Abstract: A communication system comprises a communication platform and a wireless service node. The communication platform receives first signaling for a call to a telephone number, and in response, transfers second signaling to alert a packet network appliance of the call, and the communication platform transfers third signaling. The wireless service node receives the third signaling, and in response, transfers fourth signaling to alert a wireless telephone of the call. The wireless telephone and the packet network appliance are simultaneously alerted of the call.

Abstract: A variety of methods, systems, devices and arrangements are implemented for emergency call centers. According to one such method, a location database is populated from a plurality of endpoint devices. The location database determines locations for nodes in a data transmission route from information received from the endpoint devices. Emergency calls using these nodes are located using the populated database.

Abstract: A method of controlling an establishment of a multimedia call between at least two communication equipments, a corresponding system, and a corresponding communication equipment comprise a set up of a bearer connection resulting in a creation of a data transmission channel between at least two communication equipments. A data stream is transmitted between the at least two communication equipments after the creation of the data transmission channel for maintaining synchronization of the data transmission channel. Predefined information elements are introduced in the data stream, wherein the predefined information elements indicate local protocol settings for the multimedia call. The predefined information elements are recognized from the data stream, and parameters of an application protocol used for the multimedia call are adjusted on the basis of the predefined information elements.

Abstract: Systems and methods are provided for establishing a communication session. A communication setup request with a service address may be received by a communication server from a first communication device. At least two communication devices associated with the service address may be identified. The communication server may transmit a call setup request to the at least two communication devices. A communication session can be established exclusively between the first communication device and a first one of the at least two communication devices that responds to the call setup request.

Abstract: According to a first aspect of the present invention there is provided a method for enabling two or more TDMA-based wireless networks to share a communication channel, each wireless network implementing superframes. The method comprises within each network, implementing active superframes separated by one or more inactive superframes and only implementing active superframes when the other networks are implementing inactive superframes, wherein a network only communicates during that networks active superframes and does not communicate during that networks inactive superframes.

Abstract: A digital telecommunications system, a method of managing communications in such a system and a program product for managing audio transmission in a digital communications system can include devices at network endpoints selectively, transparently providing voice samples of sufficient quality for authentication and identification during conversations with the devices. The devices can respond to an authentication request by collecting authentication samples of an ongoing conversation with the samples having sufficient detail for authentication. The devices send the authentication samples in parallel that do not disrupt the conversation. Authentication samples may be verified prior to authentication by comparison against the corresponding portion of the ongoing conversation.

Abstract: An antenna for transmission of signals in respective element frequency bands is controlled to reduce power consumption and transmit a good quality signal. A mobile station apparatus performs wireless communication with a base station apparatus using a plurality of element frequency bands having a predetermined frequency bandwidth. The mobile station apparatus includes: a plurality of transmission sections having an antenna and power amplifier; a transmission processing section generating a transmission signal and outputting the signal to at least one of the transmission sections based on information indicating a first transmission mode in which signals are transmitted using a single transmission antenna and a second transmission mode in which signals are transmitted using different transmission antennas respectively.

Abstract: A proxy telephone system and method is implemented on or as part of a communication network. Proxy telephone numbers are assigned to network users or subscribers. The phone numbers for a user's various communication terminals (e.g., GSM wireless unit, PHS wireless unit, home telephone, work telephone) are associated with the user's proxy number in a database. When a caller initiates a call over the network, the system determines if the dialed phone number is a proxy number. If so, the system accesses the list of phone numbers associated with the proxy number. The system automatically sequentially routes the call to the phone numbers associated with the proxy number in a pre-determined order until a connection is established. For example, the system might first route the call to the GSM wireless unit, and then to the PHS unit, and so on, until the user answers.

Abstract: A method and system for communicating a variable set of contextual information relating to a conversation over a communication channel is provided. When the contextual information is exchanged, any authorized sending party of the contextual information can change the scope, content, or amount of the contextual information that is transmitted to a next receiving party in a determined communication channel path. Before transmitting the contextual information, a desirable scope of the contextual information may be determined based on the next receiving party, in conjunction with the sending party's rules. The contextual information may be updated by adding new contextual information and/or deleting part of the contextual information which is outside of the scope. No contextual information may be transmitted if the next destination desires no contextual information or does not have capabilities to receive any contextual information.

Abstract: System and method for capturing, analyzing, and accessing application level activity and other user information on a mobile data network based on various privacy controls. A platform non-intrusively and transparently monitors data activity on a mobile data network in real-time. The platform comprises collectors, data managers, and a reports manager. The collectors capture information from routers and correlate IP addresses with phone numbers. The collectors retain data allowed by a set of capture filter rules. Data managers receive the retained data and augment it with additional information. A set of usage filter rules determines the communications and additional information that can go to each data manager for use in real-time reports on aggregated usage of the network. The report manager works with the data manager to provide real-time reports to an operator. A set of access usage rules determines the reports that the operator can access.

Abstract: A first method for sharing telephone resources includes a VoIP device connecting to a first device over an IP network, receiving a request from the first device to call a second device with a telephone number, connecting to the second device through a telephone system, and transferring voice signals between the first and second devices. A second method for sharing telephone resources includes a first VoIP device joining a group of VoIP devices connected to an IP network to share their telephone resources, receiving from a caller a telephone number to call a device, connecting to a second VoIP device from the group over the IP network, transmitting the telephone number to the second VoIP device so the second VoIP device connects to the device through a telephone system, and transmitting to and receiving from the second VoIP device voice signals between the caller and a recipient at the device.

Abstract: A hybrid Unified Communications (UC) telephony deployment includes users of a tenant that are hosted between a UC cloud deployment and a UC on-premise deployment that offers PSTN connectivity for the users. An identity of a tenant and its' users are maintained consistently between the on-premise and cloud based UC deployment (e.g. telephone numbers, dialing preferences, voice mail . . . ). Each user of the tenant can register with the UC service from one or more locations (e.g. on-premise, off-premise . . . ) whether or not they are hosted by the on-premise deployment or the cloud based UC deployment. Functionality of the UC deployment may also be maintained at one or more locations within the hybrid UC deployment. For example, some services (e.g. voicemail, or other services) can be hosted in the cloud while the remaining services are hosted on-premise. Different Telco providers may be chosen by the tenant to provide PSTN services for one or more users of the tenant.

Abstract: Certain exemplary embodiments can comprise a method of use comprising: for a call between a local IP network and a remote non-IP network, converting between IP packets and PCM robbed bit signaling via a VoIP channelized router; providing the PCM robbed bit signaling to a TDM switch via the VoIP channelized router; and/or converting between IP packets and GR303 call reference values via the VoIP channelized router.

Abstract: An apparatus for processing session related protocol signaling messages includes a simplification element. The simplification element may be configured to receive a message associated with a session related protocol, to determine whether the message is a per call based message and, in response to a determination that the message is the per call based message, to interpret the per call based message without the removed per session based information. The per call based message is free of per session based information that has been removed.

Abstract: Fiber to the user (“FTTU’), fiber to the curb (“FTTC”), fiber to the node (“FTTN”), and fiber to the premise (“FTTP”) platforms (referred to herein as “FTTx”), require plain old telephony service (“POTS”) emulation using voice over IP (“VoIP”) signaling and bearer channels. Such a POTS emulation service requires emulation of existing fault isolation mechanisms in POTS and also must be extended to the topologies in the VoIP environment. The present invention is adapted to provide POTS emulation using existing fault isolation mechanisms in POTS and also provides POTS emulation for topologies in VoIP environments in the scope of FTTx.

Abstract: A method and an apparatus for detecting relocation of endpoint devices used for accessing services provided on packet networks, such as Voice over Internet Protocol (VoIP) networks are disclosed. For example, the method receives a report of a detected inertial movement of a customer endpoint device, and the method then uses the report of the detected inertial movement for determining whether a relocation of the customer endpoint device has occurred.

Abstract: In a communication system using a standby screen, a portable communication terminal accesses a communication server at a predetermined time interval, uses RSS to receive and store new messages in a cyber sub-community in which the portable communication terminal participates, and sequentially displays tag buttons on the standby screen to display new-message arrivals with message-issuing member names and titles displayed on the buttons.

Abstract: A technique for allowing a non-SIP user to call a SIP user includes dialing an established service number that indicates a desire to place a call to a SIP user. The SIP URI of the intended call recipient is included in a call setup protocol message associated with dialing the service number. A non-SIP network recognizes the call to the service number and the SIP URI from the UUI parameter of the call setup message. The call is then routed to a gateway for interfacing between the non-SIP network and the appropriate SIP network where the SIP URI is extracted from the message received by the gateway and used to generate an SIP INVITE message for establishing the call with the intended SIP user.

Abstract: A communication system for implementing IP cross-domain interconnecting via Border media Gateway (BG) includes: different IP domains, each of which includes: a Media Gateway Controller (MGC) for call control and a Media Gateway (MG) for media bearing; the communication system includes a BG for implementing interconnecting of the IP domains as well. A communication method based on the communication system, includes: a calling MG and a called MG located in different IP domains implements call switching and media stream creating with a BG under the control of MGCs in the different IP domains; and the calling MG communicates with the called MG based on the media streams created. Thus the cross-domain interconnecting between different IP domains via BGs is realize.

Abstract: In one embodiment, a device comprises an input port for receiving network packets that are being communicated via a computer network. A processor is in communication with the input port. The processor is configured to facilitate dynamic monitoring and updating of port addresses assigned by at least one non-ALG PAT router so as to facilitate communication of packets that are exchanged between an inside user agent and an outside user agent in a manner that mitigates a likelihood of communication interruption during a VOIP session. An output port is in communication with the processor for transmitting packets via the computer network.