Abstract: Method and apparatus for providing integrated network and operations control for a converged communications network providing all service types on a common architecture.

Abstract: A method and apparatus for controlling an optimization of handover procedures between universal terrestrial radio access (UTRA) release 6 (R6) cells and UTRA release 7 (R7) cells are disclosed. When a wireless transmit/receive unit (WTRU) is moving between an R6 cell and an R7 cell, or between R7 cells, a handover is initiated from a source Node-B to a target Node-B. In the R7 cell, the enhanced medium access control (MAC) functionality including flexible radio link control (RLC) protocol data unit (PDU) size and high speed MAC (MAC-hs) segmentation and multiplexing of different priority queues are supported. After the handover, a MAC layer and/or an RLC layer are reconfigured or reset based on functionality supported by the target Node-B.

Abstract: In a communication system in which computer 20 transmits an HTTP request to WWW server 70 via broadband router apparatus 30 and the Internet 50, and WWW server 70 transmits an HTTP response to computer 20 according to the received HTTP request, broadband router apparatus 30 establishes an IP phone session and a TCP connection with HTTP proxy server 620 upon receiving the HTTP request from computer 20, and transmits the received HTTP request to WWW server 70 via HTTP proxy server 620.

Abstract: Selected modalities of a multimodal communication session may be diverted over different networks by an administrator or automatically to reduce burden on utilized networks. Modalities may be selected and diverted over different networks including the Internet for the same multimodal communication session based on network conditions, communication session attributes, participant attributes, and similar factors.

Abstract: Each of a number of gateway network radios operates with a transmission protocol of an associated one of plural incompatible communication networks. Each network radio has an audio terminal for producing first audio signals detected from transmissions of user radios in its associated network, and a microphone terminal for receiving second audio signals for transmission to the user radios. A voice bridging gateway server has multiple ports each corresponding to one of the networks for coupling to the network radio associated with the one network. Each port receives the first audio signals from the audio terminal of the associated network radio, and produces the second audio signals for applying to the microphone terminal of the network radio. The server implements a private branch exchange application whereby user radios in a given communication network can be voice bridged with user radios in a selected one or more of the other networks.

Abstract: A system and method for remote monitoring through a voice over Internet Protocol phone. An indicator configured to activate a video interface of the VoIP phone is received. A determination is made whether the indicator authorizes the video interface. The video interface of the VoIP phone is activated in response to determining the indicator authorizes the video interface. Video communications are sent and received between the VoIP phone utilized by a receiving party and a communications device utilized by a calling party. An action is performed in response to recognizing a response from the receiving party.

Abstract: Call spoofing in a Voice over Internet Protocol (IP) system may be detected and/or prevented. A device may receive a call request to initiate a VoIP call. The device may analyze marker information in the call request to determine whether the call request is valid. The device may also analyze the call request, based on call pattern information of a quantity of other calls, to determine whether the call request is suspicious; and complete the call based on the determination of whether the call request is valid and the determination of whether the call request is suspicious.

Abstract: A mobile communication system according to the present invention includes the steps of: causing an UE to switch a radio access network in a connection destination from E-UTRAN to UTRAN/GERAN; causing an MME to receive “Delete Bearer Request” to a voice communication packet bearer from an S-GW; and causing the MME to transmit to the S-GW a “Delete Bearer Response” to the “Delete Bearer Request”, without giving an instruction to the UE#1 or the E-UTRAN to release the voice communication packet bearer, if a core bearer disconnection flag is set in the received “Delete Bearer Request”.

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: The present invention relates to a solution for handling handover of connections for a user equipment from a packet switch network to a circuit switched network. This is provided as a method implemented in a node and system. The method is provided in a telecommunications network (100) connecting user equipment, i.e. UE, (101) communicating wirelessly (111) with the network. The method comprising steps of detecting (301) that handover is required from a packet switched, i.e. PS, based network (102) to a circuit switched, i.e. CS, based network (103), initiating (302) in a source mobility management node (104) a single radio voice call continuity, i.e. SVRCC, initiating (303) a PS to PS handover for non-voice components with information about voice related parameters and a PS to CS handover indicator, executing (304) hand over, sending (305) an update to a serving gateway, i.e.

Abstract: A system for managing voice and data communications for at least one user is coupled to at least one wide area telecommunications network and includes at least a processor and data storage resources. A first bus is adapted to couple video data between the system and a video unit coupled to the system under control of the processor. A second bus is adapted to couple audio data between the system and a microphone unit coupled to the system under control of the processor, and audio data are coupled to a TDM bus in the system operating under control of the processor. A digital transmission link of the wide area telecommunications network couples the video and audio data from the system, under control of the processor. A user participates in a video call with the video data, while concurrently the user participates in a telephone call with the audio data.

Abstract: A method and apparatus for enabling a tool to collect alerting information of network disruptions across access networks external to a packet network, e.g., a VoIP network, is disclosed. By monitoring the status of external access networks, service disruptions in the VoIP network that can be potentially caused by such access networks can be anticipated, prevented, diagnosed and/or resolved rapidly.

Abstract: A system and method for monitoring call quality for calling centers using packet based call technology. A distributed system manages packet flow between a caller and a call center agent and storage servers. The distributed system is used to monitor, record and analyze real time communications between the caller and the agent and to identify whether certain predetermined parameters are occurring in any particular call. In the event that such a predetermined parameter does exist, a message can be sent to a supervisory station or dialog guidance messages may be sent to the agent.

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: In a system and method for establishing communications in a communications network, a network service provider can assign IP addresses to mobile devices dynamically to conserve IP address resources. A network service provider can also implement network address translation to further conserve IP address resources and to provide improved security. If a requestor seeks to obtain an IP address of a mobile device and the address is a local address, the system determines if the mobile device has a network address translation (NAT) binding that associates the IP address of the mobile device with a public IP address. If the mobile device does not have a NAT binding, the system creates a NAT binding that associates the IP address of the mobile device with a public IP address.

Abstract: A method and apparatus for handling internet access telephone calls made via cable company telephone services. A head end data terminal receives cable signals and converts them into individual signals. An intelligent switch detects signals destined for an internet service provider and routes those signals on a separate path to the internet service provider. A central switch routes the other signals along a telephone network. A computer program can control the steps of receiving cable signals, converting them into voice band signals, routing the signals that are not for the intended recipient to a central switch, multiplexing the signals for the intended recipient together, and sending the multiplexed signals to the intended recipient.

Abstract: A gateway apparatus 110 includes: a communication unit 111 communicating a frame signal conforming to an IuUP (Iu User Plane) protocol with a wireless base station controller connected to a circuit switching network 101; a determination unit 112 for determining whether the frame signal received by the communication unit indicates a time alignment request for a sound signal to a circuit switching network or not; a conversion unit 113, converting information included in the frame signal to information to be set in a call control signal of an IMS (IP Multimedia Subsystem) when the frame signal indicates a time alignment request; and a call control signal transmission unit 114 transmitting the call control signal including the converted information to a communication apparatus in the IMS 102.

Abstract: In a distributed antenna system (DAS) and a local area network (LAN), a common communication infrastructure distributes data from radio-based and Internet-based sources. A radio equipment (RE) of the DAS interfaces to a LAN segment. For the downlink, a gateway maps radio signal data from a radio equipment controller (REC) and data packets from a switch to mixed-data frames using a radio data interface protocol for transmission in the DAS. At the RE, the signal data and data packets are retrieved from the mixed-data frames and provided to the air interface and LAN segment, respectively. For the uplink from the RE, the radio signal data from the air interface and the data packets from the LAN segment are mapped to mixed-data frames and transmitted to the gateway. The gateway retrieves the signal samples and data packets from the mixed-data frames for transfer to the REC and switch, respectively.

Abstract: In a communications system, a first packet network is provided. Packetized data is transferred between the system and one or more packet-based devices. A TDM network is provided, and data is transmitted in frames having slots. Data transmitted via the TDM network includes data for voice communications for telephony devices. The TDM network is selectively coupled to the first packet network and a WAN. A processor and a control bus interface circuit control transfer of packetized data and transmittal of data for voice communications. The processor controls processing of packetized data and data for voice communications. A switch/multiplexer selectively controls providing data to/from particular slots. The processor selectively controls voice communications from telephony devices over the TDM network and packet-based communications over the packet network. Voice communications that stay in a circuit-switched form occur over the TDM network and the WAN.

Abstract: A temporary ENUM architecture assigns a temporary ENUM at call time. For an originated communication requiring routing to a party which needs a particular callback format not supported, the TempENUM provides consistency with the specific format required by the receiving party. The TempENUM is assigned for the duration of the call or communication, or for a given amount of time thereafter. The TempENUM service gateway includes a TempENUM database, a module that assigns a suitable TempENUM at the time of call routing, and a module that handles a lookup in the TempENUM database, and maps the assigned TempENUM to a calling party identifier (CgPID). The TempENUM database stores a TempENUM from a coordinator system and relates the actual calling party identifier (CgPID) for use in callback attempts. Either during the initial call or communication, or shortly thereafter, the receiving party uses the TempENUM number to return communication the initial caller.

Abstract: A femtocell architecture and network are described. In one example the femtocell has a wireless access point having a first communications interface for wireless communications with a subscriber unit and a second communications interface for communications with a cellular radio communications system, and a secure gateway between the access point and the cellular system to secure communications between the access point and the cellular system.

Abstract: A method, server and terminal for providing a Voice Call Continuity (VCC) service, are discussed. According to an embodiment, the terminal includes a storage unit to store operator policy information and user preference information, the user preference information including domain selection information specifying a user-preferred domain to be used when originating the outgoing call from the terminal, the operator policy information including domain selection information specifying a network-preferred domain to be used when originating the outgoing call from the terminal; and a controller to perform a domain selection for the outgoing call based on the user preference information or the operator policy information.

Abstract: Quality of Service (QoS) and/or bandwidth allocation can be provided for interactive gaming via a Regional/Access Network (RAN). The RAN can provide end-to-end transport between a Gaming Application Service Provider (GASP) and a Customer Premises Network (CPN) that includes a Routing Gateway (RG). Quality of Service (QoS) and/or bandwidth allocation can be provided by determining different levels of subscription service for different subscribers to an interactive game between a plurality of participants. Different QoS and/or bandwidth allocations can be provided to different gaming traffic flows associated with the different subscribers to the interactive game based on the different levels of subscription service of the different subscribers.

Abstract: The traffic of wireless communication between wireless access points can be lowered without requiring a change in the access point to which an associated station belongs, to thereby efficiently use a band for the wireless communication. For this purpose, (1) a station sends a packet, and a wireless access point and a wired access point receives the packet; (2) the wired access point stores an identifier indicating proxy transmission in an ACK packet and then transmits the ACK packet, and (3) the wireless access point confirms the proxy processing identifier of the ACK packet from wired access point and stops delivery of packets to the backhaul. The wired access point delivers the packet to a destination of the packet.

Abstract: The present invention relates to a wireless network, and more particularly, a method (*“and system”?*) which a mobile station discovers a neighbor network in an Internet protocol based wireless network, a network management server for enabling the method, and a method of operating the network management server.

Abstract: An interconnect block for a data processing apparatus, said interconnect block being operable to provide data routes via which one or more initiator devices may access one or more recipient devices, said interconnect block comprising: a first and a second portion; said first portion comprising at least one initiator port for communicating with one of said initiator devices, and at least one recipient port for communicating with one of said recipient devices; said second portion comprising at least two recipient ports for communicating with at least two recipient devices, said second portion being connected to said first portion via at least two parallel connecting routes, said at least two recipient ports being connectable to said at least two parallel connecting routes; wherein in response to a request received from one of said initiator devices at said first portion to perform a transaction accessing one of said at least two recipients in communication with said second portion, said interconnect block is op

Abstract: Methods for multiple-termination routing in a wireless network environment that includes an Internet Protocol (“IP”) core are described herein. One method includes receiving a call delivery request. The call delivery request includes a called number. The call features associated with the called number are determined. A determination is made whether the call features include multiple-termination routing information for a plurality of potential terminating devices. When the call features include the multiple-termination routing information including at least one termination to be routed utilizing Session Initiation Protocol (“SIP”), a call setup is initiated to each of the plurality of potential terminating devices.

Abstract: Disclosed is a radio (wireless) communication system providing a radio communication service and a terminal, and an operation method of a Packet Data Convergence Protocol (PDCP) entity in a Universal Mobile Telecommunications System (UMTS), and more particularly, to an operation method of transmitting, by a transmitting side (transmitter), data blocks by including a Connection Frame Number (CFN) therein such that a receiving side (receiver) processes each of received data blocks by a reference time, as a method of providing a Circuit Switched (CS) service using a High Speed Downlink Packet Access (HSDPA) or High Speed Uplink Packet Access (HSUPA) technology.

Abstract: When receiving an extension connection request from a first mobile terminal, a first route selection device selects a first extension connection route from among a plurality of extension connection routes and transmits the type of the first extension connection route and calling-side address information used on the first extension connection route to a second route selection device. The second route selection device selects a second extension connection route from among the plurality of extension connection routes and transmits the type of the second extension connection route and called-side address information used on the second extension connection route to the first route selection device. If the second extension connection route coincides with the first extension connection route, an extension connection is established between the first mobile terminal and the second mobile terminal via the first extension connection route.

Abstract: The objective of the present invention is to provide a lower main device capable of connecting a non-IP terminal such as a digital terminal to an upper main device using an IP network and of accommodating call control. The lower main device behaves as one or more IP terminals toward the upper main device and houses one or more non-IP terminals which each correspond to the one or more IP terminals.

Abstract: Traditional systems and methods of billing for text messages for prepaid customers are improved by offering a service provider a series of features relating to intelligent and customized billing. These features include the ability to determine priority of messages sent for certain purposes such as SMS Voting, participating in contests, and other uses, and subsequently processing the billing requests generated by these messages in a dynamic manner in real time or close to real time through dynamic late-billing. The service provider has the option of arranging relevant billing requests in queues and queue groups, and allocating system resources for processing billing requests in order of priority. The end result is increased revenue for the mobile operator and greater customer satisfaction.

Abstract: In one embodiment, a system includes a local processor, a peripheral component interconnect express (PCIe) switch electrically coupled to the local processor, one or more local I2C bus devices, a dedicated processor electrically coupled to the one or more local I2C bus devices and the PCIe switch, and a local network switch electrically coupled to the dedicated processor and the PCIe switch, wherein the dedicated processor is adapted for routing interrupts from the one or more local I2C bus devices to the local processor, and wherein the local processor is adapted for handling the interrupts from the one or more local I2C bus devices. Other distributed fabric protocol (DFP) systems, computer program products, and methods are presented according to additional embodiments.

Abstract: A method and apparatus for enabling broadband access providers to use the subscriber service address associated with the Media Access Control (MAC) address and the Internet Protocol (IP) address of the subscriber's Digital Subscriber Loop (DSL) modem or cable modem as the service address that is sent to the appropriate Public Safety Answering Point (PSAP) are disclosed. The present method enables a mapping between a subscriber's VoIP telephone number, broadband access modem MAC and IP addresses, and the subscriber's registered broadband service address to determine the correct PSAP to route an E911 call to.

Abstract: In one embodiment, a scheme is disclosed for effectuating a call in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network. Call information associated with a call is sent from a user equipment (UE) device to an application server (AS) node disposed in the IMS network. The call information includes at least one of a call reference number and a called party's URI. When a message is received at the UE device from the AS node, which message includes the call reference number and an IP multimedia routing number (IMRN), the returned call reference number is verified that it remains valid based on a local timer mechanism associated with the UE device. The IMRN is then sent to the application server in order to facilitate a session with respect to the called party.

Abstract: A system may include a destination device that receives information about a caller via a first outgoing leg and exchanges content with the caller via the first outgoing leg. The system may further include a network device that receives an incoming call from the caller, establishes the first outgoing leg, connects the incoming call to the first outgoing leg so the content can be exchanged, monitors the first outgoing leg, selectively establishes a second outgoing leg in response to the monitoring, and connects the incoming call to the second outgoing leg.

Abstract: Methods, devices, and storage media provide for receiving a message pertaining to a telephone call set-up; identifying in the message a called telephone number; selecting a calling telephone number based on the called telephone number; replacing a calling telephone number included in the message with the selected calling telephone number; and transmitting the message that includes the selected calling telephone number.

Abstract: Described is automatically testing the quality of an audio channel between a caller and a callee that includes a device under test, such as a VoIP or other gateway. An analyzer receives timestamps from a caller and callee during a calling session, including timestamps for when the callee initially provides audio (e.g., speech) to the caller, when the caller initially detects sound, when the caller initially provides audio to the callee, and when the callee initially detects sound. The analyzer uses the relative timing of the timestamps and the speech recognizer's outcome to determine whether the audio channel is experiencing interference or echo. When the audio includes speech, a confidence level corresponding to accuracy of speech recognition also may establish the audio channel's quality. Random selection and timing of output may be employed, such as to vary the testing patterns during repetitive tests.

Abstract: An apparatus and methods for utilizing a stand-alone Voice over Internet Protocol (VoIP) telephone with traditional plain old telephone service (POTS) telephone support is disclosed. A stand-alone VoIP telephone with an analog telephone adaptor supporting a plurality of external analog telephone sets and a digital Internet protocol (IP) interface to support VoIP voice data communication over a data channel is presented. The VoIP telephone is used to establish VoIP service line data connections between a VoIP telephone and a data switch to an IP network. The VoIP telephone receives an analog telephone signal from an external analog telephone and establishes a voice connection between the external analog telephone and the VoIP telephone.

Abstract: Embodiments of the present invention provide systems and methods for recording media communication. An exemplary method comprises the steps of: receiving an incoming call from a calling phone; routing the incoming call to a called phone in a contact center; transmitting voice communications associated with the incoming call via the calling phone; duplicating the voice communications via the called phone; transmitting the duplicated voice communications via the called phone; and recording the duplicated voice communications.

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: Virtual machine environments are provided in the switches that form a network, with the virtual machines executing network services previously performed by dedicated appliances. The virtual machines can be executed on a single multi-core processor in combination with normal switch functions or on dedicated services processor boards. Packet processors analyze incoming packets and add a services tag containing services entries to any packets. Each switch reviews the services tag and performs any network services resident on that switch. This allows services to be deployed at the optimal locations in the network. The network services may be deployed by use of drag and drop operations. A topology view is presented, along with network services that may be deployed. Services may be selected and dragged to a single switch or multiple switches. The management tool deploys the network services software, with virtual machines being instantiated on the switches as needed.

Abstract: Network devices provide seamless offloading of data communications from a service provider's cellular network to an alternate access network outside of the service provider's cellular network. After a cellular mobile device has initially been authenticated by devices in the cellular network to communicate with a packet data network via the cellular network, the cellular mobile device is configured to leverage its prior authentication on the cellular network and automatically obtain credentials for use in logging onto the alternate access network, without requiring any action by a user of the cellular mobile device. After the cellular mobile device is logged on to the alternate access network, the cellular mobile device sends wireless data communications to the packet data network via the alternate access network instead of the cellular network.

Abstract: Methods for enabling floor control are performed in an endpoint terminal (140, 142, 146) and a session controller (106). The session controller: receives (410) a message requesting a session that includes an SDP packet with a requested floor control profile having a requested set of floor control parameters or a requested list of potential floor owners; accepts (410) the session; determines (420) a final floor control profile having a final set of floor control parameters and a final list of potential floor owners; and communicates at least a portion of the floor control profile within a system (100). The endpoint terminal: communicates a message requesting a session that includes an SDP packet with a requested floor control profile; and receives a message having an SDP packet with a final floor control parameter. The final floor control profile may be the same as or different from the requested floor control profile.

Abstract: Access control methods include receiving an access authorization message from an authentication server computer at a blocking device that connects a first network to a second network, modifying access criteria of a transparent firewall at the blocking device responsive to the received access authorization message and operating the transparent firewall according to the modified access criteria to control transfer of messages between the first and second networks. The invention may also be implemented as apparatus and computer readable media.

Abstract: A system including a location server configured to receive registration data from an IP device and a location database configured to store location information of the IP device, wherein the location server queries the location database to determine whether the registration data matches the location information stored at the location database.

Abstract: The present disclosure includes a system and method for blended telephone network and computer network customer support interactions. The system may include a web server, one or more controllers, a telephone exchange, a monitoring server, and one or more client devices. The method may include receiving a multimedia support request, establishing a multimedia communications session, instructing a telephony application associated with the multimedia communications session to place a telephone call, routing and connecting the telephone call to an available endpoint, identifying the available endpoint when the telephone call is connected to the available endpoint, and transmitting an invitation to participate in the multimedia communications session to the available endpoint.

Abstract: A mechanism for managing and enhancing communication streams in a VOIP service using SIP is discussed. SIP is used to manage and enhance a communication stream at a centralized switching element in the VoIP environment rather than at the communication stream endpoints. The communication stream set up by SIP is directed through a central switching element under administrative control rather than traveling directly between the stream endpoints. As a result, the media stream may be conditioned with treatments and may be provided with enhanced services at the central switching element without the specific treatments and services being supported at both stream endpoints. The use of the centralized switching element allows full administrative control of the VoIP environment.

Abstract: A system comprising a first network gateway coupled to a circuit-switched network, the first network gateway to initiate a tunnel through the circuit-switched network responsive to packetized fax relay data received from a packet-switched network, and a second network gateway couple to the circuit-switched network, the second network gateway to receive packetized fax relay data through a tunnel over the circuit-switched network from the first network gateway.

Abstract: Disclosed is a method and system for mutually exchanging capability information of UEs for communication between a Circuit-Switched (CS) and IMS services (CSI) UE, which can use a CS call for voice communication while using an IMS session for PS service, and an IMS UE which uses an IMS session for both voice communication and PS service. A MGCF performs interworking of capability information between a CS message and an IMS message so that the IMS UE can process UE capability information through an IMS session request/response. When a call request message containing no capability information is created from a UE not supporting capability exchange, a CSI AS transfers the call request message to a peer UE after inserting estimable capability information of the UE into the call request message, and stores capability information of the counterpart UE for the UE.

Abstract: Method and apparatus for voice traffic management in a data network includes establishing a default maximum bandwidth setting at a LAN egress port when voice-type traffic is not present in a LAN portion of the data network, detecting voice-type traffic, reducing the bandwidth setting at the LAN egress port to effect a change in a rate of non voice type traffic and monitoring non voice type traffic and voice quality statistics to determine if the rate of non voice type traffic entering the data network has changed. Once the desired change has occurred, performing a linear increase of the bandwidth setting at the LAN egress port to a first value while monitoring voice quality statistics, determining if voice quality has degraded during increase of the bandwidth setting and repeating the last two steps if voice quality has not degraded.