1075 results about "Call setup" patented technology

A method adapts scheduling priorities in a CDMA wireless communications system to conserve battery power in mobile terminals operating within the system. A base station, within the system, receives battery power level information and other setup information from mobile terminals operating within the service area of the base station during call setup procedures. Based on the battery power level information and other setup information, the base station adapts scheduling priorities for the mobile terminals to expedite wireless transmissions from those mobile terminals reporting low battery power levels. The base station schedules the transmissions from low battery power mobile stations to be clustered together in a low-power time slot which is separate in time from the scheduled transmissions from high battery power mobile stations. The base station transmits a power control message to the low battery power mobile stations, to reduce the transmission power required for those mobile stations during the low-power time slot.

A push-to-talk (“PTT”) group call system, for use as, e.g., a public safety wireless network, includes a CDMA-based 1x-EVDO radio access network operably connected to a PTT server over an IP network. The radio access network includes base stations for radio communications with a number of distributed mobile stations. In carrying out wireless communications, the group call system combines IP-based voice and other real-time multimedia services with the 1x-EVDO radio access network's Broadcast Multicast Service. This allows a number of users to receive the same copy of an IP-based media stream for point-to-multipoint, group transmissions. To reduce call setup times, the group call system uses “standing” call groups, which are ongoing group communication channels pre-established between the PTT server and authorized group users. Thus, mobile stations link to one or more standing call groups of interest upon power-up, prior to users speaking.

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.

Firewalls and network address translators (NAT) provide many advantages for client and the Internet itself, however, these devices break many existing transmission control protocol (TCP) / Internet Protocol (IP) applications, since they conceal the identity of IP clients (i.e., peers) and block transmission control protocol (TCP) call setup requests. Firewalls and NATs make it impossible for one TCP peer to discover another and establish a connection. Embodiments of this invention provides a system and a protocol to enable two TCP peers that exist behind one or more firewalls and NATs to automatically setup a true peer-to-peer TCP connection and exchange data without making changes to the firewall or NAT devices or existing TCP-based applications. In embodiments of this invention, the synchronization between the blind TCP peers is achieved using a system that consists of a registration server, an agent application, and a virtual network interface that together relay and replicate the control signals between the two TCP peers. In addition, embodiments of this invention are also used to traverse the NAT and establish a bi-directional peer-to-peer TCP connection in the firewall.

A telecommunications system wherein the dialing of a directory number which has been ported triggers an intelligent network signal which is directed to the Internet. The signal is transmitted through the Internet to a database in the Internet. The database returns call set up directions which are used by the originating switching system to establish a voice link from the calling station to the station having the ported number. The intelligent network signal may be transmitted from the originating switching system to a signal transfer point (STP) and to the Internet. Within the Internet the signal is directed to an Internet database where a real number for the ported number is obtained. This is transmitted back to the signal switching point, which then uses the real number to complete the connection. In another embodiment, where a number has been ported to a station that is connected only to the Internet, the Internet database may provide a domain name address. This address is then used to establish a link through the Internet between the calling and called station. In this instance the Internet handles both the signaling and the voice connections.

Systems and methods for reducing latency in call setup and teardown are provided. A network device with integrated functionalities and a cache is provided that stores policy information to reduce the amount of signaling that is necessary to setup and teardown sessions. By handling various aspects of the setup and teardown within a network device, latency is reduced and the amount of bandwidth needed for setup signaling is also reduced.

A method and system for providing call forwarding in an IP telephone network is disclosed. First, when a telephone number for a first telephone from a second telephone is dialed, the call is routing to a call manager. Call setup procedures with a BRG responsible for the first telephone are then initiated. The BRG then checks stored call forwarding profiles to determine whether there is an active call forwarding profile for the first telephone. The call is connected to the first telephone if an active call forwarding profile is not found. However, if an active call forwarding profile is found, the call forwarding information is sent to the call manager. The call is routed to at least a third telephone based on the call forwarding information.

A system for enabling setup and control of conference calls using short message service (SMS) messages (SMs). The SMS messages may be used to set up, or control a scheduled or real-time conference call. The SMS messages can also be used to exercise real-time control over a conference call. Invitations can be sent in real time to add participants to a call in progress, or to transfer the call from any participant's telephone to another telephone connected to the PSTN.

A dispatch call setup method selects (101) either a forced dispatch call (105) or an invite dispatch call (103). The originating unit (20) of the dispatch call may select either option. Depending on how the required target users (30, 40) respond, the originating terminal has the option to complete the call (119). The terminating unit may accept, reject or convert the forced dispatch call (127). Further, the target may establish preset preferences which accept, reject or allow user controls for an invite dispatch call (139).

PCT No. PCT / SE96 / 00210 Sec. 371 Date Aug. 15, 1997 Sec. 102(e) Date Aug. 15, 1997 PCT Filed Feb. 16, 1996 PCT Pub. No. WO96 / 25830 PCT Pub. Date Aug. 22, 1996The position of a mobile station in a cellular mobile telephone system, particularly a GSM system, is determined by carrying out a simulated call setup, i.e., the call setup is interrupted subsequent to a telephone switching center (MSC) having received a paging response containing the identity of the cell and, optionally, a timing advance. The simulated call setup is initiated by generating a modified short message signal (SMS) which is not registered in the SMS catalogue of the mobile station and which is not shown to the user of the mobile station. The SMS commands the mobile station to carry out a position determining sequence in order to establish parameters for use in establishing the position of the mobile station, for example by commanding the mobile station to connect itself to a base station contained in its neighbor list, analyze the geographical position of the base station, and subsequently send the position determining parameters to a position handler. The geographical position of the base station is analyzed from the cell identity and, if available, the timing advance, the position of the mobile station being presented graphically on a picture screen and constantly updated after each call setup. The call setups are generated by the position handler.

Two advanced group-based voice services, known as Push-to-Conference (P2C) and Push-to-Message (P2M) services are provided by a real-time exchange (RTX) that interfaces to the wireless network. The RTX provides a full-duplex Push-to-Conference (P2C) session between an initiator and two or more other participants, wherein the P2C session comprises a full-duplex conference call, and both the real time exchange and handsets participating in the P2C session communicate with each other using call setup and in-band signaling within the wireless network. The realtime exchange may be coupled to and work with a Push-to-Message (P2M) server to deliver multimedia messages in a non-real time manner from an originator to one or more recipients, without establishing voice paths between the originator and recipients, wherein the P2M server, and an optional Voice Mail Server, provide a message storage facility for the multimedia messages. Additionally, a technique based on handset speech buffering permits zero delay call set-ups to be achieved.

A system and method are provided for performing voice data handoff from a cellular network to a portable Internet / wireless local area network (WLAN) network by a mobile terminal in a heterogeneous network environment. The system and method are provided wherein whether to perform handoff is determined, and a location of a mobile terminal in the portable Internet / WLAN network is registered, a request for handoff is sent to the cellular network, and in response thereto, a request for voice-over-Internet protocol (VoIP) call setup through a specific upper node of the portable Internet / WLAN network is received, and the VoIP call to the upper node of the portable Internet / WLAN network is set-up, and resources of a circuit voice call to a specific upper node of the cellular network are released.

An architecture, client specification and application programming interface (API) for supporting advanced voice services (AVS) for use in handsets in order to support advanced voice services (AVS) for wireless communications systems. The handset or mobile unit executes a client application therein for performing the call setup and in-band signaling with the wireless network for the group voice services, and executes a presence / group management application therein for performing presence and group management functions related to the group voice services in the mobile unit.

A system and method of providing a subscriber service to service users in a telecommunications network. In networks utilizing Session Initiation Protocol (SIP) control signaling for call setup and control, the SIP REGISTER message is modified to indicate service capability information and optionally a traffic load indication for service providers. The REGISTER message is sent to a modified Presence and Instant Messaging (PIM) server that stores presence information and the service capability information for registered service providers. The PIM server then notifies subscribing service users of the identity of the service provider that is registered on the network. The PIM server may utilize the traffic load information to balance the traffic load between service providers by providing users with the identity of the service provider that is the most lightly loaded.

A technique and apparatus to provide status tracking of presence and / or location of a mobile, wireless device to a requesting entity even outside of a particular wireless system. This allows wireless service providers the ability to monitor and log changes in the status of mobile stations within and / or outside their networks. Embodiments are disclosed wherein presence and / or location information is provided to entities outside of a particular servicing wireless network using the mechanisms of call processing components of a mobile network (e.g., call setup procedures), and using standard mechanisms currently available to any appropriately conforming Mobile Switching Center (MSC) element. In one disclosed embodiment, a wireless chat tracking system is implemented which utilizes a change in mobile registration status to automatically notify a chat group system outside the wireless network of current status information activity regarding a relevant device, e.g., registration activity or inactivity timeout. In the wireless chat automatic status tracking system, a registration notification (REGNOT) message is either explicitly forwarded or copied to an external IP based application (e.g., to a mobile chat group system). The change in mobile registration is communicated via a suitable signaling link (e.g., SS7, TCP / IP, etc.) between a Home Location Register (HLR) and the chat group system. Therefore, instead of a conventionally closed system using SS7 messages, REGNOT messages are pushed out over TCP / IP connections to external applications (e.g., chat servers) to automatically notify the external system of the location of a particular user.

Rate charging rules and principles concerning mobile calls in a cellular communication system can be improved by providing the mobile station with a locator object which, at certain intervals, determines the cell in which the mobile station is currently located. Stored in the mobile station is a list of cells or areas in which a special rate is to be applied in charging for calls originating from the mobile station. The locator object monitors calls made from the mobile station and, when it detects that a call is being made, it determines whether the mobile station was in a denoted special rate area at the time that its location was most recently determined. If so, then the object informs the network that the call is entitled to the special rate. Since the last location function was executed just before call setup commenced, a special rate call is possible even if the mobile station has thereafter moved some distance into a cell applying a normal rate. Disposed in a billing center is a billing object, to which the locator object sends data indicating whether the call was initiated from a cell or area in which a special rate is applicable, as well as data identifying the particular call. The second object receives the billing records generated by the mobile switching center, which also contain call identifying data, and compares the call-specific data in those records with the data sent by the first object. In this way, the second object can identify from the billing records those calls that are entitled to a special rate, regardless of whether the mobile station has moved during call setup from the original cell into a cell where another rate, e.g. a higher rate, is applicable.

A system and method of providing a subscriber service to service users in a telecommunications network. In networks utilizing Session Initiation Protocol (SIP) control signaling for call setup and control, the SIP REGISTER message is modified to indicate service capability information and optionally a traffic load indication for service providers. The REGISTER message is sent to a modified Presence and Instant Messaging (PIM) server that stores presence information and the service capability information for registered service providers. The PIM server then notifies subscribing service users of the identity of the service provider that is registered on the network. The PIM server may utilize the traffic load information to balance the traffic load between service providers by providing users with the identity of the service provider that is the most lightly loaded.

A system and related method for supporting non-intrusive and effective voice communication among mobile users in which voice calls between mobile users are managed based on callee availability, caller-callee relationships, and non-intrusive information exchange, including interactive exchange at the time of call setup but prior to the call being answered. Callee availability can be based on callee device status, calendar activities, location, past behavior and other factors. The caller-callee relationships specify whether the callee is available, unavailable or on restricted availability relative to the caller. The interactive exchange can be implemented using voice and / or text / graphic displays on caller and callee mobile device. Callees are provided with options for handling the call. Callers are provided with information regarding the callee's current and future availability and willingness to receive a voice call from the caller, thus maximizing the chance of caller reaching the callee on every voice call.

Techniques for utilizing multiple carriers to substantially improve transmission capacity are described. For multi-carrier operation, a terminal receives an assignment of multiple forward link (FL) carriers and at least one reverse link (RL) carrier. The carriers may be arranged in at least one group, with each group including at least one FL carrier and one RL carrier. The terminal may receive packets on the FL carrier(s) in each group and may send acknowledgements for the received packets via the RL carrier in that group. The terminal may send channel quality indication (CQI) reports for the FL carrier(s) in each group via the RL carrier in that group. The terminal may also transmit data on the RL carrier(s). The terminal may send designated RL signaling (e.g., to originate a call) on a primary RL carrier and may receive designated FL signaling (e.g., for call setup) on a primary FL carrier.

A method of improving communication in a SIP-enabled communication network which involve a variety of communication devices. By the method, a call, either entirely or just a selected medium stream of the call, may be switched, easily and seamlessly to a different channel for different media, either on the same device or on a separate device connected to the network, either during call-setup or after a call session has been established. The call may also initiate a application sharing session so that caller, callee and those participated in the call (in conference mode) may see the content of the same document or whiteboard on a screen while keeping voice communication over the phone.

Embodiments of the method and system provide network support for automatic anti-virus protection by a telecommunication network for mobile terminals. The method may have the steps of: storing in the telecommunication network at least one anti-virus protection level for at least one mobile terminal; initiating, by the telecommunication network, a call setup for a call associated with the mobile terminal, the call having a data stream; using, by the communication network, the at least one anti-virus protection level to recognize a virus in the data stream intended for the mobile handset; and filtering, by the network, the recognized virus from the data stream before the data stream arrives at the mobile terminal. The system implements the method.

Disclosed is a method for providing service of a multimedia mail box in VOIP service. A service providing method applied to a terminal that processes a multimedia message, comprises the steps of: (a) being in a state in which a terminal at a receiving side cannot make a response to a request from a transmitting side; (b) receiving a call setup request from the transmitting side when a user of the terminal at the receiving side is absent; (c) receiving the call setup request from the transmission side when the user of the terminal at the receiving side is in a busy state; and (d) performing a connection to a location information server and a multimedia mail box according to the presence of a response, from the terminal at the receiving side, to the request from the transmitting side at the steps (a), (b) and (c).

An apparatus and method for rapid translation of geographic latitude and longitude into any of a number of application-specific location designations or location classifications, including street address, nearest intersection, PSAP (Public Safety Answering Point) zone, telephone rate zone, franchise zone, or other geographic, administrative, governmental or commercial division of territory. The speed of translation meets call-setup requirements for call-processing applications such as PSAP determination, and meets caller response expectations for caller queries such as the location of the nearest commercial establishment of a given type. To complete its translation process in a timely manner, a memory-stored spatial database is used to eliminate mass-storage accesses during operation, a spatial indexing scheme such as an R-tree over the spatial database is used to locate a caller within a specific rectangular area, and an optimized set of point-in-polygon algorithms is used to narrow the caller's location to a specific zone identified in the database. Additional validation processing is supplied to verify intersections or street addresses returned for a given latitude and longitude. Automatic conversion of latitude-longitude into coordinates in different map projection systems is provided. The memory-stored database is built in a compact and optimized form from a relational spatial database as required. The R-tree spatial indexing of the memory-stored database allows for substantially unlimited scalability of database size without degradation of response time. Maximum performance for database retrievals is assured by isolating the retrieval process from all updating and maintenance processes. Hot update of the in-memory database is provided without degradation of response time.

Systems and methods for hybrid call setup in a wireless communication system are disclosed. A page is sent on a first network informing a target device of a pending communication on a second network. The page also informs the target device a short data burst (SDB) will be sent on the first network. The short data burst (SDB) is sent on the first network containing setup information regarding the pending communication on the second network.

Methods, apparatus, and software for accessing location based Internet services by automatically populating DNS tables with subscriber location information. The DNS tables are updated responsive to information provided on subscriber call set-up. The location may be derived directly for the content of the call set-up, or may be derived in conjunction with one or more pre-existing databases. Ideally the DNS entry is a mapping from IP address (or subnet address) to location, though a mapping from domain name to location will also be effective. When a location based service is requested by a subscriber, the requested service may interrogate the DNS table to retrieve the requesting user's location without requiring further subscriber intervention.

Methods and apparatus for providing a seamless switching of voice calls between different wireless networks are disclosed. In one illustrative example, a mobile communication device has a processor and one or more wireless transceivers coupled to the processor. The one or more wireless transceivers include a first transceiver portion operative in accordance with a first wireless network (e.g. a GSM / GPRS cellular network) and a second transceiver portion operative in accordance with a second wireless network (e.g. an 802.11 wireless network). A voice call is maintained between the mobile device and a communication terminal through call control equipment. The processor of the mobile device is operative to maintain voice communications for the voice call over a traffic channel established between the mobile device and the first wireless network using the first transceiver portion; cause a connecting call to be established with the communication terminal through the call control equipment in response to a predetermined condition, where the connecting call involves a traffic channel established between the second wireless network and the mobile device using the second transceiver portion; and after the connecting call is established, maintain voice communications for the voice call over the traffic channel established between the second wireless network and the mobile device.

An apparatus and method of variable bandwidth multi-codec quality of service (QoS) control are provided. The apparatus for controlling the QoS of a variable bandwidth multi-codec includes: a network state detection unit detecting a network state including at least one of a packet loss ratio, a packet loss interval, and a packet delay time based on an RTP packet transmitted to and received from a destination for which a call connection is established; and a codec control unit updating a transmission rate by comparing the detected resultant value with an already set reference value and increasing or decreasing the transmission rate, and controlling the variable bandwidth multi-codec to code data with the updated transmission rate. According to the apparatus and method, data can be coded with a codec transmission rate suitable for a network state identified during a voice call after the call is set up.

Systems and methods for reducing latency in call setup and teardown are provided. A network device with integrated functionalities and a cache is provided that stores policy information to reduce the amount of signaling that is necessary to setup and teardown sessions. By handling various aspects of the setup and teardown within a network device, latency is reduced and the amount of bandwidth needed for setup signaling is also reduced.

In a communications network, voice and facsimile calls are established between first and second and second terminals via first and second media gateways in a communications network. Each gateway is capable of autonomous transitioning between a voice over IP (VoIP) call mpde and a facsimile ov er IP (FoIP) call mode. A bearer path for the call is established via a media gateway controller. During call setup, the media gateways exchange confirmation of each gateway's capability to perform autonomous transitioning between VoIP and FoIP call modes. The call is established as a VoIP call, and, on a determination by the media gateways that a facsimile call is required, each media gateway autonomously transitions between its VoIP and FoIP call modes without the intervention of the media gateway controller.