Abstract: A method and system for call capacity control in a mesh network are disclosed. The mesh network uses on-device SIP proxy and includes multiple client devices, mesh points (“MPs”), and mesh access points (“MAPs”). The method of updating call capacity information (“CCI”) is also disclosed. Adaptive Wireless Routing (“AWR”) protocol is used to establish connectivity among all the mesh access points. Messages are sent to one or more of the plurality of nodes on a periodic basis or as triggered by changes in the call capacity of an MAP, wherein the routing update messages include call capacity information. CAC modules on mesh portals or mesh access points can use the call capacity information to build up a CCI database for all the MAPs in the mesh cloud to assist in the CAC decision-making.

Abstract: A telephone network access device switchable between a VOIP service and a PSTN service includes an Rj-11 port, a voice splitter, a first relay, a second relay, a SLIC chip, a CPU, and a ring detection unit. The ring detection unit outputs a first switch signal in response to detecting a ring signal of a PSTN communication. The second relay connects the telephone to the VOIP network in response to a power on event. The CPU controls the first relay and the second relay to connect the telephone to the PSTN network in response to the CPU receiving the first switch signal, and controls the second relay to reconnect the telephone to the VOIP network when the PSTN communication ends.

Abstract: A system is provided for updating the conditions under which a telecommunication device accesses services provided by a telecommunication network. The system includes a network access point through which the device accesses the network, and a database, wherein the system authenticates the device via the access point on the basis of authentication data transferred by the device as well as the database storing the profile associated with the authentication data. The access point controls the conditions under which the device accesses the network services once the device has been authenticated and on the basis of the device profile. The system generates a second authentication command for the device via the access point following an alteration of the profiled associated with the authentication data.

Abstract: A distributed switch may include a hierarchy with one or more levels of surrogate sub-switches (and surrogate bridge elements) that enable the distributed switch to scale bandwidth based on the size of the membership of a multicast group. Moreover, each surrogate may optimize the hierarchy according to one or more optimization criteria. For example, each surrogate in the hierarchy may have the necessary information to ensure that if the next surrogate in the hierarchy is unavailable, the data may be routed to a backup surrogate. The selected hierarchy may be further optimized by skipping surrogates (or a surrogate level) such that the data intended for a skipped surrogate is sent to a surrogate in a lower-level of the hierarchy. This may better utilize the connection interfaces in the transmitting sub-switches and eliminate any unnecessary surrogate-to-surrogate transfers.

Abstract: A device includes a first processing unit and a second processing unit. The first processing unit is configured to execute a performance test on the device. The second processing unit is in communication with the first processing unit, and is configured to migrate an application from the second processing unit to the first processing unit. The second processing unit is further configured to detect a failure of the first processing unit, to migrate the application to a third processing unit in response to the failure of the first processing unit, and to assign a first plurality of ports to the third processing unit in response to the failure of the first processing unit.

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: A server-client system or architecture that allows datacast applications to reliably transport data objects from a network server over a unidirectional packet network (“datacast network”) to one or more clients, each of which may be listening to the packet stream at different times. The invention allows the clients to listen intermittently to the datacast, yet still receive all of the data objects published by the server in a timely manner, and in a way that is more optimal in terms of client resource use. This ensures that listening clients can receive a complete set of the data objects broadcast while being able to conserve client processing and power resources by not requiring continuous listening by the client to the datacast.

Abstract: A method of managing circuit-switched calls in a mobile communications network supporting packet-switched sessions is performed as follows: upon reception by the mobile communications network of a circuit-switched call for a user of the mobile communication network, the network ascertains whether the user is engaged in a packet-switched session. In the affirmative case, the network closes the packet-switched data exchange session prior to attempting to route the call to the user.

Abstract: A session request from a first subscriber is received at a first network component of a packet-based network. The session request comprises a request to establish a communications session between the first subscriber and a second subscriber. In the event the session request originated in a trusted network, the first network component permits access to unique resource identifier (URI) information associated with the second subscriber for use in establishing the communications session via the packet-based network. In the event the session request did not originate in a trusted network and in response to determining a security configuration associated with the second subscriber allows the first subscriber to access the URI information under the circumstances, the first network component permits access to the URI information for use in establishing the communications session via the packet-based network.

Abstract: The present invention relates to providing services for circuit-switched calls from an Internet Protocol (IP) based packet network. In essence, a call originating from or intended for a subscriber that is served by a telephony switch, such as an end office or mobile switching center, in a circuit-switched network is intercepted and routed by the telephony switch to a service node via a subscriber emulator. The subscriber emulator acts as a liaison between the telephony switch of the circuit-switched network and the service node of the packet network. Preferably, the subscriber emulator presents itself to the service node as a user agent, which acts on behalf of a subscriber that is served the telephony switch. The service node provides any requisite processing of the call to implement services for the call.

Abstract: Disclosed are new methods of fax-over-IP (FoIP), modem-over-IP (MoIP) or text-over-IP (ToIP) call establishment based on detecting a combination of contra-propagating signals from calling and answering terminals for autonomous connection of media gateways.

Abstract: In one embodiment, a distributed, dynamic and call based feature controller election is executed in a distributed call processing system which is simple, robust, and consistent. The election is based on which VOIP telephone returns an appropriate SIP message first to a feature controller elector, which may be implemented on a proxy communicating with a wide area network.

Abstract: The present invention provides systems and methods employing Voice over Internet Protocol (VoIP) technology to provide multi-channel, multi-access voice communication capabilities.

Abstract: A distributed switch may include a hierarchy with one or more levels of surrogate sub-switches (and surrogate bridge elements) that enable the distributed switch to scale bandwidth based on the size of the membership of a multicast group. When a sub-switch receives a multicast data frame, it forwards the packet to one of the surrogate sub-switches. Each surrogate sub-switch may then forward the packet to another surrogate in a different hierarchical level or to a destination computing device. Because the surrogates may transmit the data frame in parallel using two or more connection interfaces, the bandwidth used to forward the multicast packet increases for each surrogate used.

Abstract: A communication control apparatus includes a storing unit that stores terminal information including respective identifiers of a plurality of communication terminals, respective user identifiers, and respective media types, each media type indicating data format processed by each communication terminal.

Abstract: In one embodiment, a method can include: (i) routing a primary call from a transferor to a transferee; (ii) routing a consultation call from the transferor to a transfer target; and (iii) forwarding a sequence of session initiation protocol (SIP) messages to a third-party device for SIP media renegotiation for transferring between the transfer target and the transferee.

Abstract: An intelligent service system and method are provided. The method includes: receiving a calling information transmitted by a switching device through an Internet-protocol-based call control protocol and which carries the identification information of the service control function requested by the call as well as the identification information of the call; parsing the received calling information, and after the identification information of the service control function and the identification information of this calling are obtained, initiating an assist request to the service control device; receiving the resource operation instruction, and supplying service resource service accordingly; said instruction is sent out by the service control device after said device has received the assist request and according to the service requirement. The media stream leading to the media gateway is established through the media stream transport protocol based on the Internet protocol.

Abstract: A hosted private branch exchange (PBX) platform includes associated application programming interfaces (APIs) that provide a range of integration points with the PBX platform that, in turn, enables the development of a broad range of applications that can customize and/or enhance the basic functionality of the underlying PBX platform.

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 server for providing a message service. The server receives, via a packet switched WLAN, a request message including a destination address associated with a recipient device. The device of the sender is capable of sending messages via the packet switched WLAN or alternatively via an SMS bearer. The server determines whether the destination address corresponds to a subscriber of a service for receiving the outgoing message via a packet switched bearer. The server sends a response message to the device of the sender. If the destination address corresponds to a subscriber of the service for receiving the outgoing message via a packet switched bearer, the response message provides an indication to the device of the sender to send the outgoing message via the packet switched WLAN. Alternatively, the response message provides an indication to send the outgoing message via the SMS bearer.

Abstract: A method of providing a message service in a sender's wireless device is disclosed. The sender's wireless device may be capable of sending messages via a packet switched base station in a first mode, or via a short message service (SMS) base station that is independent of the packet switched base station, in a second mode. The sender's wireless device determines whether a destination address corresponds to a subscriber of a service for receiving the outgoing message via a packet switched bearer. The sender's wireless device sends, if the destination address corresponds to a subscriber of the service, the outgoing message to the recipient's wireless device via the packet switched base station in the first mode. If the destination address does not correspond to a subscriber of the service, the outgoing message may be sent via the SMS base station in the second mode.

Abstract: Systems and methods are disclosed having one or more mesh network appliances (such as ZigBee appliances); a mesh network telephone base station in communication with the mesh network appliances, the mesh network telephone base station including a telephone jack wired to a plain old telephone service (POTS) or a public switched telephone network (PSTN) land-line; and a telephone in communication with the one or more mesh network appliances including the mesh network telephone base station.

Abstract: Implementations described herein may provide for VoIP phone or server devices, where the phones include visual menus through which a user of the phone can modify options or features relating to the user's account. In one implementation, the phone may receive a data structure defining features relating to communication services for the phone device. The phone may parse the data structure to obtain a menu corresponding to the features and present the menu to a user of the phone device.

Abstract: A telecommunications addressing system/method allowing selection of a telephone instrument device (TID) using arbitrary identifiers is disclosed. The system/method allows a source TID (STD) to select a target TID (TTD) by the use of a Target Telephone Identifier (TTI) data string rather than a traditional numeric telephone identification (NTI). This TTI is then indexed within a TTI/NTI mapping server (TMS) that functions as a hierarchical repository of TTI/NTI mappings. STD/TTD communication is established by first performing a lookup of the STD-selected TTI within the TMS to identify the NTI of the TTD. Once the NTI of the STD has been identified by the TMS, communication between the STD and TTD is established using the NTI via the normal public switched telephone network (PSTN). TMS TTI lookup may be performed via STD TID web application and/or via PSTN infrastructure interface.

Abstract: A distributed switch may include a hierarchy with one or more levels of surrogate sub-switches (and surrogate bridge elements) that enable the distributed switch to scale bandwidth based on the size of the membership of a multicast group. When a sub-switch receives a multicast data frame, it forwards the packet to one of the surrogate sub-switches. Each surrogate sub-switch may then forward the packet to another surrogate in a different hierarchical level or to a destination computing device. Because the surrogates may transmit the data frame in parallel using two or more connection interfaces, the bandwidth used to forward the multicast packet increases for each surrogate used.

Abstract: An apparatus and a system for hybrid-transmitting and bridging of a circuit service and a packet service are provided. The apparatus includes a TDM framing unit, a TDM three-direction cross unit, and a mapping and encapsulating unit. The TDM framing unit implements conversion between a line layer TDM signal outside the apparatus and channel layer TDM signals inside the apparatus, and transmits the channel layer TDM signals to the TDM three-direction cross unit. The TDM three-direction cross unit implements cross scheduling on the channel layer TDM signals from a TDM cross apparatus, the mapping and encapsulating unit, and the TDM framing unit. The mapping and encapsulating unit implements conversion between a channel layer TDM signal from the TDM three-direction cross unit and packet service signals from the packet processing unit.

Abstract: An apparatus for wirelessly connecting a smartphone to a remote FM radio system, for controlling playback of audio content delivered wirelessly to the smartphone via a wide area network, on the remote FM radio system. An apparatus housing receives a DC power signal from a remote DC power source. A control panel has a display and controls for controlling playback of the audio content on the remote FM radio system, at least one of the controls performing a function in common with a playback control appearing on a display of the smartphone. An electronic subsystem communicates with the control panel and wirelessly receives audio content from the smartphone, and communicates the audio content as an FM signal via a user playback command input at the controls of the control panel, to the remote FM radio system.

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: A method and terminal for controlling Voice Call Continuity (VCC) functions initiated by a network, are discussed. According to an embodiment, the method includes receiving, by a terminal, a VCC function notification message, the VCC function notification message including VCC enabling information indicating whether a VCC function of the terminal is to be enabled or disabled; and enabling or disabling the VCC function of the terminal based on the VCC function notification message.

Abstract: A method, an apparatus and computer readable storage media facilitate establishing an email thread utilizing an email application used by a client device that communicates with other client devices across a network, where the email thread includes at least one email circulated to a group of users over the network relating to a discussion topic. The email application is linked with a synchronous communication application so as to establish a real-time discussion between at least some of the users of the group in relation to the topic utilizing the synchronous communication application and so that each email of the email thread is accessible via a graphical user interface associated with the synchronous communication application.

Abstract: Method, apparatus, and computer readable medium for use in a first wireless device, and a method for use in a server. The first wireless device is capable of transmitting messages via a packet switched network and via an SMS bearer via a cellular core network. A numeric destination address associated with a second wireless device is sent, via a packet switched network and the Internet, to a server. The server is not a component of the cellular core network. An indication is received from the server at the first wireless device of whether the second wireless device is associated with a subscriber of a message service for receiving the message being composed via a packet switched bearer. A result of the indication is displayed on the first wireless device. The message is transmitted to the second wireless device.

Abstract: There is a need to enable decompression of a speech signal even if no network synchronizing signal is output from a baseband processing portion. For this purpose, an information processing device includes a first serial interface. The first serial interface includes a notification signal generation circuit that generates a notification signal each time compressed data incorporated from the baseband processing portion reaches a predetermined data quantity, and notifies a speech processing portion of this state using the notification signal. The speech processing portion includes a synchronizing signal generation circuit that generates a network synchronizing signal based on the notification signal. A clock signal for PCM communication is generated based on the network synchronizing signal. A speech signal can be decompressed even if no network synchronizing signal is output from the baseband processing portion.

Abstract: A method and system for presenting an MRT for an IM is provided. In an Instant Messaging Application Server (IM-AS), an MRT presenting rule of the IM is set for a user; when the IM-AS receives an IM from the user, the IM-AS determines, according to the MRT presenting rule of the IM, whether an MRT is to be presented for a receiver of the IM, selects an MRT to be presented from MRT resources of the user when the MRT needs to be presented, and sends address information of the MRT to be presenting in a Resource Server (RS) along with the IM to the IM receiver; and the IM receiver downloads the MRT from the RS according to the address information and presents the MRT.

Abstract: A system and method for providing PBX-like functionality at a mobile location includes the capability of maintaining a network session with a mobile number associated with a remote worker. The system utilizes a remote office platform interposed between a data communications network and a conventional PBX switch. A remotely located individual (such as a telecommuter) “logs in” to the remote office platform and, once the individual is authenticated, PBX-like communication with the mobile device is established. If the device is a computer-enabled device and includes a display, the platform can enable a graphical user interface emulating a PBX station set to the device. Alternatively, the remote worker can enter either spoken commands or series of DTMF sequences to utilize the PBX-like features. For these applications, the remote office platform includes a speech recognition module and a translation table for converting a sequence of DTMF tones into a PBX-like call feature (e.g., transfer, hold, mute, etc.).

Abstract: A visual telephony apparatus, system and method are provided. In one embodiment, the visual telephony method includes a method implemented using a processor, the method including the steps of: detecting at least one telephony attribute associated with a voice call directed from a voice communication device to a publicly switched telephone network number; identifying a visual content that has been preassociated with the at least one telephony attribute; initiating a request for the visual content in response to the detecting step; and receiving the visual content in response to the initiating step, the visual content being received at a display device.

Abstract: A Session Initiation Protocol (SIP) message adapted for use by a multi-media services provider system to form a multi-media communication path between at least a calling communication device adapted to operate using a first protocol (e.g. SIP) and at least a destination communication device adapted to operate using a second protocol, such as Integrated Services Digital Network User Part (ISUP). The SIP message includes a header region having a number of header fields, a first body region having Session Description Protocol (SDP) information related to the calling communication device and a second body region having ISUP related addressing information associated with the destination communication device.

Abstract: A method and system of an embodiment of the present invention may include receiving a call request directed to a called party, identifying announcement information associated with the call request, inserting the announcement information into a field of the call request and forwarding the call request.

Abstract: A portable telephone unit 10 receives management information comprising size information for contents data wanting to be downloaded, and adjusts an environment for receiving the object data, based on size information for contents data contained in the management information. To adjust the environment for receiving the object data it is possible, for example, to determine a communication system for receipt, or to secure a storage region for storing object data that has been received. After that, the portable telephone unit 10 transmits a contents data download request, and receives contents data returned in response to the download request. As a result, it is possible for the communication terminal to download the contents data in a state where the receive environment has been optimized according to the size of the contents.

Abstract: Methods and apparatus to route a communication session directly to a voicemail mailbox are disclosed. An example method includes receiving a communication session initiation message from a caller at a source telephone number, the communication session initiation message including a destination telephone number and a direct voicemail feature identifier, determining that the direct voicemail feature identifier indicates that the caller requests that the call be sent to a voicemail mailbox associated with the destination telephone number, and forwarding the call directly to the voicemail mailbox associated with the destination telephone number without ringing the destination telephone number.

Abstract: The present invention moves service control, including call control, for a user element from a cellular network to a multimedia subsystem (MS), such as the Internet Protocol (IP) Multimedia Subsystem (IMS). Call control is provided by the MS irrespective of whether the user element is using cellular or WLAN access for the call. Call control for originating and terminating calls in the CS or MS as well as transferring calls between a circuit-switched subsystem (CS) and MS is anchored at a continuity control function (CCF) in the MS. All call signaling for the call is passed through the CCF. The CCF is a service provided in the user element's home MS and anchors the user element's active CS calls and MS sessions to enable active roaming across the CS and MS.

Abstract: A system and method for provisioning Internet access to a computing device comprising an Internet access adjustment facilitator arranged to receive a request from the computing device for adjusting Internet access from a paid Internet access mode to a toll-free Internet access mode; and a whitelist in communication with the Internet access adjustment facilitator, the whitelist maintaining a list of web resources available for toll free access by the computing device; wherein upon successful processing of the request, the list of web resources are toll free for access by the computing device. The system may further be adapted for billing/charging based on either pay-per-specified-time model or pay per action model.

Abstract: A method is provided for Voice over Internet Protocol (VoIP) devices to communicate over an Internet Protocol (IP) network. The method includes synchronizing the VoIP devices using one or more dual-tone multi-frequency (DTMF) codes over a telephone network, retransmissions of voice packets in bursts, retransmissions of voice packets following a time lag, adjusting the number of retransmissions based on quality of service, retransmission of a missing voice packet identified in a list received from a peer device, discarding low energy voice frames in a jitter buffer to prevent overflow, stopping playout at a low energy voice frame when the jitter buffer is below a minimum buffer size, and selective transmission and retransmission of voice packets based on their energy levels.

Abstract: A method and an apparatus for providing mobility for a Voice over Internet Protocol Service (VoIP) provided on packet networks are disclosed. For example, the method receives a register request from a user endpoint device and retrieves an Access Point-address parameter from a contact header in the register request. The method then determines a physical location of an access point device in accordance with the Access Point-address parameter and updates location information for the user endpoint device in accordance with the physical location of the access point device.

Abstract: A method and apparatus to restore always on connectivity to user equipment, the method having the steps of: monitoring whether the user equipment has transitioned to a network that supports concurrent voice and data; performing, upon the monitoring step detecting the transition, a routing area update; checking whether the user equipment was in a voice call at the time of the transition, and if yes, sending a data packet to the network.

Abstract: A method of routing calls from a mobile unit to a destination location can include registering the mobile unit with a wireless adjunct, wherein the wireless adjunct is in communication with a switching unit and a packet based network and is operable to receive data from the switching unit and place the data, in packetized form, on the packet based network. The method can further include responding to a call setup request from the mobile unit and querying a subscriber database to determine if the mobile unit is a resident or a guest. If the mobile unit is a resident, communications data can be routed from the mobile unit through the switching unit to a first destination location. If the mobile unit is a guest, communications data can be routed from the mobile unit through an RF source to a second destination location.

Abstract: IMS networks and methods are disclosed for storing information on the access network of a user in a subscriber profile for the user. When a communication device of a user attempts to register with an IMS network, a P-CSCF receives a register request message from the communication device, and identifies access network information for the user. The P-CSCF then transmits another register request message to an S-CSCF that includes the access network information, and the S-CSCF in turn forwards the access network information to an HSS. The HSS then stores the access network information in the subscriber profile for the user. Nodes in the IMS network that retrieve the subscriber profile may acquire the access network information to provide services based on the access network information.

Abstract: A method for switching the session control path of IMS centralized services is provided. When the condition for switching the session control path is satisfied during the ICS session based on the first session control path, the following steps are performed: one party of the ICS UE and ICCF of the ICS session transmits the request of switching the session control path to the other party; the receiving party identifies the ICS session corresponding to the request, and transmits an acknowledgement response to the transmitting party; and the ICCF and ICS UE set the session control path corresponding to the identified ICS session as the second session control path and transfer the subsequent session control information associated with the ICS session via the second session control path; wherein the first or second session control path is one of the PS session control path and the CS session control path.

Abstract: When a location of a portable terminal connected to another network beyond a packet communication network side changes, the gateway apparatus that is arranged between a mobile circuit switching network and a packet communication network (IMS network) and that performs mutual conversion between a circuit switching protocol and a packet communication protocol to execute voice communication, exercises control such that at least one of a sequence number and a timestamp in a packet destined to a portable terminal and output to the packet communication network, is made to vary continuously before and after movement of the portable terminal, in case the portable terminal connected to a network beyond the packet communication network side, moves its location.

Abstract: Systems and methods are disclosed for correlating messages used to deliver an Application Originated (AO) text message to User Equipment (UE). A text message gateway is implemented between a text message application and a packet-switched network that uses Session Initiation Protocol (SIP). The text message gateway receives a delivery request from the text message application that includes an Application Originated (AO) text message intended for the UE. The text message gateway identifies a correlation identifier (ID) for the AO text message, inserts the correlation ID in a delivery response, and transmits the delivery response to the text message application (to store for later use). The text message gateway also converts the delivery request into a SIP request that encapsulates the AO text message, inserts the correlation ID in the SIP request, and transmits the SIP request to the UE.

Abstract: A Multiservice Access Concentrator (MAC) provides a time limit for a first ringing voltage signal in response to an attempted call. The call is attempted via a voice over packet-data-network system (VOPS), wherein the VOPS comprises voice over Internet Protocol (IP), voice over Frame Relay, voice over Asynchronous Transfer Mode (ATM), and voice over High-level Data Link Control (HDLC) network systems. Generation of the first ringing voltage signal is terminated upon expiration of the time limit. A control message is transmitted to terminate the attempted call, wherein the control message is transmitted via the VOPS.