Abstract: An indirect communication system and a method of indirect communication include a mobile phone as either a calling device or receiving device. The mobile phone calls another device to set up a prospective communication, and then uses Voice over Internet Protocol (VoIP) to communicate with the other device over the Internet. The receiving device receives a generated signal notifying the receiving device of a proposed communication with the calling device. A server sets up a meeting point channel after the calling device has connected to the server. The server receives outgoing VoIP packets from the calling device and redirects the outgoing VoIP packets to the receiving device.

Abstract: An integrated circuit chip implements a high-speed switch that includes: a switch fabric; control logic that controls the transmission of digital signals through the switch fabric; a transceiver block comprising one or more transceivers, each transmitting digital signals between the control logic and a corresponding external device; a data converter physical interface comprising one or more data converters, each performing a conversion between analog and digital signals, wherein digital signals associated with the one or more data converters are routed through the switch fabric; and a signal processing engine coupled to the control logic, wherein the signal processing engine performs on-chip processing of digital signals received from the transceiver block and the data converter physical interface.

Abstract: A high-speed switch that includes a switch fabric, and both high-speed serial ports and data converter physical ports. A first set of data converter physical ports may perform analog-to-digital conversions, such that an external analog signal may be converted to a digital input signal on the switch. The converted digital input signal may then be routed through the switch fabric in accordance with a serial data protocol. A second set of data converter physical ports may perform digital-to-analog conversions, such that an internal digital signal received from the switch fabric may be converted to an analog output signal on the switch. The converted analog output signal may then be transmitted to an external destination in accordance with a serial data protocol.

Abstract: An online home improvement document management service includes a host server having a web-based interface adapted to facilitate secure customer access to the host server. The host server is configured to receive home improvement data communicated from the customer through use of a computing device. A processing module is in operative communication with the host server and is configured to process the home improvement data communicated from the customer. A processing software application is trained to classify the processed home improvement data and selectively extract data therefrom based on the classification. The processing software application is configured to selectively present the processed home improvement data for a customer verification via the web-based interface upon at least one of an unsuccessful classification and an unsuccessful extraction of data.

Abstract: A method for data security policy enforcement including inspecting incoming and outgoing data packets from a server computing device for attributes in accordance with a data security policy, processing the data packets in accordance with the security policy based on the inspected attributes, and routing the data packets in accordance with the security policy based on the inspected attributes, wherein incoming and outgoing data from the server computing device composed of the data packets is processed and routed in accordance with the security policy on a per-packet basis. A system and computer program product is also provided.

Abstract: A method of setting up a video telephone call between a first video telephone terminal and a second video telephone terminal on a data network includes the steps of: establishing a telephone call over a telephone network between the first and second terminals; exchanging between the first and second terminals during the telephone call availability signals to seek availability to hold a video telephone call; if the availability is confirmed, releasing the telephone call; establishing respective telephone calls between each of the first and second terminals and a corresponding access server of the data network, for achieving connection of the first and second terminals to the data network; establishing a signalling exchange between the first and second terminals through a service center of the data network to achieve the set up of the video telephone call; and establishing a peer-to-peer video telephone call between the first and second terminals over the data network.

Abstract: A method and device are provided for conducting a real-time, voice over IP media type of communication session between at least two peer users, each associated with a platform comprised in a distributed communication network. The method and device comprise providing at least one replication of at least one data object to all other platforms associated with that distributed communication network.

Abstract: Systems and methods are provided that allow the delivery of location based services within a communication network. The location information can be retrieved using information from the mobile node when the mobile node registers in the network. The location information can then be cached or stored in one or more places in the communication network and correlated with the mobile node's addressing information. If a request for location based services is received without location based information, the gateway can use location based information regarding the mobile node to provide location based services. The gateway can enable non IMS mobile nodes to obtain IMS location based services or incompatible mobile nodes to obtain location based services.

Abstract: Methods and systems of authorizing a user of a first packet-based communication network to access a second packet-based communication network are disclosed. A call setup request is received from a user terminal of the user at a first network element of the first packet-based communication network, the authorization request comprising a first user identity. Responsive to the call setup request, a request is transmitted to create a second user identity from the first network element to a second network element of the second packet-based communication network. The second network element creates the second user identity for use in the second packet-based communication network. The second user identity is derivable from the first user identity according to a predetermined rule. The second user identity is stored in the second packet-based communication network for use with establishing a call associated with the call setup request over the second packet-based communication network.

Abstract: A system and a method for integrating the receipt and transmission of location information into VoIP phone systems. The system may include a location device that includes a SATPS receiver and an antenna together with one or more connectors that allow connection of the location device to other devices, such as a VoIP telephone and a router, that are connected to and made a part of a VoIP phone system. A method of operating such a system is also disclosed.

Abstract: One embodiment of the present invention provides a collect call method and system thereof, more particularly, in order to charge the called party with a uniform toll for collect call, which is determined by only the type and location of called party terminal. In one embodiment, the collect call method, system and a counsel service providing method use a free VoIP network for part of the voice call link and a charge PSTN network for the rest of the voice call link. In one embodiment, if the first link corresponding to the collect call request is established, the collect call switch calls the called party terminal to establish the second link, and billing on the second link is initiated.

Abstract: According to one aspect, the subject matter described herein includes a method for routing Diameter messages. The method includes steps occurring at a Diameter signaling router (DSR) comprising a plurality of Diameter message processors. The method also includes receiving, by a first of the plurality of Diameter message processors and from a first Diameter node, a Diameter message, wherein the Diameter message is received via a first Diameter connection. The method further includes determining, by the first Diameter message processor, a next-hop Diameter node for the Diameter message. The method further includes communicating, by the first of the plurality of Diameter message processors and to a second of the plurality of Diameter message processors, the Diameter message. The method further includes communicating, by the second Diameter message processor and to the next-hop Diameter node, the Diameter message, wherein the Diameter message is communicated via a second Diameter connection.

Abstract: A system and method are provided for allowing a user to consent to call recording. A user is prompted to make a recording preference. The recording preference is stored in a table indexed by UCID (universal call ID) or as part of a UUI (user-to-user information) field of the call. A recording module and/or a compliance manager, both associated with a call center, are configured to record or not record the call based on the recording preference. The recording preference can be enforced on incoming and outgoing calls to and from the call center. Advantageously, the user's recording preference is enforced throughout the call, regardless of how the call is routed/transferred within the call center.

Abstract: Internet Protocol (IP) audio conferencing bridging provisioning/configuration systems receive a service order request associated with a customer, where the service order request requests enrollment of the customer in a Virtual Private Network (VPN) bridging network service for IP Audio Conferencing. The provisioning/configuration systems orchestrate, based on the service order request, configuration of multiple network nodes in a network that provides the VPN bridging network service for IP Audio Conferencing to the customer to enable IP audio conferencing calls to be made from the customer via the network.

Abstract: A method and system of reachability indication between a wireless device and at least one push server, the method comprising the steps of: sending device status information from the wireless device to the at least one push server; and receiving the status information at the at least one push server; wherein the at least one push server is enabled to selectively start and stop serving the wireless device on the basis of the status information. The method further comprises sending status information to a packet data serving node that stores a list of push servers associated with a wireless device, and having the packet data serving node forward the status information to the push server. The system and method further includes selectively starting and stopping the serving of the wireless device by the push server during a voice call.

Abstract: There is discussed a method of selecting an access domain for receiving a speech and/or video call at a mobile station of a mobile communications network in which calls are routed via a central service control common to a plurality of access domains. The method comprises the central service control transmitting an invite to establish a call to the terminating mobile station; the mobile station deciding on an access domain for receiving the call taking into account the capabilities of the local access network, and transmitting the selected access domain to the central service control; and establishing the call in accordance with the selected access domain.

Abstract: Techniques for performing handover in order to maintain call continuity for a user equipment (UE) are described. The UE may communicate with a first cell in a radio access network (RAN) for a packet-switched (PS) call, e.g., for Voice-over-Internet Protocol (VoIP) via High-Speed Packet Access (HSPA) in W-CDMA. The UE may send measurement reports to the RAN and may receive trigger from the RAN. The UE may establish a circuit-switched (CS) call with the first cell while the PS call is pending at the first cell. The PS call and the CS call may be for a voice call, and the UE may switch data path for the voice call from the PS call to the CS call and then terminate the PS call. The UE may then perform handover of the CS call from the first cell to a second cell, which may not support VoIP.

Abstract: 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.

Abstract: In one configuration, the present invention is directed to an enterprise network that includes geographically dislocated first and second network regions communicating with one another through first and second networks and respectively comprising first and second gateways and first and second groupings of trunks. A media server is positioned in the first network region that includes a routing agent. When a WAN outage occurs, the routing agent, using a dial plan, maps the dialed digits to an electronic address addressable over the PSTN and routes the call over the PSTN to the destination second communication device in the second network region.

Abstract: To change between a packet-oriented PTT session and a circuit-oriented telephone connection as selectable communication connections between at least one first connected radio communication appliance and at least one second connected radio communication appliance in a radio communication system, a first of these two communication connections is selected and set up by at least one connected radio communication appliance and/or by at least one control unit in the radio network in the radio communication system. A change is made from this first communication connection, which has been set up, to the second communication connection under the control of a radio communication appliance which is connected to the first communication connection and/or under the control of that control unit in the radio network in the radio communication system which is connected to the first communication connection by activating the second communication connection while the first communication connection still exists.

Abstract: A system, method, and computer readable medium comprising instructions for establishing a conference call comprising sending a first SIP INVITE message from a Media Gateway to an Application Server, sending a second SIP INVITE message to a first Media Server, establishing a session between the Media Gateway and the first Media Server, sending a third SIP INVITE message from the Application Server to a Conference Controller, allocating resources related to the conference call by the Conference Controller on a second Media Server, requesting by the Conference Controller for the Application Server to move data related to establishing the conference call from the first Media Server to the second Media Server, sending a fourth SIP INVITE message by the Application Server to the second Media Server and sending a SIP RE-INVITE message by the Application Server to the Media Gateway, and establishing an RTP session between the Media Gateway and the second Media Server thereby establishing the conference call.

Abstract: An audio and video communication system, including a handset including a handset receiver and an audio output component, for receiving audio data via the handset receiver and for playing the received audio data on the handset audio output component, a display device, not physically connected to the handset, including a display receiver and a display screen, for receiving video data via the display receiver and for displaying the received video data on the display screen, and a telephony service, communicatively coupled with the handset and with the display, that receives audio and video data from a transmitter computer during a communication session, and transmits the audio data to the handset and the video data to the display device.

Abstract: A method and system for routing a call connection signal and corresponding contextual information to an appropriate destination is provided. A call center and at least one user may exchange signals, and/or contextual information during a call set-up phase. Relevant contextual information and other detailed information may be identified and collected from their corresponding sources. Based on the relevant contextual information and other detailed information, one or more appropriate routed destinations may be determined. In addition, contextual information may be tailored for each of the determined routed destinations. The call connection signal and the tailored contextual information are routed to each routed destination. In this manner, necessary contextual information may be readily available for the routed destination at the time of the call connection and thus the routed destination can provide a faster service.

Abstract: A technique for routing a call to a unified service domain using one or more client applications providing call routing support from a circuit-switch access domain to unified service domain is described. A first one of the client applications will be provided on a network side and a second one of the client applications may potentially be provided on a terminal side. A method implementation of this technique comprises, on the network side, the steps of receiving a message from the terminal side, detecting, in response to the message, the client application that is to provide the routing functionalities, and controlling an activation state of at least one of the two client applications dependent on a result of this detection.

Abstract: Apparatus, systems, and methods may operate to detect a device connecting to a network, determine the identity of the network, and publish at least a portion of a list of active contacts to a display, while hiding a list of inactive contacts. The list of active contacts may be selected from an original list of contacts, based on the Identity of the network and a personal rule set associated with a user of the device. The list of inactive contacts is included in the original list and not included in the list of active contacts. Additional apparatus, systems, and methods are disclosed.

Abstract: Techniques to provide integrated voice service management are described. An apparatus may comprise a mobile computing device having a voice service manager operative to receive control directives from one or more application programs, and manage one or more service features provided by multiple voice service platforms based on the received control directives, with at least one of the multiple voice service platforms comprising a voice over packet voice service platform. Other embodiments are described and claimed.

Abstract: Under one aspect, a system for transmitting an IP-based message from an initiator to a receiver lacking an IP address via a packet-switched (PS) network capable of communicating IP-based messages and a circuit-switched (CS) network capable of communicating non-IP-based messages, includes: a serving node in communication with the PS network and the CS network, the serving node including logic to: receive the IP-based message from the initiator over the PS network; generate a trigger message responsive to the IP-based message, the trigger message including a non-IP-based message including instructions for the receiver to initiate a connection to the PS network; and transmit the trigger message to the receiver via the CS network, and the receiver including logic to initiate a connection to the PS network and to receive the IP-based message responsive to the trigger message. Methods are also disclosed.

Abstract: A user of a first packet-based communication network is authorized to access a second packet-based communication network. In at least some embodiments, an authorization request is received from a user terminal of the user at a first network element of the first packet-based communication network, the authorization request comprising a first user identity. Responsive to the authorization request, a request is transmitted to create a second user identity from the first network element to a second network element of the second packet-based communication network. The second network element creates the second user identity for use in the second packet-based communication network, the second user identity being derivable from the first user identity according to a predetermined rule. The second user identity in the second packet-based communication network is stored for use with subsequent communication events over the second packet-based communication network.

Abstract: A technique is provided for use by a user terminal or mobile services switching center (MSC) for responding to a request to switch an on-going call from speech mode to multimedia mode within a wireless communication network conFig.d to automatically disconnect a call if a requested mode switch is not confirmed within a predetermined period of time specified by a call drop timer. In one example, the terminal or MSC activities an automatic response timer upon receipt of a request from a user to switch an on-going call from speech to multimedia, with the timer set shorter than the call drop timer. The user receiving the request is then prompted to accept or decline the request. If no response is received before expiration of the automatic response timer, the terminal or MSC automatically declines the request on behalf of the user so that the call drop timer does not expire.

Abstract: Messages transmitted between a receiver and a transmitter are used to maximize a communication data rate. In particular, a multicarrier modulation system uses messages that are sent from the receiver to the transmitter to exchange one or more sets of optimized communication parameters. The transmitter then stores these communication parameters and when transmitting to that particular receiver, the transmitter utilizes the stored parameters in an effort to maximize the data rate to that receiver. Likewise, when the receiver receives packets from that particular transmitter, the receiver can utilize the stored communication parameters for reception.

Abstract: Combinational networks may provide simultaneous connectivity over networks of different type between terminals. Communication sessions on different network types such as Circuit switched and Packet switched, belonging to the same user equipment can be correlated. In case a communication session on a circuit switched network is halted by a supplementary service e.g. at an event such as acceptance of Call Hold, a communication session on a correlated packet switched network should be halted as well. A user equipment that detects the event sends a halt message to the circuit switched network and a message to the packet switched network or a session state manager node. The session state manager node either forwards the halt-message to the packet switched network, or sends a halt-message to the packet switched network when the packet switched network does not notify that a halt has occurred.

Abstract: A method, device, and system for transferring a Service Control Signalling Path are provided. The method for transferring a Service Control Signalling Path includes: establishing a connection with an opposite end by a User Equipment (UE), where the UE uses a Circuit Switched (CS) bearer in a CS network and a Service Control Signalling Path in a first Packet Switched (PS) network; sending a transfer request via a second PS network, to instruct a network side to transfer the Service Control Signalling Path according to the transfer request. Thus, the UE can replace a current Gm reference point with a Gm reference point of a new and available PS network when the PS network where the current Gm reference point is located is unavailable, so as to ensure smooth data transmission.

Abstract: Methods and systems for dual digital microphone processing in an audio CODEC are disclosed and may include demultiplexing one or more received time-multiplexed digital audio signals from one or more digital microphones, and separately processing each of the demultiplexed digital audio signals. The digital microphones may include microelectromechanical (MEMS) microphones. The demultiplexed digital audio signals may be level-converted, downshifted, and/or filtered. The filtering may include a finite impulse response (FIR) filter. A sampling rate of the one or more demultiplexed digital audio signals may be converted by repeating the demultiplexed digital audio signals. Audio beamforming and/or diversity processing may be performed utilizing the digital microphones.

Abstract: A smooth and seamless terminal switching guaranteeing continuity of communication at an application level is realized. A call control part for setting and managing a physical and logical transmission path between communication terminals, a session management part for managing establishment of data communication between the communication terminals, and a switching request processing unit for controlling interruption and resumption of a session are provided, wherein the switching request processing unit exercises a function to temporarily interrupt a session after receiving a session switching request signal from a communication terminal, a function to instruct the call control part to establish a call connection with a switch destination communication terminal, a function to establish a session with an application on the switch destination communication terminal, and a function to resume, after establishing the session, a session with a communication terminal of an opposite side.

Abstract: An emergency call (e.g., a 911 call, an alarm company call) forwarded by a telematics call center is routed over the switched PSTN to a Voice Over Internet Protocol (VoIP) call server, where the switched call is converted to a session initiating protocol (SIP) packetized IP call for presentation to an emergency services gateway, which reconverts the call to TDM and connects to a selective router via dedicated circuits, gaining full access to the Enhanced 911 network. This provides a PSAP receiving a call from a telematics call center or other call center with all features available in an Enhanced 911 network, e.g., callback number of the 911 caller, and location of the 911 caller. Location of the caller is provided using a VoIP positioning center (VPC).

Abstract: A report message may be transmitted with collaborated information related to the occurrence of an emergency event. A plurality of devices, such as mobile communications devices, that are out of range of a communications network can communicate via direct communication, such as device-to-device communication, to corroborate characteristics that are indicative of an occurrence of an emergency event. Information related to the occurrence may be collaborated amongst the plurality of devices. A transmitting device, which may be one of the plurality of devices that corroborates characteristics or collaborates information, may generate and transmit the report message comprising the collaborated information. A receiving device may provide the message over a communications network, or the report message may continue to be handed off between devices capable of direct communication, the message eventually reaching a device that is within range of the communications network.

Abstract: A terminal, method and system for providing a CS service, a SIP-based service, or a CSI service, are provided. According to an embodiment, the terminal includes a controller to receive a terminal capability version of at least one target terminal, to compare the received terminal capability version with a previously stored terminal capability version of the at least one target terminal, and to determine whether to request for terminal capability information of the at least one target terminal based on the comparison result, wherein the terminal capability version identifies a version of capabilities of the at least one target terminal.

Abstract: A method for realizing non-loss transmission of media stream after switching between the open call and the secret call, the method includes the steps: after the base station receiving the message of switching from the open call to the secret call from the communication terminal, inserting a special media frame in the media stream and transmitting it to the media gateway; the media gateway judging whether checking the special media frame according to the form of the load-bearing of the media stream's incoming and outgoing or the form of the media's coding decoding, whether detect the special media frame or not, the media gateway transmitting the media stream from the incoming to the outgoing. When the special media frame is detected, deciding to transmit directly or run an PCM adaptation process in 64 kbps/56 kbps according to the form of the coding decoding.

Abstract: Multimedia IMS communications are established by matching callee capabilities with caller preferences and identifying a URI parameter in response thereto. When a session initiation protocol (SIP) request identifying caller preferences for an internet communication session and an identified callee is received (602), a determination is made of whether the identified callee is a registered address of record (AOR) or not (604). If the identified callee is a registered AOR (604), a uniform resource identifier (URI) parameter and a globally routable user agent URI (GRUU) associated with callee capabilities are identified (608, 612, 614). The URI parameter is used to create a modified SIP including the URI parameter (616) for sending (618) to an IMS enabled wireless communication device (120).

Abstract: A modem to process calling packets includes receiving a calling request packet from a software phone of a communication terminal, and determining if the calling request packet includes a special tag. If the IP phone is idle, the modem records a source IP address of the calling request packet, and modifies the source IP of the calling request packet to be an IP address of the IP phone, then the modem transmits the modified calling request packet to a server, and receives a calling reply packet from the server, then modifies a destination IP address of the calling reply packet to be the IP address of the communication terminal. The modem transmits the modified calling reply packet to the software phone to establish the call.

Abstract: A disclosed method for providing service provider information to a user communication device includes determining a telecommunications service provider associated with a targeted node such as a called party telephone number. An indicator of the telecommunications service provider is provided to and displayed by the user communication device. Communication services available to the communication device are then determined based at least in part on the identity of the telecommunications service provider. Indications of the available communication services may be presented to the communication device by audible or visual signals.

Abstract: A method and system for establishing a conference call is presented. The method uses the Session Initiation Protocol for transporting conference information to a conference server, wherein the conference is represented in a markup language.

Abstract: A method of call control in which a first communications network, detecting calls directed to a destination in the first network; redirects at least some of the calls to a destination in a second network (e.g. on no answer from the destination in the first network). The first network then operates in two phases. In a first phase, for a group of calls redirected to the second network, the time delay before the call is answered in the second network is recorded. When a number of time delays have been recorded, a delay period is set to a value less than the maximum value of the recorded time delays. In a second phase, for a second group of calls redirected to the second network subsequent to the calls of the first group, when a call of the second group is not answered in the second network within the delay period, the call is redirected to a message service associated with the first network.

Abstract: Disclosed is a mobile communications system including a plurality of base station apparatuses which are located at geographically dispersed locations and perform a wireless communication with a mobile device. The system further includes a maximum ratio combining unit which performs maximum ratio combining processing regarding a signal the base station apparatuses receive from the mobile device, and a selection combining unit which performs selection combining processing regarding the signal from the mobile device. A selection unit included in the system selects one or both of the maximum ratio combining unit and the selection combining unit according to at least one of an external factor and a change of a distribution state of the mobile device under communication.

Abstract: Methods and apparatus for originating a Session Initiation Protocol (SIP) call from a user equipment (UE) device in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network to a called party are disclosed. In one illustrative example, when the SIP call is originated by the UE device in the CS network domain, a SIP Invite message which includes a SIP Uniform Resource Indicator (URI) or Tel URI of the called party is sent from the UE device to the IMS network (e.g. to an application server (AS) node). At the AS node, a pool of E.164 numbers are maintained as IP multimedia routing numbers (IMRNs) which are utilized for mapping to or otherwise associating with called party URIs. Thus, the AS node dynamically allocates a select E.164 number with respect to the called party's URI received from the UE device, and returns it to the UE device in a SIP Response message, e.g., a SIP 380 (Alternative Service) message. Subsequently, the dynamically-allocated E.

Abstract: A system and method for processing a plurality of requests for multi-media services received at a call control element (CCE) defined on the system from a plurality of IP-communication devices. The system includes at least one Network Routing Element (NRE), a Service Broker (SB), a media sever, a plurality of application servers (ASs) and a plurality of border elements, all of which are coupled to the CCE. The CCE is adapted to receive requests for multi-media services and to generate subsequent requests for the multi-media services, which are communicated to the SB for processing. The subsequent requests can each include a Session Initiation Protocol (SIP) message including a message identifier portion having at least a first predetermined information field and a second predetermined information field. The message identifier portion of the SIP message declares the content and format of the SIP message to a recipient device defined on the system.

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: System and method for IP telephony. The system includes an IP telephone (IPT) and a Service Gateway (SG). The SG receives an identifier, e.g., a vendor class identifier, included in a DHCP discover message from the IP telephone and determines if the identifier is valid. If so, the SG issues a DHCP offer comprising DHCP lease information to the IP telephone, including a range of port numbers assigned to the IP telephone based on the identifier, where the range of port numbers comprises ports which are not reserved for use by other IP protocols. The DHCP lease information includes information indicating operational software for the IP telephone which the IP telephone executes to enable IP communications. The SG mediates IP communications between the IP telephone and an IP device, where the IP telephone uses at least a subset of the range of port numbers to send or receive IP communications.

Abstract: A system and method for sending long distance telephone calls over the Internet utilizes cost and quality of service data to optimize system performance and to minimize the cost of completing the calls. The system utilizes a network of gateways connected to the Internet. The gateways receive calls from various service providers and convert the analog calls into data packets which are then placed onto the Internet. Similarly, the gateways take data packets off the Internet, convert the data packets back into analog format, and provide the analog telephone calls to the same or another service provider. Then system periodically checks the quality of communications between each of the gateways, and uses this information, in combination with cost information, to determine how to route the calls over the Internet. Special addressing protocols can be used by a system embodying the invention to reduce or eliminate unnecessary signaling between gateways as call setup procedures are carried out.

Abstract: In one embodiment, a scheme is disclosed for managing call routing in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network. When a call is originated by a user equipment (UE) device in the CS network, call information associated with the call is provided to a call continuity control function (CCCF) network node disposed in the IMS network. At the CCCF node, a pool of E.164 numbers are maintained as IP multimedia routing numbers (IMRNs) which are mapped to or otherwise associated with called party numbers. The CCCF node dynamically allocates a select IMRN with respect to a called party number received from the UE device and returns it to the UE device. The dynamically allocated IMRN is then utilized for routing the call towards the called party.