes consider a target node as including or as being associated with a VoIP call agent for the telephone number based on demonstrated knowledge of a PSTN call. The system may determine a target trust relationship to the target node based on the trust relationship information, where the target trust relationship indicates a level of trust a validating node has in the target node being associated with or including the VoIP call agent for the telephone number, and where validating node and the one or more nodes may be in different respective administrative domains of a network.

Abstract: In one embodiment, an apparatus may receive a call over a Public Switched Telephone Network (PSTN) from a Voice over Internet Protocol (VoIP) adapter. The VoIP adapter may be one or more devices that may create and accept VoIP connections over a network, such as the Internet, and that may transmit a call over the PSTN. The apparatus may store a call detail of the received call in a registry service, where the call detail is associated with a node identifier of the apparatus in the registry service. The apparatus may further determine a dial sequence at which the apparatus may be reached over the PSTN based on corresponding call details also stored in the registry service.

Abstract: Techniques are described for providing managed virtual computer networks that have a configured logical network topology with virtual networking devices, such as by a network-accessible configurable network service, with corresponding networking functionality provided for communications between multiple computing nodes of the virtual computer network by emulating functionality that would be provided by the virtual networking devices if they were physically present.

Abstract: Presently described is a system and method for switching multimedia data communications, including but not limited to Voice over IP (VoIP) telephony, cable TV, digital audio and video. The system utilizes a single, integrated device to provide all PacketCable-compliant functionality, including enhanced user privacy, compliance with CALEA, E911 and other mandated services not available in conventional distributed PacketCable systems. High speed and efficient, low cost operation are provided by means of an optimized data unit encapsulation scheme for internal switching and routing. A proprietary fiber optic backplane and removable optical connectors are used to enable lightspeed internal communications hot-swapping of components. Furthermore, the present system is extensible to all forms of digital data switching and is secure, resistant to Denial of Service attacks, and fault-resilient.

Abstract: An apparatus and method for realizing the access of a legacy terminal to an IMS system. The apparatus includes a session control module, a downlink signaling interface function module, a downlink bearer interface function module, an uplink signaling interface function module, an uplink bearer interface function module and a media interworking module. The session control module registers the terminal that has entered service status to the I-CSCF on IMS side. During the session, the uplink signaling interface function module provides SIP signaling interaction with the CSCF function entity of IMS core network; the downlink signaling interface function module provides signaling interaction with the legacy terminal; the media interworking module provides the connection and media adaptation between the uplink bearer interface function module and the downlink bearer interface function module.

Abstract: In an internet telephony system, toll-free service and the calls that are accepted and paid for can be specified by the IP telephony system user to be limited according to the calling party's identity, time of day, day of week or other criteria. Data that is used to screen or filter incoming calls is programmed into a call manager by the IP telephone system user's broadband residential gateway. Using such a system the user can carefully screen calls that are paid for and more closely control telecommunications service costs. More generally, the user has the ability to store predefined call handling criteria, which may be executed by the broadband residential gateway and/or the call manager.

Abstract: A method of initiating a call includes receiving an identifier associated with a called party, the identifier being generated by a telephone handset. The method further includes determining a network category associated with the called party and generating an instruction to initiate a call by the telephone handset on one of at least two networks accessible for communication by the telephone handset. The one of at least two networks is selected by the instruction in response to the determined network category associated with the called party.

Abstract: A device is configured to detect a wireless device and determine if the wireless device is participating in a telephone call with a remote party. The device is further configured to establish a first voice communication link with the remote party, and establish a second voice communication link with the wireless device. In addition, the device is configured to bridge the first and second voice communication links together, thereby establishing voice communications between the wireless device and the remote party.

Abstract: Provided is a method for multiple-termination routing in a wireless network environment that includes an Internet Protocol (“IP”) core. The method includes receiving a call delivery request. The call delivery request is based upon intelligent network protocols and includes a called number. The call features associated with the called number are retrieved, a determination is made whether the call features include multiple-termination routing information for a plurality of potential terminating devices, wherein at least one termination to be routed utilizes Session Initiation Protocol. When the call features include the multiple-termination routing information, call setups are initiated to each of the plurality of potential terminating devices. Upon detecting a first potential terminating device to complete one of the plurality of initiated call setups, any remaining initiated call setup or setups are released.

Abstract: The present invention allows calls that are controlled by a multimedia subsystem (MS) to employ bearer paths that are supported in whole or in part by a circuit-switched subsystem (CS). As such, calls controlled by the MS can have a portion of the bearer path provided through the CS when needed or desired. To facilitate such control, a session control signaling path is established between a user element currently supported by the CS and a remote user agent (RUA), which represents the user element in the MS. While a portion of the bearer path for the call is supported by the CS, the session control signaling path extends the reach of the MS to the user element.

Abstract: A media portal is provided that is used to connect multimedia devices to other multimedia devices and content servers. When a call directed to a multimedia device busy in a call with the media portal is received, the media portal can provide a call waiting indication to the busy multimedia device without breaking the connection that is making the device busy. For example, if a first multimedia device is connected through the media portal to a content server providing streaming content of a football match, when a second device attempts to place a call to the first device, the media portal can provide a call waiting alert to the first device as part of, or as a substitute for, the delivery of the football match. Additionally the media portal can provide for the eventual connection of both devices while the previous call is still active.

Abstract: A system for verifying caller ID information in received VoIP calls. In particular implementations, a method includes receiving a caller identification (ID) identifying a calling party telephone number in a call initiation message transmitted from a VoIP call agent; determining the identity of the VoIP call agent; verifying that a public switched telephone network (PSTN) call to the calling party telephone number would arrive at a VoIP call agent having the determined identity; and applying, responsive to the call initiation message, one or more rules based at least in part on the verifying step.

Abstract: An apparatus is provided for remotely managing various network elements of a Voice over Internet Protocol (VoIP) communication system. In general, an apparatus is described for communicating with different network elements of a VoIP communication system, receiving data from each network element, and storing the data. The apparatus includes a processor that communicates with different network elements of a VoIP communication system over a packet-switching network and queries the network elements for particular information. The processor receives and stores the data regarding each network element. In this way, data from multiple different network elements of a VoIP communication system may be remotely accessed and stored, and the data may be analyzed or manipulated to address system problems or monitor conditions of the system from the remote location. A method and computer program product are also provided for remotely managing various network elements of a VoIP communication system.

Abstract: Information on a communication relation or communication path to be monitored is automatically generated to reduce load of a user. A path information generation part receives a destination IP address designated by a user, through an input receiving part. When the input of the destination IP address is received, the path information generation part uses configuration information tables, each of which stores configuration information of a device on a network, to identify IP addresses of networks each having a plurality of terminals, as branch IP addresses. The path information generation part uses the configuration information tables and transfer destination information tables each storing a routing table of a router, to identify the connection order of routers between the designated destination IP address and the identified branch IP addresses, to generate path information.

Abstract: A system for verifying VoIP call routing information. In particular implementations, a method includes verifying one or more Voice-over-Internet-Protocol (VoIP) call agents for respective destination telephone numbers based on demonstrated knowledge of previous public switched telephone network (PSTN) calls to the respective destination telephone numbers; receiving a call initiation message identifying a destination telephone number; and conditionally initiating a call over a VoIP network to a target VoIP call agent, or over a circuit switched network, based on whether the target VoIP call agent has been verified for the destination telephone number identified in the call initiation message.

Abstract: In one embodiment, a network device attaches a header including an identifier for an original call to an outgoing refer message and may also modify a table entry for a consultative call to include the identifier. The network device then monitors incoming invite messages for a header containing the identifier and observes that an incoming invite message containing the header correlates to the original call. When a header is not included, the network device extracts information from the incoming invite messages and compares the extracted information to the modified table entry to correlate one of the incoming invite messages to the original call.

Abstract: A method and apparatus for reserving an appointment in a communications network is described. In one embodiment, a request is received from a caller to schedule an appointment with an enterprise customer, wherein the request is processed in accordance with a media server in the communications network. A scheduling calendar is then accessed. Afterwards, the appointment is reserved with one of a plurality available appointment time slots.

Abstract: Authentication of a request for access to user specific information associated with a VoIP device is performed transparently, without any user identification or password required. A unique device identifier associated with the VoIP device is included in the request for access and used as an authentication token.

Abstract: A cellular communication system comprises a media gateway (111) which interfaces between a packet switched network (113) and a circuit switched network (115). The media gateway (111) time aligns received packet data in response to timestamps of these. Each of first and second RNC (105, 109) comprise a Node B interface (201) which receives real time data being data of an air interface communication from the remote station (101) to a base station (103, 107) supported by the RNC (105, 109). A packet processor (203, 205) then transmits the data packets to the media gateway (111) via the packet switched network (113). A timestamp processor (207) determines timestamps for the data packets in response to a timing characteristic of the air interface communication.

Abstract: The system contains an RF demodulator. A packet aggregation switching device is in communication with the RF demodulator. At least one packet processing engine is in communication with the RF demodulator. The packet aggregation switching device controls communication between the RF demodulator and the packet processing engine. An RF modulator may also be in communication with the packet processing engine along an egress path. The packet aggregation switching device may output traffic into the egress path.

Abstract: A system and method is described for connecting a logic circuit simulation to a hardware peripheral that includes a computer running software for communicating data to and from the hardware peripheral. The software transmits the data received from the hardware peripheral to the device being simulated by the logic circuit simulation. The computer also transmits the data received from the device being simulated by the electronic circuit simulation to the hardware peripheral. This allows the user to test the device being simulated using real hardware for input and output instead of simulated hardware.

Abstract: A method, system, and medium are provided for processing customer orders associated with providing telephony services via a telecommunications network and with services provided via a cable network. In one embodiment, the method includes providing a service-order-entry (SOE) system associated with a set of legacy components, the SOE system being operable to receive a first type of customer order that is associated with providing services via a telecommunications network, the telecommunications network not including a cable television (CATV) network; supplementing the SOE system with a secondary order-reception system operable to receive a second type of customer order that is associated with providing telephony services over a CATV network and further operable to receive the second type of customer order without substantially modifying the legacy components; receiving the second type of customer order; and fulfilling the second type of customer order without substantially modifying the legacy components.

Abstract: An apparatus for switching a user between an IP network and a PSTN is disclosed in the present invention and includes a user port, a Subscriber Line Interface Circuit, a detection circuit, a PSTN port, an IP network interface circuit and an action unit. A method for switching a user between the IP network and the PSTN is also disclosed. According to the method, the action unit is controlled by the control signal from a CPU, thereby implementing the user switching between the IP network and the PSTN on the apparatus. The apparatus and the method enable one or more users to switch between the IP network and the PSTN, implements flexible user selection between the two networks and offers user communication backup when failure occurs. Accordingly, the present invention increases the satisfaction of users and the core competence of enterprises.

Abstract: A network device includes a data store with first and second parameters, the first parameter affecting a communication operation of the network device and the second parameter affecting a security operation of the network device. A device reset function enables a user to selectively and independently refresh the first and second parameters. For example, a Session Initiation Protocol (SIP) device user can restore a default communications configuration of the SIP device while preserving the device decryption key.

Abstract: A method for facilitating VoIP communication between VoIP providers. First and second VoIP service providers are registered with a VoIP communication system, which generates a table of registered VoIP subscribers from the first and second VoIP service providers. The information from the table is applied to a call generated from a subscriber on the first VoIP service provider, such that if the desired party is a subscriber to the second VoIP service provider as noted in the table, the call generated from the subscriber on the first VoIP service provider is sent to the desired party as a packet switched call.

Abstract: A cryptographic processor for redundantly-processing cryptographic operations is disclosed. The cryptographic processor includes a number of input ports, a first and second cryptographic engines, comparison logic and a plurality of output ports. The number of input ports is configured to accept both plaintext and ciphertext. Each of the number of input ports is coupled to both the first and second cryptographic engines. The comparison logic is configured to determine if the first and second cryptographic engines produce a result that is different. The number of output ports is configured to produce both plaintext and ciphertext.

Abstract: Systems and methods are disclosed for system for processing calls to a SIP or VoIP device using the called party's Email address in which a call redirect server component is provisioned with the called party's Email address and VoIP or SIP device address, and a DNS system is updated with the called party's Email address and an address of the call redirect server. VoIP calls or other SIP sessions directed to the called party's Email address are resolved by the DNS system to the address of the redirect server, which in turn redirects the calls to the called party's VoIP or SIP device. The subscriber called party may provision the call redirect server with updated Email address or VoIP or SIP device address information to allow the subscribed to change VoIP phone service providers while callers only need to remember the subscriber's Email address.

Abstract: A system for managing voice and data communications coupled to a wide area telecommunications network. Video data is coupled between the system and a video unit under control of a processor. Audio data is coupled between the system and a microphone unit under control of the processor. The audio data are coupled to a TDM bus in the system operating under control of the processor. The video and audio data are coupled to a digital transmission link of the wide area telecommunications network. A user participates in a video call with the video data, while concurrently the user participates in a telephone call with the audio data. Voice communications of the telephone call that stay in a circuit-switched form are provided via the TDM bus. The video call and the telephone call provide a conference call.

Abstract: An exemplary method involves creating a master session over a first connection through a server; and creating a virtual channel over the connection, the virtual channel operable to communicate a feature session. The method may involve establishing a direct connection that bypasses the server, and switching communication of the feature session to the direct connection. A system for network communication includes a plurality of transport bridges, each transport bridge corresponding to an active network device configuration, and a switching module operable to choose one of the transport bridges to form a connection between two computing devices based on the active network device configuration.

Abstract: A method and apparatus for supporting call control of circuit domain telephone and Voice over IP (VoIP) are provided by the present invention, and the apparatus includes a call flow control module, which is for receiving and storing off-hook events and judging if the user makes a circuit domain telephone call or VoIP call according to the number dialed by the user; if making a VoIP call, sending the off-hook event to a VOIP call control module, and sending all the events in the present call to the VoIP call control module for processing; if making a circuit domain telephone call, sending the off-hook event to a circuit domain telephone call control module, and sending all the events in the present call to the circuit domain telephone call control module for processing.

Abstract: Subject: Since today's conventional Web browsing employs Pull-Type Telecommunications, presently, it is necessary to employ keyboard operation in order to request information. Furthermore, the portability of today's devices is still substandard, and offering information based on another's situation is presently unavailable. Method for Solution: When each telephone caller uses a telephone (including a cellular telephone), the computer connects the public Internet network and the Telecommunications with the exclusive software carried at each device, and the server performs Push-Type Telecommunications on the network, with which any information that is selected and transmitted by one side at any time is displayed or played automatically on the other side's monitor during conversation, the method by which audiovisual information is offered during the course of conversation is realized.

Abstract: An architecture is described for providing IP push-to-talk (IPP2T) service using a wireless local area network (WLAN) serving a plurality of subscriber terminals (STs), having at least one broadband access network terminal (BANT), the BANT coupled to and interacting with at least one of a plurality of Access Points (APs) via a local area network (LAN), the plurality of APs in communication with the plurality of subscriber terminals; a multicast-enabled network, the multicast-enabled network coupled to and interacting with at least one BANT via a broadband access network; an IP network coupled to, and interacting with, the multicast-enabled network via an edge router; and a WLAN mobile radio service (WLMRS) controller (WLMRSC) coupled to and interacting with the IP network via a multicast-enabled router (MR).

Abstract: In one embodiment, interworking procedures between H.323 and SIP ensure that in-band tones and announcements, along with local and remote ringback, are properly played or generated.

Abstract: Systems include configuring a first mobile device to enable communication between the first mobile device and one or more processing devices; receiving a message from a computing device that is directed to a second mobile device, where the message is a text message or a voice message, and where the computing device is different from the first mobile device; and routing the message to the first mobile device for transmission to the second mobile device from the first mobile device.

Abstract: A signal monitoring apparatus and method involving devices for monitoring signals representing communications traffic, devices for identifying at least one predetermined parameter by analyzing the context of the at least one monitoring signal, a device for recording the occurrence of the identified parameter, a device for identifying the traffic stream associated with the identified parameter, a device for analyzing the recorded data relating to the occurrence, and a device, responsive to the analysis of the recorded data, for controlling the handling of communications traffic within the apparatus.