3095results about "Telephone service networks" patented technology

Enhancements of and to cell phone operations are based in whole or in part on determining the location of the cell phone. Systems and methods select and determine locations or areas of importance or relevance, and based on that information and other programmed factors affect or alter the operations of the cell phone. While the systems and methods are illustrated by use of cell phone embodiments and applications, they are equally applicable to virtually any portable or mobile communication device, including for example, wireless laptop computers and PDAs.

A system and method of interactive and live computerized matching in which a first user is provided with a personal data device including a wireless communication system. A secure network is configured to include a server programmed with a personal characteristic database and a database searching algorithm, and a wireless communication system at a specific geographic location that is capable of communicating with the user's personal data device. The first user is allowed to log on through the wireless communication system to the secure network using his or her personal data device, and communicates to and stores in the database personal data defining his or her characteristics and those for his or her ideal match. Either the secure network automatically identifies the first users precise geographic location, or the user himself otherwise indicates and stores in the database of the network an identifying address that indicates his current specific geographic location. The user then instructs the database searching algorithm to search for and identify other users similarly logged on to the network at the same specific geographic location. The user is able to initiate a search of the database for possible matches within the pool of other users also logged on to the network at the same specific geographic location. The secure network then indicates to the first user the results of the search.

In order that emergency service vehicles can be dispatched to the correct destination promptly, accurate information about the location of the caller is needed. Another problem concerns routing emergency calls to the correct destination. For emergency calls a universal code is used such as 911 in North America and 112 in Europe. This universal code cannot be used to identify the destination of the call. These problems are particularly acute for nomadic communications systems such as voice over internet protocol communications networks. That is because user terminals change network location. These problems are solved by enabling the geographical location of the emergency caller to be determined by entities within a packet-based network without the need for modification of existing emergency services network infrastructure.

A networked telephony system and method allow users to deploy on the Internet computer telephony applications associated with designated telephone numbers. The telephony application is easily created by a user in XML (Extensible Markup Language) with predefined telephony XML tags and easily deployed on a website. The telephony XML tags include those for call control and media manipulation. A call to anyone of these designated telephone numbers may originate from anyone of the networked telephone system such as the PSTN (Public Switched Telephone System), a wireless network, or the Internet. The call is received by an application gateway center (AGC) installed on the Internet. Analogous to a web browser, the AGC provides facility for retrieving the associated XML application from its website and processing the call accordingly. The architecture and design of the system allow for reliability, high quality-of-service, easy scalability and the ability to incorporate additional telephony hardware and software and protocols.

Systems and methods are presented for routing 911 or other emergency calls from VoIP terminal equipment, wherein the terminal includes GPS or other means to obtain geographic location information for the current location when a 911 call is initiated, and the terminal includes the geographic location information in a call setup request message to the service provider. Routing logic receives the set request and uses the geographic information to search one or more databases to identify the proper emergency service center to which the call is routed, and also the street address corresponding to the caller's current location. The emergency call is then routed to the selected service center, where the call may be delivered with the street address or the street address information is updated in an ALI database of the 911 system by the routing logic before or during call delivery to ensure the emergency service operator or dispatcher knows where to direct emergency services.

A system for scheduling a conference between callers includes a database that stores scheduling information indicating at least a start time, a duration, and a maximum number of callers for one or more scheduled conferences, the scheduling information reflecting available conferencing resources. A server complex coupled to the database communicates, to a requesting Internet Protocol (IP) user, at least one page including one or more scheduling input fields. The server complex receives scheduling input from the requesting IP user for a requested conference according to the scheduling input fields. The server complex accesses the database to determine, according to the scheduling input, whether sufficient resources are available for the requested conference. If so, the server complex allocates at least some available resources to the requested conference and generates confirmations of the requested conference for communication to the callers.

The present invention relates to an improved order fulfillment system. The system is provided with improved data entry system for selecting items for purchase by a customer, and an improved item collection terminal and order delivery system. The portable terminal to be used for collecting of items is provided with an audio as well as video presentation means which are used to provide assistance to the to terminal user.

A system and method are disclosed for packet voice conferencing. The system and method divide a conferencing presentation sound field into sectors, and allocate one or more sectors to each conferencing endpoint. At some point between capture and playout, the voice data from each endpoint is mapped into its designated sector or sectors. Thereafter, when the voice data from a plurality of participants from multiple endpoints is combined, a listener can identify a unique apparent location within the presentation sound field for each participant. The system allows a conference participant to increase their comprehension when multiple participants speak simultaneously, as well as alleviate confusion as to who is speaking at any given time.

The present invention provides a system and method for routing voice and non-voice communications to an answering point via a packet network. Incoming communications from a communication network are converted into packet signals suitable for the packet network. An answering point and a workstation in the answering point are selected. The call is routed to the selected answering point and workstation over the packet network.

The present invention provides a system and method for a reliable, low-cost, secure Internet Protocol (IP) based network that provides broadband-based voice communications as well as video and data communications. The IP network is arranged to function with the infrastructure of the Public Switched Telephone Network (PSTN), to control telephone calls in SS7 type networks and to provide the features, applications, and services of the typical SS7 networks in a voice over IP network. The present invention supports large effective call volumes, allows accommodation of a wide range of broadband-based service platforms, provides flexibility to support current and future calling feature services, and provides high quality voice transmission.

Packet-based central conference bridges, packet-based network interfaces and packet-based terminals are used for voice communications over a packet-based network. Modifications to these apparatuses can reduce the latency and the signal processing requirements while increasing the signal quality within a voice conference as well as point-to-point communications. For instance, by selecting the talkers prior to the decompression of the voice signals, decreases in the latency and increases in signal quality within the voice conference can result due to a possible removal of the decompression and subsequent compression operations in a conference bridge unnecessary in some circumstances. Further, the removal of the jitter buffers within the conference bridges and the moving of the mixing operation to the individual terminals and/or network interfaces are modifications that can cause lower latency and transcoding within the voice conference.

In accordance with the principles of the present invention, an audio-conference bridging system and method are provided. The present invention discards the traditional notion of a single mixing function for a conference. Instead, the novel, flexible design of the present invention provides a separate mixing function for each participant in the conference. This new architecture is described generally herein as “EdgePoint mixing.” EdgePoint mixing overcomes limitations of traditional conferencing systems by providing each participant control over his/her conference experience. EdgePoint mixing also allows, when desired, the simulation of a “real-life” conference by permitting each participant to receive a distinctly mixed audio signal from the conference depending on the speaker's “position” within a virtual conference world. The present invention also preferably accommodates participants of different qualities of service. Each participant, thus, is able to enjoy the highest-level conferencing experience that his/her own connection and equipment will permit.

A unique method for providing entertainment to a portable device, such as a wireless communication device, is disclosed. The method includes steps of receiving location information related to said wireless communication device; providing challenges associated with a game to a user of a wireless communication; providing rewards in response to answers based upon the location of said wireless communication device. The method finds particular application in vehicles having telematics communications products.

A method and apparatus for adding a member to an active call in a group communication network provides for receiving a member list from a user and sending a request to a server to add the member list to the active group call. The method and apparatus further provides for announcing each member in the member list that they are being added to the group call, receiving acknowledgement from a member who wishes to participate in the group call, and forwarding media to the member. The method and apparatus also provides for a significant reduction in the actual total dormancy wakeup time and latency by exchanging group call signaling even when mobiles are dormant and no traffic channel is active.

A multimedia telecommunication automatic call distribution center which allows access to the call center via a plurality of access means, including telephone and data networks to provide simultaneous voice, data, and video access, and ensures effective transparent spreading of agents over different geographical locations. The automatic call distribution center of the present invention uses a multimedia automatic call distribution server acting as connection manager for data network callers and provides for automatic caller identification.

A wireless system is provided for dynamically configuring a multimodal user interface based on user preferences to communicate with a local appliance. An appliance node is operably linked to the local appliance and configured to transmit interface format data of the local appliance for reception by the portable terminal when in the transmit vicinity of the appliance node. The portable terminal transmits an interface request therefrom in response to the reception of the interface format data. A server node is configured to receive the interface request from the portable terminal for identifying a user interface for use in interacting with the appliance. The server node transmits the identified user interface or the information for generating user interface to the portable terminal and the portable terminal presents the user interface or generates the user interface based on provided information to a user via a display of the portable terminal wherein the user may interact with the appliance in accordance with the identified user interface.

A system and method for performing messaging between telephones in a data network telephone system. The message communications may be managed by a messaging application connected to the data network. The messaging application detects when a telephone is connected to another telephone and sends a messaging input screen to the first telephone. The messaging application may determine the availability of other users for messaging and may control messages communicated between the first telephone and the other users. The messages may be text or voice messages.

A system and method for providing Value-Added Services (VAS) in an integrated telecommunications network having a packet-switched network portion (PSN) operable with Session Initiation Protocol (SIP). The integrated telecommunications network includes a SIPext SSP server, a trigger server, and a service node having a Service Logic Program (SLP) that is operable with Intelligent Network Application Protocol (INAP). The SIPext SSP and service nodes are provided with the capability to communicate using SIP-compliant messaging. New header fields are provided that specify operations to be performed by the service node with respect to a service. INAP service parametric data is also provided in the header fields in a sequential form. When a call is received in the SIPext SSP server for a user having a subscription for a VAS, it queries the user profile stored in the trigger server. If the user is subscribed for a service, a SIP request message is formulated based on the user profile, wherein appropriate headers are populated with relevant parametric information and call context data. The service node launches the SLP based on the information provided in the request message and sends a SIP response message to the SIPext SSP server with an instruction concerning the provisioning of the VAS. The SIPext SSP server, thereafter, takes an appropriate action based on the response message and any parametric information contained therein.

Embodiments of the invention provide systems and methods for determining location of a principal. According to one embodiment, a method of providing location information for a principal can comprise receiving a presence event related to the principal. A location of the principal can be indicated by or determined based on the presence event. The location of the principal can be stored as a location attribute of a presence profile of the principal. The method can further comprise providing access to the location attribute of the presence profile of the principal to one or more subscribers or users of a presence service. In some cases, providing access to the location attribute of the presence profile of the principal to the one or more subscribers can be based on one or more policies of the presence service.

In a VoIP system, a method and apparatus for tracking emergency callers is provided. A VoIP service provider network includes a plurality of VoIP phones and is connected to an emergency service provider system. The emergency service provider system includes a call server connected to the VoIP service provider network; a subscriber database; a VPC SBC; and a media gateway for connection to a PSTN. The call server is adapted to receive an emergency call from a VoIP telephone in the VoIP network; verify if the SIP URI has a DID bound to the SIP URI; if the SIP URI does not have a DID bound to the SIP URI, obtain a temporary DID from a DID pool and temporarily bind the temporary DID to the SIP URI; and forward the call to an appropriate PSAP in the PSTN. Should the emergency call be dropped, a person at the PSAP can call back the emergency caller without unnecessary delays.