 Patent Application Title |
Patent App Num. |
Date |
 Method to implement e911 services in ims (ip multimedia subsystem) | 20130148549 | 20130613 |
 Efficient processing and/or routing of emergency calls in an IMS (IP Multimedia Subsystem) for VoIP (Voice over IP) subscribers can be enabled. In one aspect, an E911 profile is generated, for example, by an application server, for each VoIP subscriber based in part on current location information associated with the subscriber, such as, the subscriber's IP address. The E911 profile is pushed to the HSS (Home Subscriber Server) of the IMS. The E911 profile can then be stored in the HSS and can be queried by an E-CSCF (Emergency Call Session Control Function) during emergency call processing. Moreover, when an emergency call is received from the subscriber, the E-CSCF can retrieve the E911 profile associated with the subscriber from the HSS and route the call to... |
| Wireless emergency services protocols translator between ansi-41 and voip emergency services protocols | 20130148555 | 20130613 |
 A protocol converter or translator between ANSI-41 ORREQs and VoIP V2 messaging. The protocol converter may alternatively (or also) provide conversion between GMS MAP and VoIP V2 messaging. Interaction of VSPs with a Mobile Positioning Center (MPC) or a Gateway Mobile Location Center (GMLC) is permitted, as is interaction of wireless carriers with a Voice Positioning Center (VPC). In this way existing GMLCs or MPCs may be used to service VoIP 9-1-1 calls. Moreover, operators of Voice Positioning Centers (VPCs) who implement wireless offerings can re-use their existing VPCs to service wireless 9-1-1 calls.
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| Systems and methods for handoff of a mobile telephone call in a voip environment | 20130148567 | 20130613 |
 When a mobile telephony device is conducting a voice over Internet protocol (VOIP) telephone call via a data network, and the mobile telephone device transitions from a first wireless data connection to a second wireless data connection, signaling between the mobile telephone device and a media relay inform the media relay how to address data packets bearing the media of the call after the transition has occurred. In some embodiments, the mobile telephone device establishes multiple simultaneous wireless data connections, and the mobile telephony device switches the media of the call between the wireless data connections depending on the quality of the wireless data connections.
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| Dynamic application integration associated with hosted voip pbx using client-side integration proxy | 20130148648 | 20130613 |
 A system for dynamically integrating and synchronizing a plurality of software applications of an end user as part of a telephonic communication between the end user and a third party managed and handled by a hosted VoIP PBX includes a client side integration proxy having (i) an API for communicating and exchanging data with the software applications and (ii) memory cache dedicated to storing information about the telephonic communication handled and managed by a hosted VoIP PBX, wherein the client side integration proxy assigns a record in the memory cache for storing information about the telephonic communication and enables the software applications periodically to access and dynamically update, modify, or add to the record as the information about the telephonic communication is updated or changed by... |
| Enhanced e911 network access for a call center using session initiation protocol (sip) messaging | 20130149988 | 20130613 |
 A switched emergency call (e.g., a 911 call, an alarm company call) forwarded by a telematics call center is converted into a session initiation protocol (SIP) packetized phone call at the call center, and routed over an IP network, for presentation to an emergency services gateway, which connects to a selective router via dedicated circuits, gaining full access to the Enhanced 911 network. This provides a PSAP receiving a call from a telematics call center or other call center with all features available in an Enhanced 911 network, e.g., callback number of the 911 caller, and location of the 911 caller. Location of the caller is provided using a VoIP positioning center (VPC), queried from the call center. In this way, the switched emergency call is... |
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| Systems and methods for handoff of a mobile telephone call in a voip environment | 20130150038 | 20130613 |
| When a mobile telephony device is conducting a voice over Internet protocol (VOIP) telephone call via a data network, and the mobile telephone device transitions from a first wireless data connection to a second wireless data connection, signaling between the mobile telephone device and a media relay inform the media relay how to address data packets bearing the media of the call after the transition has occurred. In some embodiments, the mobile telephone device establishes multiple simultaneous wireless data connections, and the mobile telephony device switches the media of the call between the wireless data connections depending on the quality of the wireless data connections.
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| Using indirect communication to provide a solution to use international dialing convention and incorporating phone numbers for non-phone devices | 20130142193 | 20130606 |
| An indirect communication system and a method of indirect communication include a mobile phone as either a calling device or receiving device. The mobile phone calls another device to set up a prospective communication, and then uses Voice over Internet Protocol (VoIP) to communicate with the other device over the Internet. The receiving device receives a generated signal notifying the receiving device of a proposed communication with the calling device. A server sets up a meeting point channel after the calling device has connected to the server. The server receives outgoing VoIP packets from the calling device and redirects the outgoing VoIP packets to the receiving device.
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| Systems and methods for ip and voip device location determination | 20130135147 | 20130530 |
| A method and system for precise position determination of general Internet Protocol (IP) network-connected devices. A method enables use of remote intelligence located at strategic network points to distribute relevant assistance data to IP devices with embedded receivers. Assistance is tailored to provide physical timing, frequency and real time signal status data using general broadband communication protocols. Relevant assistance data enables several complementary forms of signal processing gain critical to acquire and measure weakened or distorted in-building Global Navigation Satellite Services (GNSS) signals and to ultimately extract corresponding pseudo-range time components. A method to assemble sets of GNSS measurements that are observed over long periods of time while using standard satellite navigation methods, and once compiled, convert using standard methods each pseudo-range into usable path distances... |
| Transparent bridge device | 20130139246 | 20130530 |
| The device provides protection for VoIP or like time-sensitive traffic. Packets arriving at a network interface in the data link layer are inspected to identify signaling packet, which are then queued for further analysis. The signaling packets are analyzed for compliance with adaptive criteria to determine whether the packets are considered safe to pass to a user, and the signaling packets failing to meet the adaptive criteria are rejected. The adaptive criteria based are updated based on historical data pertaining to the signaling packets from the same source address for the same user account.
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| Local access to a mobile network | 20130121215 | 20130516 |
| Methods and apparatuses for billing a user associated with a mobile service for a VoIP call made by the user using a wireless local area network are disclosed The VOIP call can be received by a local area network. At least a portion of the VoIP call is decoded to obtain session information related to the VoIP call including information identifying the user. The VoIP call is coupled to a public data network. At least a portion of the session information is sent to a provider of the mobile service, to bill the user, and not an owner of the wireless local area network, for the VoIP call. A credit is initiated to an account associated with said owner of the wireless local area network to... |
| System and method for facilitating voip communications | 20130121332 | 20130516 |
| A method for facilitating VoIP communication between VoIP providers. First and second VoIP service providers are registered with a VoIP communication system, which generates a table of registered VoIP subscribers from the first and second VoIP service providers. The information from the table is applied to a call generated from a subscriber on the first VoIP service provider, such that if the desired party is a subscriber to the second VoIP service provider as noted in the table, the call generated from the subscriber on the first VoIP service provider is sent to the desired party as a packet switched call.
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| Methods and apparatus to control a flash crowd event in a voice over internet protocol (voip) network | 20130121333 | 20130516 |
| Methods and apparatus to control a flash crowd event in a voice over Internet protocol (VoIP) network are disclosed. An example method comprises receiving at a VoIP border element a VoIP registration response message having a field representing a priority assigned to a VoIP endpoint, receiving a message from the VoIP endpoint at the VoIP border element, detecting whether a network congestion condition exists, and placing the message received from the VoIP endpoint into one of a plurality of queues based on the priority when the congestion condition is detected.
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| Method and system for dynamically assigning a server | 20130107752 | 20130502 |
| A method of dynamic server assignment in a VoIP network includes detecting a telecommunication device on the VoIP network and querying a geo-location server for geographic information related to the telecommunication device's location and receiving the geographic information. Based on the geographic information a server assignment is determined. A configuration file for the telecommunication device including the server assignment and location is created and the configuration is provided to the telecommunication device over the network. The configuration file may also include features related to local customs or practice.
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| Communication system and communicating connection method thereof | 20130109366 | 20130502 |
| A communication system and a communicating connection method are disclosed. The communication system comprises a first user side, a second user side and a cloud platform. The first user side has a first identity identification code, a first space identification code or a combination thereof. The second user side has a second identity identification code, a second space identification code or combination thereof. The cloud platform comprises a cloud database and a cloud processing module. When the first user side is positioned, the cloud processing module matches and establishes the first user side and the second identity identification code, the second space identification code or the combination thereof, to connect the first user side and the second user side by a VoIP technology.
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| Voip device, voip conferencing system, and related method | 20130094653 | 20130418 |
| Voice over internet protocol (VoIP) devices and conferencing systems may include a spatial encoder associated with a first endpoint and a spatial renderer associated with a second endpoint. The spatial renderer may configured to receive audio data. The audio data may be rendered among a plurality of speakers based on a first set of spatial information for a plurality of microphones associated with the first endpoint, and a second set of spatial information for the plurality of speakers associated with the second endpoint. A method for generating a sound field may include determining spatial information for a plurality of microphones in a local room, determining spatial information for a plurality of speakers in a remote room, mapping the spatial information for the plurality of microphones and... |
| Regional independent tandem telephone switch | 20130089086 | 20130411 |
| Implementations of the present disclosure involve an apparatus and/or method for a regional independent tandem switch of a telecommunications network. The tandem switch processes communications between a long distance carrier and a local exchange carrier through a voice over IP (VOIP) network. By utilizing the VOIP network, the regionally independent tandem may process long distance communications to and from any point in the network, regardless of the physical proximity of the tandem to the originating/destination communication device. The regionally independent nature of the tandem also allows for flexibility in communication routing through the network, load balancing between the network tandem switches and reduction of needed components of the network for proper processing of the long distance communications.
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| Activating and deactivating semi-persistent scheduling for an lte voip radio bearer | 20130083702 | 20130404 |
| A method, an apparatus, and a computer program product for wireless communication are provided. The apparatus determines an operational state of a header compressor or a header decompressor by determining a transition between different operational states associated with the header compressor and/or by determining a transition between different operational states associated with the header decompressor. A persistent scheduling mode is changed in response to a change in the operational state of the header compressor. The persistent scheduling mode may be changed by activating uplink persistent scheduling when the operational state of the header compressor changes from a first order state to a second order state, and/or by deactivating the uplink persistent scheduling when the operational state of the header compressor exits the second order state.
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| Controlling registration floods in voip networks via dns | 20130077487 | 20130328 |
| A mechanism controls global synchronization, or registration floods, that may result when a large number of endpoints in a Voice over Internet Protocol (VoIP) network such as an Internet Protocol Multimedia Subsystem (IMS) come online simultaneously after a catastrophic failure. The mechanism allows the Domain Name System (DNS) infrastructure to efficiently control the overload condition by registering user end points with backup border elements, and by staggering and by randomizing the time-to-live (TTL) parameter in registrations with backup border elements.
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| Connection of telephone calls | 20130077535 | 20130328 |
| A method of connecting a telephone call from a called party to a caller attempting to place an initial call to the called party comprises the steps of identifying an incoming call as originating from the called party using contact information stored in a device of the caller (4), dropping the initial call, and automatically connecting the call from the called party (5). Apparatus for carrying out the method may comprise a mobile telephone, a PABX switchboard, an item of Voice Over IP (VOIP) hardware, a device such as a computer, phone or tablet programmed with VOIP software, or an item of telephone exchange equipment.
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| Method and apparatus for detecting devices on a local area network | 20130080629 | 20130328 |
| A method, apparatus, and system for discovering devices on a local area network. The method comprises sending a DNS request from a host networking element, receiving the DNS request at one or more devices, and responding to the host networking element with one or more DNS responses. The DNS request comprises one or more particular domain names. The one or more DNS responses comprise local network address information for the one or more VoIP devices. The apparatus comprises means for filtering a set of network traffic for a DNS request, means for responding to a DNS request with a local network address, and means for forwarding the DNS request to another device. The system comprises a computer for generating a DNS request for a particular domain,... |
| Overload call control in a voip network | 20130070591 | 20130321 |
| The invention provides a system, method and apparatus for controlling a mass calling event in a VoIP network. The invention provides for monitoring port traffic information on a network edge device and detecting the occurrence of a mass calling event in conjunction with a call control module. Control instructions or information may then be provisioned to network edge devices to alleviate congestion. The invention enables network edge devices to gap or provide other controls to reduce call overloading. Controls for alleviating congestion include blocking the packets, providing an announcement to the call, re-routing the traffic associated with this specific mass calling event, and reducing the quality of the service.
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| Method, system and software for establishing a communication channel over a communications network | 20130070756 | 20130321 |
| The establishment of a VoIP connection between first and second telecommunication devices (109,110) comprises receiving user input data at an input terminal remote (100) from the telecommunications devices (109,110). The input data comprises a user identifier (102) and an identifier associated with one of the telecommunications devices to be called (110). The user input data is passed to a server (103) remote from the input terminal (100). A suitable connection pathway is determined at the server (103) and a communication channel (114,115,116) is established between the devices.
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| Instant internet browser based voip system | 20130070759 | 20130321 |
| The present invention is an instant Internet browser based VoIP system with a VoIP client in the form of temporary VoIP applets that can start in a Web browser and can establish an instant peer-to-peer connection with another web-based or hardware embedded/installed VoIP client using session initiation protocol (SIP) and real-time transport protocol (RTP) audio streaming. The applet is a small file that is easily loaded onto a user's browser and uses application program interfaces (APIs) that require no additional libraries. The applet is written in JAVA, although other programming languages may also be used to write the applet.
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| Activity-and dependency-based service quality monitoring | 20130064110 | 20130314 |
| Many services measure quality of service (QoS) according to abstract metrics based on general heuristics of QoS determinants (e.g., VoIP service providers may presume that QoS is predominantly determined by network performance). However, users' QoS perceptions are often based on their experiences with particular activities of the service, which may utilize different service paths having different QoS determinants. Therefore, QoS may be measured by identifying the activities of the service, and the dependencies among the components of such services; for respective activities and dependencies, identifying a service path from the source to the user, and the segments comprising the service path; measuring the quality of the segments of the service path; and calculating the QoS of the activity according to the QoS of the segments of... |
| Methods and apparatus to measure market share for voice over internet protocol carriers | 20130064242 | 20130314 |
| Methods and apparatus to measure market share for VoIP carriers is disclosed. An example method includes querying a plurality of VoIP carrier servers to determine the VoIP carrier server that owns the telephone subscriber number (SN), in response to the querying, receiving a plurality of messages operable to determine whether the telephone SN is found within any one of the plurality of VoIP carrier servers, when the received plurality of messages is at least one of inconclusive or when the telephone SN is not found within any one of the plurality of VoIP carrier servers, placing a first partial call to the telephone SN from a first VoIP number within a first VoIP carrier network, receiving a first signal from the first VoIP carrier network, and... |
| Instant internet browser based voip system | 20130067559 | 20130314 |
| The present invention is an instant Internet browser based VoIP system with a VoIP client in the form of temporary VoIP applets that can start in a Web browser and can establish an instant peer-to-peer connection with another web-based or hardware embedded/installed VoIP client using session initiation protocol (SIP) and real-time transport protocol (RTP) audio streaming. The applet is a small file that is easily loaded onto a user's browser and uses application program interfaces (APIs) that require no additional libraries. The applet is written in JAVA, although other programming languages may also be used to write the applet.
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| Method for handover from circuit switched domain to packet switched domain, device, and communications system | 20130051368 | 20130228 |
| Embodiments of the present invention disclose a method for a handover from a circuit switched domain to a packet switched domain, a device, and a communications system. One of CS to PS handover methods includes receiving, by a target mobility management entity, a CS to PS handover request message that corresponds to a first UE and is from an MSC; triggering setup of a dedicated VoIP bearer that corresponds to the first UE and is between an SGW and an MGW; sending a handover request message to a target access network to instruct the target access network to prepare a handover resource for the first UE; and sending a CS to PS handover response message corresponding to the first UE to the MSC.
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| Method and system for securely authorizing voip interconnections between anonymous peers of voip networks | 20130054696 | 20130228 |
| A peering authority or settlement clearinghouse can be used to control access, collect session accounting information, and provide financial settlement of interconnect or session fees among anonymous Internet Protocol (IP) peers or networks. The addition of peering policy criteria, such as price and quality of service, to peer to peer route discovery mechanisms enable a trusted intermediary, such as the settlement clearinghouse, to authorize acceptable interconnection or peering sessions between anonymous IP peers. Any financial settlement transactions which result from the peering sessions may be subsequently executed by the settlement clearinghouse.
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| System and method for measuring quality of multimedia service | 20130044620 | 20130221 |
| A system and a method for measuring quality of a multimedia service transceived via a VoIP service and notifying a user of a result of the measurement are disclosed. An embodiment of the present invention includes a VoIP multimedia server configured to loop back sample multimedia data received from a multimedia terminal providing a VoIP multimedia service, analyze a quality measurement result received from the multimedia terminal, calculate a quality measurement score, convert the quality measurement score into voice data, and transmit the voice data to the multimedia terminal, and a multimedia terminal configured to transmit the sample multimedia data to the VoIP multimedia server via a packet network, measure the quality of loopback sample multimedia data looped back from the VoIP multimedia server, transmit the... |
| Mobile terminal and method of controlling the same | 20130044648 | 20130221 |
| A mobile terminal and a method of controlling the mobile terminal are provided. The mobile terminal includes: a display; a wireless Internet module; and a controller for acquiring information about a plurality of voice over Internet protocol (VoIP) accounts that can communicate through the wireless Internet module and for distinguishing information about a first user group corresponding to the acquired plurality of VoIP accounts from information about a second user group not corresponding to the acquired plurality of VoIP accounts and for displaying the information in the display. Therefore, by displaying to distinguish information about a user group corresponding to a plurality of VoIP accounts from other information, a function can be easily selected through a plurality of VoIP accounts.
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| Communication signaling engine | 20110016488 | 20110120 |
| A verbal request for an action to be performed in connection with a user's set-top box may be received by a Program Content Manager (PCM) from a voice-enabled input/output device such as a VoIP-enabled device or a mobile communication device. Alternatively, the user may request an action via email, text message, and the like. The PCM receiving the action request may execute, or schedule the request for later execution, depending on whether the request is for a real-time or future action. The user may also request to send an alert or a program recording request to others. Recipients may be able to request further alerts or confirm a program recording invitation if certain criteria are met.
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| Method and entities for inter-domain handover | 20110013597 | 20110120 |
| Aspects of the present invention relate to methods and entities involved in enabling and improving inter-domain handover. A PS based RAN such as eUTRAN can identify which potential handover to prepare and which handover to initiate, PS-PS or SR-VCC based on: (1) the UE's SR-VCC capabilities, (2) “SR-VCC indication” from the MME, (3) active VoIP bearer, and (4) target cell capabilities.
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| Voice over internet protocol (voip) location based 911 conferencing | 20110013541 | 20110120 |
| Voice Over Internet Protocol (VoIP) emergency calls to an Emergency Response Center (ERC) are handled through a VoIP conference bridge on a VoIP service provider's soft switch. The soft switch works with a VoIP positioning center (VPC) to obtain location information, which is compared against a PSAP database to find an initial best-appropriate PSAP for the location of the emergency caller. The PSAP is issued an Invite message to join the conference, establishing an emergency call. Third parties such as police, ambulance may be issued Invite messages to join the conference. Cold transfers are avoided by Inviting participants to join a single emergency conference rather than passing an emergency call from party to party (e.g., from PSAP to police to ambulance, etc.) The PSAP, other emergency... |
| Enabling incoming voip calls behind a network firewall | 20110010752 | 20110113 |
| A network device is configured to receive a registration message from a private user device including a private internet protocol (IP) address associated with the private user device. A public IP address and discrete port number are assigned to the private user device and private IP address and stored in an incoming call table. The registration message is translated to include the public IP address and discrete port number. The registration message is forwarded to a proxy server for registration. An incoming call invitation message is received from a public user device, where the call invitation message is directed to the public IP address and discrete port number associated with the private user device. The call invitation message is translated to include the private IP address... |
| Method and apparatus for acoustic echo cancellation in voip terminal | 20110002458 | 20110106 |
| A method of acoustic echo cancellation in the VoIP terminal using processing of the far-end signal with the digital adaptive filter in order to obtain the echo estimate that is subtracted from the microphone signal in which the far-end signal, before it is converted to the analog form and passed to the loudspeaker (4), is marked by embedding an encoded digital signature obtained from the signature generator (14) and then detection of the digital signature is performed in the signal collected by the microphone (7) and converted to digital form, depending on the result of the digital signature detection, adaptation of the digital adaptive filter (9) is resumed or stopped. A circuit for acoustic echo cancellation in VoIP terminal contains the digital adaptive filter with the... |
| Systems and methods of deciding how to route calls over a voice over internet protocol telephone call routing system | 20110002330 | 20110106 |
| A system and method of monitoring Voice over the Internet Protocol (VoIP) and facsimile over Internet Protocol (FoIP) calling over the Internet includes compiling information about each call after the call is terminated. By compiling information about each of the calls immediately after they are terminated, the system can quickly generate billing reports. The system can also quickly react to developing problems.
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| Method for dialing from internet extension to conventional extension | 20110002326 | 20110106 |
| A method for dialing from Internet extension to conventional extension is disclosed. A VoIP gateway or an IP auto attendants is used for dialing from Internet extension to conventional extension. The phone number of the Private Branch Exchange and the voice guidance are not needed. The calling number of SIP message is interpreted directly and converted into DTMF (Dual-tone multi-frequency) messages for dialing into a conventional extension.
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| Interactive internet browser based media broadcast | 20100333156 | 20101230 |
| A method of providing one-way video transmission and corresponding interactive two-audio communication to remote recipients via a world wide computer network comprising the steps of creating a source digital video signal, broadcast transmitting the source digital video signal via an internet connection, transmitting a corresponding source digital audio signal a VoIP protocol, and transmitting a recipient audio signal to the source location via an Internet connection. A system for broadcast transmitting a one-way digital video signal and for transmitting and receiving a corresponding interactive two-way audio signal over an Internet connection comprises a broadcast digital video server for broadcast transmitting a source digital video signal, a VoIP audio server for transmitting a source digital audio signal over an Internet connection, and an Internet web page configured... |
| Spam control method and apparatus for voip service | 20100332607 | 20101230 |
| A spam control method and apparatus for a user equipment capable of receiving a VoIP service are provided. The spam control method includes obtaining at least one IP address corresponding to a target domain name, creating a whitelist for spam control using the target domain name and obtained IP address, and performing, when a message is received, spam control by determining whether the received message is a spam message based on the whitelist.
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| Mobile fast alerting | 20100331055 | 20101230 |
| Where a separate bearer channel is required to be set up in a mobile network to complete a hybrid VoIP call on the mobile network as specified in 3GPP ICS standard TR 23.892, an alerting process may be started after a provisionable delay rather than upon completion of the network call setup. This allows the mobile network user to start responding to the impending call before signaling of the call has been completed, so that the person can answer closer in time to completion of call signaling. Additionally, the person who placed the call may be provided with ringing feedback to make it sound like the call has been completed. The amount of delay between receipt of a SIP invite and the onset of an alerting... |
| Method and system for enabling emergency calling from nomadic voip extension telephones | 20100329433 | 20101230 |
| A method and system for handling calls to emergency telephone numbers by a VoIP telephone that is configured as an extension to a home telephone. The VoIP telephone is provided with a home telephone number that is different than the home telephone number. The location of the VoIP telephone is maintained in a caller ID location database and is regularly updated when necessary. Location updates are made at least when the VoIP telephone registers using a different IP address than was used for the most recent communication. The system is configured to enable the VoIP telephone to ring when a call is placed to the home telephone number. In addition, non-emergency calls from the VoIP telephone are accompanied by caller ID information associated with home telephone.... |
| Apparatus and method for preventing spams in voip system | 20100329241 | 20101230 |
| A system for preventing a spam-call for a VoIP system includes a communication network, a plurality of terminals connected via the network, and a server. The server includes a server black list DB, a connection control module, a membership information management module, and a server-side management module. Each of the terminals includes a terminal-side management module. With the system, VoIP spam can be prevented in a cost-effective way and users' convenience can be increased.
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| Voice over internet protocol diagnostics | 20100329130 | 20101230 |
| Embodiments disclose a method that may be used for diagnosing, for example, Voice over Internet Protocol (VoIP) sessions. The method may include establishing a first streaming session between a first telephone and a server using session control messages and establishing a second streaming session between the first telephone and a second telephone using session control messages. The method may further include transmitting or receiving data packets using the second streaming session, wherein the data packets carry voice or video data between the first and second telephones. The method may further include echoing the session control messages used to establish the second streaming session or the data packets carrying the voice or video data to the server using the first streaming session.
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| Graphical user interface (gui) for administering a voice over internet protocol (voip) network implementing media aggregation | 20100325293 | 20101223 |
| A graphical user interface (GUI) is disclosed that allows a user to identify routers, communities, residents and media aggregation managers existing on a network. The user may interactively select a path that includes two media aggregation managers and at least one router. The GUI predicts scheduled bandwidth utilization along the selected path for a variety of residents wishing to communicate across the path. The GUI may be utilized for initializing all media aggregation managers on the network along with provisioning each of the routers on the selected path. The GUI provisions all of the routers and initializes all of the media aggregation managers on the selected path simultaneously in order to accomplish the predicted schedule of usage provided by the GUI to the user and force... |
| Self-configuring cellular basestation | 20100322426 | 20101223 |
| A basestation 110 provides coverage for voice and data services within a building using GSM/GPRS and UMTS interfaces within existing networks, allowing use of existing mobile phones. The basestation also provides flexible interfacing to the network operator's core network 130 via the Unlicensed Mobile Access (UMA) standards as opposed to the usual Iub (UMTS) or Abis (GSM) interfaces used by conventional cellular basestations. Backhaul from the basestation units is achieved through the use of Digital Subscriber Line (DSL) connections over fixed telephone lines. This approach enables low-cost transport of data and voice using VoIP.
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| System and methods for enhanced multimedia in voip networks | 20100322403 | 20101223 |
| Calls placed to a call recipient bypass the terminating messaging system, creating an enhanced message in the call originator's messaging system and delivering the enhanced message using alternate routing information. A caller creates enhanced messages for a call recipient even when the call recipient's messaging system does not support such functionality. The caller also embeds stored content and/or use links in the enhanced messages to allow the message to be modified and/or deleted before it is accessed by the call recipient.
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| Voip device and method for adjusting interrupt time thereof | 20100322231 | 20101223 |
| A voice over Internet protocol (VoIP) device for providing VoIP service for a telephone includes a time detecting module and a time adjusting module. The time detecting module is operable to receive a dual tone multiple frequency (DTMF) signal, detect interrupt time of the DTMF signal, and determine whether the interrupt time is less than a predefined time interval. The time adjusting module is operable to adjust the interrupt time to the predefined time interval upon the condition that the interrupt time is less than the predefined time interval.
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| System for and method of using a loopback mechanism to perform network testing | 20100322085 | 20101223 |
| A system for and method of actively running performance tests is presented. The system and method allow a device connected, via a VPN or IPSec tunnel, to a VoIP or other service provider network, to run network performance tests using the IP address assigned by a VPN concentrator. The device utilizes the IP address assigned by the VPN concentrator as a means for performing loopback testing. These performance tests characterize the connectivity of the network. Systems and methods then use this characterization to do call admission control, traffic shaping or prioritization.
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| Emergency location information gateway for public safety answering points (psaps) and method of use | 20100317317 | 20101216 |
| The emergency location information gateway provides Public Safety Answering Points (PSAPs) with the capability to receive NG911-compliant (i.e., Session Initiation Protocol (SIP-based) and/or SIP-Voice over Internet Protocol (VoIP)) and other types emergency calls including embedded emergency location information and determine a physical location of a mobile or non-mobile target network device based on the embedded emergency location information. Conversely, a legacy network gateway (LGN) enables the conversion of traditional 911, and E911 calls into NG911 calls for transport and dissemination by so-called Emergency Services IP networks, or ESInets.
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| Distributed record server architecture for recording call sessions over a voip network | 20100316199 | 20101216 |
| Devices, systems, and methods for recording call sessions over a VoIP network using a distributed record server architecture are disclosed. An example recording device for recording segments of a call session includes a record server configured to receive an agent voice data stream and an external caller voice data stream from an agent telephone station, and a file repository configured to store voice data and call data associated with each recorded segment of the call session. The recording device is configured to tag recorded segments of each call session, which can be later used by a third-party application or database to check the status and/or integrity of the recorded call session.
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| Method and apparatus for scheduling in a wireless communication network | 20100315979 | 20101216 |
| Techniques to efficiently schedule and serve stations in a wireless network are described. An access point may aggregate stations with flows carrying traffic having similar characteristics, e.g., VoIP flows. The access point may schedule these stations together in an overall service period. The access point may serve each station in a respective service period within the overall service period. The access point may send a multi poll frame at the start of the overall service period to indicate the start time and/or service period for each station. Each station may decide to power down until its start time. The service periods for the stations may overlap one another. The service period for each station may cover an initial transmission as well as additional transmission and/or retransmission.... |
| System and method for directing targeted advertisements in ip telecommunications | 20100306061 | 20101202 |
| A Voice over Internet Protocol (VoIP) targeted advertisement delivery method that is operable in a processor coupled to a memory. The method includes monitoring all outgoing communications on a server in order to identify a VoIP telephony signal. Upon detection of a VoIP telephony signal indicating that a VoIP call setup is in progress, the method determines whether a caller identifier associated with the VoIP telephony signal identifies an associated caller as a candidate for a targeted advertisement delivery. If the caller is a candidate for a targeted advertisement delivery, identifying and retrieving a targeted advertisement for presentation to the caller during the call setup.
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| Interactive language education system and method | 20100304342 | 20101202 |
| An interactive system for improving conversational listening and speaking skills in a target language through interaction between personal communications devices (for example, mobile telephones using cellular or other wireless networks, telephones using PSTN lines, VOIP-enabled communications devices, smart phones, voice-enabled PDAs) and an automated system that provides oral/aural instruction and practice in vocabulary, pronunciation and grammar/syntax, engages the learner in simulated conversations, and provides personalized feedback and suggestions for further practice based on an analysis of the type and frequency of specific pronunciation and grammatical/syntactical errors.
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| Handling emergency calls using eap | 20100303064 | 20101202 |
| A user (terminal) is allowed to make an emergency voice-over-Internet Protocol (VoIP) phone call through an access network, such as a wireless local area network (WLAN) using Extensible Authentication Protocol (EAP). The emergency call can be made with or without authentication credentials and is identified by the user's terminal transmitting a Network Access Identifier (NAI) having a user part and/or realm part that indicates the emergency nature of the call, such as e911@e911.com. In response to such an NAI, the caller can be immediately granted limited authentication for the purpose of connecting to an emergency call center. Alternatively, the user (terminal) can be authenticated through networks supporting emergency calls, such as the user's home network, if the terminal indicates to the access network authentication server a... |
| Computer assisted voip communication method and system | 20100303057 | 20101202 |
| A computer assisted VoIP communication method and system. A remote server comprises a database in which is stored generic data of a plurality of displayed mobile phones. A personal computer in communication with the remote server comprises a local memory device in which is storable user-specific data of a known mobile phone, and an input device for selecting a desired mobile phone displayed in the remote website and for downloading the generic data associated with the selected phone to the memory device. Following generation of a virtual phone having a shape, key arrangement and functionality similar to those of the known mobile phone on the computer screen, a type and recipient of a session to be established are defined by virtually selecting a desired number and... |
| Method and communication terminal for providing voip | 20100290453 | 20101118 |
| The invention relates to a method for providing Voice over IP (VoIP) in a communication system with a number of terminals operating with VoIP, between which a transmission of voice data according to VoIP or a signalling is achieved, wherein the signalling is achieved using the Computer Supported Telecommunication Application (CSTA) interface standard. Telephone services can be controlled by a computer using the CSTA protocol. As for conventional application, H323 protocol and SIP are used for IP telephony processing of audio/video streams of a conversation. The invention is based on replacing H323 protocol and SIP by a CSTA protocol only when the latter is correspondingly extended.
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| Method and apparatus for providing service in a multi-ran communication system | 20100285819 | 20101111 |
| To address the need for more resource efficient methods of providing service in multi-RAN communication systems (600), various embodiments are described for informing a cross-paging RAN that a targeted remote unit (601) is not accepting a service request, for informing a serving RAN that the targeted remote unit is accepting the service request, and for informing an IMS (IP Multimedia Subsystem) network (611) that the targeted remote unit is moving from the serving RAN to the cross-paging RAN (or simply moving to another RAN without being cross-paged). Thus, embodiments such as those described herein can enable multi-RAN communication systems to more quickly release unneeded resources, to reduce the amount of unnecessary paging, and to minimize cross-paging when an IMS core network is available to deliver voice... |
| Voip anomaly traffic detection method with flow-level data | 20100284288 | 20101111 |
| The present invention relates to a method and system of detecting anomaly traffic for VoIP applications, in which a router captures an IP packet of a VoIP flow being transmitted, the captured IP packet is classified into an SIP message and a RTP traffic, header information is extracted from the classified SIP message and RTP traffic, the extracted header information is stored in a database, and whether SIP-based anomaly traffic or RTP-based anomaly traffic is generated is determined based on the extracted and stored information, and to a system therefor.
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| Voice over internet protocol (voip) access terminal | 20100284284 | 20101111 |
| An access terminal includes a processing system configured to determine whether a voice over internet protocol (VoIP) communication channel is available. The processing system is also configured, upon entry into a VoIP-enabled communication environment, to provide without manual intervention a first request for one or more VoIP network connection resources to initiate communication between the access terminal and a packet data gateway over a VoIP communication channel. A method and a processing system are also provided for initiating a VoIP service.
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| Method and system for routing calls over a packet switched computer network | 20100278173 | 20101104 |
| The present invention describes how a trusted network routing authority, such as a VoIP inter-exchange carrier or clearinghouse can provide routing and secure access control across multiple network domains with a single routing and admission request. This technology can improve network efficiency and quality of service when an Internet Protocol (IP) communication transaction, such as a Voice over IP (VoIP), must be routed across multiple devices or administrative domains. This technology defines the technique of performing multiple route look-ups at the source of the call path to determine all possible routes across intermediate domains to the final destination. The VoIP inter-exchange carrier or clearinghouse then provides routing and access permission tokens for the entire call path to the call source.
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| Separation of validation services in voip address discovery system | 20090323677 | 20091231 |
| In one embodiment, an apparatus may receive at least one call attribute of a public switched telephone network (PSTN) call initiated to a destination telephone number. The apparatus may verify a destination Voice-over-Internet-Protocol (VoIP) call agent for the destination telephone number based on demonstrated knowledge of the PSTN call. The apparatus may transmit an indication the destination VoIP call agent is verified for the destination telephone number.
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| Method and apparatus for providing voice control for accessing teleconference services | 20090323561 | 20091231 |
| A method and apparatus for providing access to teleconference services using voice recognition technology to receive information on packet networks such as Voice over Internet Protocol (VoIP) and Service over Internet Protocol (SoIP) networks are disclosed. In one embodiment, the service provider enables a caller to enter access information for accessing a conference service using at least one natural language response.
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| Method and apparatus for monitoring and the prevention of call storms in a communications network | 20090323551 | 20091231 |
| The present invention allows monitoring and the prevention of call storms in a communications network, e.g., a VoIP network. More specifically, the present invention enables a VoIP network to monitor the calling pattern, e.g., the number of calls within a period of time that occurs between the same endpoints. If the number of calls exceed a certain threshold, then these calls with the same endpoints will be terminated and an alarm notification will be sent, e.g., to a network operator.
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| Voice portal to voice portal voip transfer | 20090318111 | 20091224 |
| A method includes receiving at a voice portal a telephone call from a caller and acquiring from the caller a billing telephone number. The method also includes determining, based on the billing telephone number, that the telephone call should be transferred to a specific call center; identifying an exchange number for the specific call center; assembling a refer message based on the billing telephone number and the exchange number of the specific call center; and sending the refer message to a network device.
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| Bluetooth control for voip telephony using headset profile | 20090318081 | 20091224 |
| Systems and methods for Bluetooth Control for VoIP telephony using the Bluetooth Headset Profile are presented. A Bluetooth Headset Profile volume control command message is received and translated to a corresponding USB HID profile menu up or menu down command message. A Bluetooth Headset Profile hookswitch command message is received and translated to a USB HID profile menu item select command message.
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| Volume adjustment for multiple voice over internet protocal streams | 20090316932 | 20091224 |
| Systems and methods for managing the volume of multiple VoIP streams are disclosed. The system includes a VoIP server configured to receive an input audio stream from a first VoIP handset, create separate output audio streams from the input audio stream for transmission to second and third VoIP handsets, and to connect to a communications network. The system also includes a volume control table coupled to the VoIP server, the volume control table including records of volume adjustments made during prior conversations between the two or more VoIP handsets. The VoIP server is further configured create the separate output audio streams such that one or more of the output streams has a volume that is different than input audio stream based on the records.
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| Devices and methods for performing n-way mute for n-way voice over internet protocol (voip) calls | 20090316870 | 20091224 |
| The disclosed are methods and computing devices configured to engage in Voice Over Internet Protocol (VOIP) conference calls and configured to participate as a moderator of VOIP conference calls allow a moderator or delegate to monitor the mute status of call participants and/or control which participants are muted. In this way, a participant would not be required to necessarily adhere to conference call muting etiquette. For example, in the event that a participant's line is noisy, which may be unknown to the participant, the moderator could mute the line. Other distractions to the conference call as a whole could therefore be controlled as well. The ability to mute determine and/or control the muting of a participant device furthermore enables the moderator to engage rules of conduct... |
| Method and apparatus for minimizing clock drift in a voip communications network | 20090316712 | 20091224 |
| A method and apparatus for minimizing clock drift between un-synchronized clocks which may occur at opposing ends of a communication link established in, for example, a Voice over Internet Protocol (VoIP) communications network, especially for use with, for example, a FAX or modem terminal device. The illustrative system employs two or more clocks, wherein at least one of these clocks operates at an intentionally higher frequency than the nominal clock frequency (e.g., 8 kHz) and wherein at least one of these clocks operates at an intentionally lower frequency than the nominal clock frequency. In operation, the illustrative system alternatively chooses one of the clocks, in order to attempt to match the clock of the far-end terminal device on average. The state and/or history of the receiving... |
| Peer to peer inbound contact center | 20090316687 | 20091224 |
| A system and method for implementing a contact center on a device node connected to a data network. The system includes a peer-to-peer inbound contact center system that executes in each device node to enable peer-to-peer connections between users making interaction requests at a requesting device and a destination interaction endpoint. Device nodes may be VoIP telephones, computers having soft-phones, computers having a CTI-enabled PBX interface to implement CTI-enabled telephones as interaction endpoints.
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| Element-based method for auto-configuration of handsets by a base station in a heterogeneous mac protocol for wireless networks | 20090316670 | 20091224 |
| An element-based method to automatically configure mobile handsets by the base station using a protocol that combines contention-free and contention-based MAC protocols for use in wireless VoIP systems with multiple base stations is disclosed.
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| Deployable secure communication system | 20090313469 | 20091217 |
| A secure Voice-Over-IP (VOIP), video and data network functionality in a single, small size deployable case, for a remote user. While capable of secure communications, the disclosed system also provides communication capability (VOIP, video and/or data) in a non-secure manner if desired. Most importantly, bulk encrypted (i.e., secure) data may be routed over a public network, e.g., the Internet.
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| System and method for migrating a large scale batch of customer accounts from one voip system to another voip system | 20090310767 | 20091217 |
| A system and method in accordance with exemplary embodiments may include receiving account data associated with one or more accounts scheduled to be migrated from one or more first VoIP phone systems to a second VoIP phone system via a network, organizing the one or more accounts based on at least one customer service records and one system log, migrating the one or more accounts into one or more shell accounts provided by the second VoIP phone system, configuring each customer device associated with the one or more accounts to receive data from the second VoIP phone system, and migrating account settings data and account call history data associated with the one or more accounts from the one or more first VoIP phone systems to the... |
| Minimal gan rtp packet length via multi-level header compression | 20090310622 | 20091217 |
| The present invention involves a method for IP header compression for Real-Time Transport Protocol (RTP) packets that includes header compression for the outer IP layer that carries IPsec ESP tunnel traffic. The preferred embodiment of the invention involves a method of producing minimal VoIP packet size in a GAN user plane by defining an ESP/UDP/IP profile for ROHC; using the ESP/UDP/IP profile to compress the outer layers carrying IPsec ESP tunnel traffic in the GAN user plane; and using a ROHC RTP/UDP/IP profile between IPsec ESP tunnel endpoints for additional compression.
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| Method for service processor discrimination and precedence in a voice-over-internet-protocol (voip) network | 20090310604 | 20091217 |
| A method and apparatus for identifying and prioritizing applications and application servers in a Voice over IP network is disclosed. In a first embodiment, elements of signaling information are extracted from a call and are mapped to parameters associated with the call. These mapped parameters are then used by a service broker in a VoIP network to identify one or more application servers adapted to process the values of the respective parameter. The service broker may illustratively identify the application servers by a pointer to permit flexible reassignment of processing of a given parameter. The matched pointer/parameter combinations are then mapped to a precedence index. Then, according to this precedence index, the aforementioned pointers are mapped to specific addresses of application servers and the elements of... |
| Apparatus and method for providing mirroring service in voip system including ip-pbx | 20090310599 | 20091217 |
| An apparatus and method for automatically mirroring Real Time Protocol (RTP) packets in a Voice over Internet Protocol (VoIP) system including an Internet Protocol-Private Branch Exchange (IP-PBX). It is possible automatically detects call startup and/or termination from an RTP packet or an RTP Control Protocol (RTCP) packet provided through a mirroring port of the IP-PBX, and based on the detection of call startup and/or termination, automatically mirrors the RTP/RTCP packet, which is transmitted/received due to call establishment.
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| Voip multimode wlan, wi-fi, gsm, edge, tdma, spread spectrum, cdma systems | 20090310591 | 20091217 |
| Voice over Internet Protocol (VoIP) voice signal processed into Orthogonal Frequency Division Multiplexed (OFDM) or Orthogonal Frequency Division Multiple Access (OFDMA) signal. Processor and transmit filter for processing and filtering a signal and for providing a Time Division Multiple Access (TDMA) signal. Processing and filtering a signal and providing cross-correlated in-phase and quadrature-phase filtered baseband signal. Modulator and a transmitter for modulation and transmission of selected or combined signal. A two antenna receiver for receiving transmitted signal, a demodulator, receiver filter and receiver processor for demodulation, filtering and processing of received TDMA signal. Receive filter for filtering of the TDMA signal is mismatched to the transmit filter for filtering of the TDMA signal and receive processor for processing provides processed received baseband mis-match filtered cross-correlated in-phase... |
| Keep alive timeslots in a heterogeneous mac protocol to track handsets in a wireless network | 20090310575 | 20091217 |
| This disclosure describes a new keep alive timeslots based method to track registered handsets for a MAC protocol that combines contention-free and contention-based MAC protocols into a heterogeneous MAC protocol used for forwarding VoIP traffic in VoIP systems on wireless networks using multiple base stations thus allowing for transmission of high bit-rate data to multiple users over wired and wireless means.
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| Method and systems providing peer-to-peer direct-mode-only communications on cdma mobile devices | 20090310570 | 20091217 |
| Embodiments of apparatus, systems and methods enable peer-to-peer direct-mode-only communication within CDMA mobile devices. In an embodiment, peer-to-peer communication is provided via an ad hoc WiFi network connection between two or more mobile devices with sound encoded into data packets using VoIP technology addressed to other members of the ad hoc network. In another embodiment, a modified CDMA transceiver enables mobile devices to receive transmissions from other CDMA mobile devices with peer-to-peer communications identified by a pseudorandom number (PN) offset that differs from the PN offset assigned by the cellular network infrastructure. In another embodiment, a modified CDMA transceiver enables mobile devices to receive transmissions from other CDMA mobile devices on a frequency that is different from the two frequencies employed and CDMA cellular communications. Synchronization... |
| Provisional hand-off mechanism in a heterogeneous mac protocol for wireless networks | 20090310560 | 20091217 |
| This disclosure describes a provisional hand-off mechanism to transfer handsets between base stations for a MAC protocol that combines contention-free and contention-based MAC protocols for use in wireless VoIP systems with multiple base stations that reduces packet overhead thus allowing for transmission of high bit-rate data to multiple users over wired and wireless means.
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| Wideband telephone conference system interface | 20090310520 | 20091217 |
| A telephone conferencing system interface is provided which is operable at high bandwidths via a VoIP telephone. A teleconference system interface module is connected to a the VoIP telephone via the telephone's headset connector and to a teleconference system via conferencing input circuitry, conferencing output circuitry and interface module connection circuitry. The teleconference interface module includes ground isolation transformers selected for high bandwidth operation and includes a privacy switch which can disconnect a teleconference from the VoIP telephone.
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| Communication terminal and communication system and display method of communication terminal | 20090309897 | 20091217 |
| A communication terminal and a communication system able to adaptively update sizes and positions of screens in accordance with the situation without operation by a user, and a display method of the communication terminal, which perform processing of individual packets in a linked manner under the control of a display image control portion 306 based on an address of a transmitting side. Based on a sound pressure (volume) of audio information (VoIP) from the same transmitting side address, the terminal etc. calculate the screen sizes for mapping of packets of the video information from the same transmitting side address and corrects the top/bottom of the same screen or displays an instructed animated character based on top/bottom information, instruction information, etc. described in the packets of the... |
| Method for switching rate and bandwidth scalable audio decoding rate | 20090306992 | 20091210 |
| A method of bitrate switching on decoding an audio signal coded by a audio coding system, said decoding comprising a post-processing step depending on the bitrate. On switching from an initial bitrate to a final bitrate, said method includes a transition step of continuous change from a signal at the initial bitrate to a signal at the final bitrate, one or both of said signals being post-processed. Application to transmission of VoIP speech and/or audio signals in data packet networks.
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| Media resource adaptation method, media gateway controller and server | 20090303987 | 20091210 |
| The embodiment of the invention provides a media resource adaptation method, a media gateway controller and a server. In an embodiment of the invention, the services, the CTIS and the MGC are deployed in the center, each dispersed area is only equipped with the access equipment or the resource equipment, the deployment structure is simple, and the maintenance workload is less. The CTIS decides the resource adaptation, and the resource can be extended to the service sides. Therefore, the resource usage is flexible. The user terminal call can obtain the local media services, which reduces the occupancy of a VoIP long-distance link, thus lowering the operation cost. Furthermore, the global share of the media resources is realized, which is favorable for load balance and reduces the... |
| Method and apparatus for voice-over-ip call recording and analysis | 20090303897 | 20091210 |
| A method and computer-readable medium for obtaining information associated with a VoIP communication session includes tapping the computer network passively to obtain signaling and media information in a first format, separating the signaling and media information, determining transport information from at least one of the signaling and media information, transcoding the media information to a second format, and storing the transcoded media information in the second format. The media information includes data, voice, audio, and/or video information. A system obtain information associated with a VoIP communication session on a computer network includes a tapping device to passively tap the computer network to obtain signaling and media information in a first format, a processing device to transcode the media information from the first format to a second... |
| Congestion control system, call session control device, border gateway device, and congestion control method used therefor | 20090303875 | 20091210 |
| A congestion control system according to an exemplary aspect of the present invention includes an SIP (Session Initiation Protocol) terminal of a subscriber; a call session control device that accommodates the subscriber using SIP, and includes a notification unit that inserts server information of the call session control device including at least an acceptable number of calls in an SIP response message; and a border gateway device that includes a control device of VoIP (Voice over Internet Protocol) services, and a congestion control unit that performs congestion control for the call session control device based on the server information of the SIP response message notified by the call session control.
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| Sip based voip multiplayer network games | 20090298587 | 20091203 |
| This invention is directed to a system and method for the close coupling of Session Initiation Protocol (SIP) based Voice over Internet Protocol (VoIP) with multiplayer network games. Game state dependent voice communications are used as another method of player interaction in the game environment. The system uses SIP enhanced game servers and conference servers with audio mixing capabilities, and the game server controls the creation, maintenance and changing of the voice conferences. Participation in a voice conference is determined by the game state features associated with each player including distance and direction to determine how the voices are mixed in a voice conference.
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| Encrypted voip calls | 20090296932 | 20091203 |
| The invention regards a system and a method to encrypted calls through a voice over Internet protocol in at least one of a telecommunication and a data network. Every call in the network (10) is encrypted in a voice over Internet protocol server (26) with a key from a key generator (28) comprising an encryption algorithm.
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| Sharing solutions for a universal linking gateway between telephonic devices | 20090296918 | 20091203 |
| A universal linking gateway interfaces between a local network of one or more local phones, such as analog phones or PBX phones, a plurality of cell phones and multiple service lines, including a public switched telephone network. Implementation of industry standard protocols such as hook flash signals enable a simplistic local phone to access enhanced calling features including answering calls on multiple service lines, to select and make calls on any desired service line as well as handle simultaneous calls on either an active service line or multiple service lines. Further, the gateway manages security and data transfer features between the gateway and the telephonic devices connected thereto. The gateway interfaces between a variety of other telephonic devices including PSTN, VoIP, and telephonic intercoms. Wireless communications... |
| Voip monitoring system and voip monitoring method | 20090296902 | 20091203 |
| A Voice over Internet Protocol (VoIP) system includes two or more VoIP terminals, a monitoring server (30), and a soft switch device (10). The soft switch is capable of determining whether or not phone number associated with at least one of the two or more VoIP terminals is predetermined to be monitored, amending information associated with communication between the two or more VoIP terminals, and establishing the communication between the two or more VoIP terminals through the monitoring server. The monitoring server is capable of reading voice data packets sent between the two or more VoIP terminals.
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| Methods and systems for managing a call session | 20090296900 | 20091203 |
| Methods and systems are provided for managing call sessions on public and private networks. The methods and systems operate to receive and send voice over internet protocol (VoIP) communications using a network, such as an IP network. The methods and systems also operate to receive and send emergency information over IP and other data networks. Based on certain criteria, the methods and systems determine whether to transfer a VoIP communication and/or emergency information to another entity associated with the IP network.
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| System and method for providing virtual multiple lines in a communications system | 20080316946 | 20081225 |
| An exemplary system and method for providing virtual multiple lines in a communication system is provided. The exemplary system allows a plurality of simultaneous phone calls to be conducted via a single access identifier (e.g., phone number). In exemplary embodiments, the system comprises a hub configured to route calls via PSTN and VoIP. The determination as to how the call should be routed may be performed by the hub, a call services server, or a combination of both. At least one client device may be coupled in communication with the hub and configured to enable placement or receipt of a second or subsequent call when a first call is active. The client device is coupled to a communication device which places or receives the second or... |
| System for deploying voice over internet protocol services | 20080316999 | 20081225 |
| A system for deploying Voice over Internet Protocol (VoIP) services is provided. A system that incorporates teachings of the present disclosure may include, for example, a Call Session Control Function (CSCF) having a controller element to receive a Session Initiation Protocol (SIP)message from an originating communication device requesting communications with a terminating communication device, and establish an Internet Protocol (IP) connection between the originating communication device and an advertisement media system to present at the originating communication device an advertisement message that replaces a ringback tone associated with the terminating communication device. Additional embodiments are disclosed.
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| Methods and apparatus to provide a call-associated content service | 20080317000 | 20081225 |
| Methods and apparatus to provide a call-associated content service to voice over Internet protocol (VoIP) devices are disclosed. An example method comprises receiving a message comprising a uniform resource identifier (URI) and a call dialog parameter at a content mediator, the call dialog parameter associated with a first communication session between a voice over Internet protocol (VoIP) endpoint and a destination, establishing a second communication session from the mediator to the destination based on the URI and the call dialog parameter, receiving content associated with the first communication session via the second communication session, and providing the content to the VoIP endpoint.
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| Sip endpoint configuration in voip networks | 20080317004 | 20081225 |
| VoIP networks and methods are disclosed for configuring SIP endpoints of VoIP networks. An application server of a VoIP network identifies an endpoint configuration for the SIP endpoints, and generates a configuration command based on the endpoint configuration. The application server formats a SIP message to include the configuration command, and transmits the SIP message to the SIP endpoints. Responsive to receiving the SIP message, the SIP endpoints process the SIP message to identify the configuration command, and set local configuration parameters based on the configuration command.
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| Voice-over-ip hybrid digital loop carrier | 20080317008 | 20081225 |
| 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.
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| Voip multiline failover | 20080317011 | 20081225 |
| Tools and techniques are provided to provide Voice-over-IP (VoIP) communications. On receiving a user request to initiate a telephonic connection from a local site, at which the user is using a VoIP telecommunication device which has no link failover functionality, a system establishes over wide area network links at least two tunnels between a controller at the local site and another controller. Then the system transmits VoIP traffic over at least one of the tunnels from the controller at the local site to the other controller. The system also detects failure of the primary tunnel during the call and performs failover by changing a data path during the call to transmit packets over at least one non-failed tunnel.
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| Method and apparatus for registering multiple phone numbers associated with a frequently called party | 20080317227 | 20081225 |
| The present invention enables the calling party to register multiple addresses of frequently called parties with a packet-switched network service provider, e.g., a VoIP network service provider, and have the network dials through the numbers on behalf of the calling party until the called party is reached or the list of available phone numbers has been exhausted. Optionally, the network can intersperse each call by asking the calling party if he or she wants to dial a particular phone number of the called party from the list of available phone numbers, and the network can dial the selected phone number.
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| Distributed media resources in voip networks for providing services | 20080317238 | 20081225 |
| VoIP networks and methods are disclosed for distributing media resources in a VoIP network for providing services. VoIP endpoints, media gateways, and/or other network nodes in the VoIP network report available media resources (i.e., available processing capacity) that are able to provide services, such as providing announcements, DTMF digit collection, etc, to a distributed resource server. The distributed resource server then assigns the available media resources among the VoIP endpoints and/or media gateways to perform services for calls in the VoIP network in order to distribute the tasks of performing services among different nodes of the VoIP network.
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| Social network based recording | 20080317439 | 20081225 |
| The disclosure relates to an enhanced user media viewing experience in a shared viewing environment. A content sharing system is provided in which one digital video recording device controls the presentation of the same video content and optionally the acquiring of that video content on disparately located digital video recording devices. Various communications devices (e.g., VOIP devices, web cameras, instant messaging, etc.) are used to facilitate interactions between viewers at the disparately located locations. User-generated commentary, whether live via the communication devices or pre-recorded, is presented while a viewer is viewing a particular piece of video content and can be synchronized to be presented at a particular time in the video.
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| Ip converged system and packet processing method therein | 20080310307 | 20081218 |
| An IP converged system includes a VoIP ALG module and a policer module. The VoIP ALG module acquires dynamically changing RTP IP/port information of a packet by parsing a VoIP SIP message, and transmits the RTP IP/port information to the policer module. The policer module sets IP/port, which provides a real-time data service, by referring to the information from the VoIP ALG module, and discriminatively sets a packet processing condition for a non-real-time data service and a packet processing condition for the real-time data service. The VoIP ALG module and the policer module share RTP IP/port information, dynamically determined by the negotiation between VoIP gateways or VoIP terminals, in call setup/release, so that the policer can discriminately drop or mark VoIP packets by referring to the... |
| System and method for semi-persistent and dynamic scheduling and discontinuous reception control | 20080310355 | 20081218 |
| Methods of combining semi-persistent resource allocation and dynamic resource allocation are provided. Packets, such as VoIP packets, are transmitted on the uplink and downlink using respective semi-persistent resources. For each mobile device, awake periods and sleep periods are defined. The semi-persistent resources are aligned with the awake periods so that most of the time the mobile device can turn off its wireless access radio during the sleep periods. In addition, signalling to request, and to allocate, resources for additional packets are transmitted during the awake periods, and the resources allocated for the additional packets are within the awake windows.
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| Method and system for packetised content streaming optimisation | 20080312922 | 20081218 |
| A method of determining the speech content of a packet carrying speech encoded data missing from speech segment communicated by in a packetised data stream communicated using at least one VOIP link between a server platform and a client platform, the method comprising at the client platform: receiving a plurality of packets carrying speech encoded data forming said packetised data stream; processing each received packet to determine a unique message segment identifier associated with a speech segment of the received packet; processing each received packet to determine if it contains another unique message segment identifier associated with a previously received packet carrying encoded speech data; determining if the unique message segment identifier for the received packet exists in storage means provided on the client platform, and... |
| Stateful and cross-protocol intrusion detection for voice over ip | 20080313737 | 20081218 |
| A method for detecting intrusions that employ messages of two or more protocols is disclosed. Such intrusions might occur in Voice over Internet Protocol (VoIP) systems, as well as in systems in which two or more protocols support some service other than VoIP. In the illustrative embodiment of the present invention, a stateful intrusion-detection system is capable of employing rules that have cross-protocol pre-conditions. The illustrative embodiment can use such rules to recognize a variety of VoIP-based intrusion attempts, such as call hijacking, BYE attacks, etc. In addition, the illustrative embodiment is capable of using such rules to recognize other kinds of intrusion attempts in which two or more protocols support a service other than VoIP. The illustrative embodiment also comprises a stateful firewall that is... |
| Apparatus and method to support voip calls for mobile subscriber stations | 20080304445 | 20081211 |
| In some embodiments, a base station includes a service flow management module having an admission control module and a data path function module in communication with the admission control module. The data path function module is adapted to generate a first dynamic service addition (DSA) request message for a first uplink service flow in an active state to provide voice over internet protocol (VoIP) signaling. the admission control module, in response to the admission control module determining that a second uplink service flow in an admitted state for a VoIP call can be supported, is adapted to generate an admit signal, with the first and the second uplink service flows being substantially in accordance with an Institute of Electrical and Electronic Engineers (IEEE) 802.16 standard. The... |
| Cellular network/wlan voip service interaction by home wireless router | 20080304460 | 20081211 |
| A home wireless router establishes a Wireless Local Area Network (WLAN). The home wireless router services a plurality of wireless terminals within the WLAN service area, the plurality of wireless terminals including at least one Voice over Internet Protocol (VoIP) wireless terminal. The home wireless router receives a request to terminate a WLAN VoIP call to the VoIP wireless terminal. The home wireless router determines that the VoIP wireless terminal is currently servicing a cellular call via a cellular network. The home wireless router causes the cellular network to be misled into concluding that another cellular call is to be serviced by the VoIP wireless terminal, the cellular network places the cellular call on hold in the incorrect conclusion that it will service the another cellular... |
| Methods and apparatus to perform call screening in a voice over internet protocol (voip) network | 20080304471 | 20081211 |
| Methods and apparatus to perform call screening in a voice over Internet protocol (VoIP) network are disclosed. An example method comprises sending a call initiation request to a VoIP endpoint, receiving a call screening indication from a user of the VoIP endpoint in response to the call initiation request and prior to a no answer determination made by an initiating call server, and playing a message for the user while a calling party is leaving the message.
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| Communication embodiments and low latency path selection in a multi-topology network | 20080304472 | 20081211 |
| In one embodiment, a source device (e.g., a VOIP phone) establishes a call connection with a remote device depending on which of multiple network paths provides an acceptable latency (e.g., a lower latency). For example, in response to receiving a request to establish a connection with a remote destination device over a network, the source device (e.g., a caller's phone) obtains multiple service code values. The source device encodes each of multiple data packets to include a unique service code value for transmission of the messages over different network topologies to a remote destination. Thus, when transmitted, each of the multiple messages follows a different logical network topology of a network as specified by a respective service code value. Based on feedback from a remote device... |
| Enhanced terminal adapter | 20080304473 | 20081211 |
| An enhanced terminal adapter (ETA) allows multiple communication devices connected to either similar or dissimilar networks, and typically accessible through different access numbers/addresses, to be used as extensions of each other when any of the devices are accessed via their access number/address. In the case of a PSTN call, communications devices connected to the ETA are considered POTS extensions. For a VoIP call, the communications devices connected to the ETA could be considered enhanced extensions in a VoIP network. In this way, the communications devices are each extensions of each other, depending on which device is accessed.
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| Enhancing subscriber location tracking mechanism for voice over internet protocol services | 20080304487 | 20081211 |
| A system for providing a Voice over Internet Protocol (VoIP) service has an Internet Protocol (IP) address detector for detecting an IP address assigned to a customer by an IP network during a VoIP communication session with the customer. An IP address database stores detected IP addresses. The IP address detector is configured for detecting a first IP address assigned to a customer during a first VoIP communication session with the customer, and for detecting a second IP address assigned to the customer during a second VoIP communication session with the customer. The second IP address is compared with the first IP address to produce an address change signal indicating a change in a physical location of the customer when the second IP address does not... |
| Automatic emergency call notification to pre-designated personal emergency contacts | 20080304630 | 20081211 |
| Automatic notification is provided to pre-designated emergency contacts for a given wireless subscriber upon the occurrence of an emergency call from the subscriber. Triggering for the automatic notification may take place at an appropriate time in the emergency call, e.g., at a time the emergency call is received, at a time during the emergency call after a current location of the emergency caller is obtained by the emergency network, once the emergency call terminates, after a set amount of time after the emergency call is received, terminates, etc. The automatic notification is preferably made using voice over Internet Protocol (VoIP), but may instead be pre-designated by the subscriber to be in an alternative mode, e.g., by SMS, switched telephone, etc. An option is included for the... |
| Splitter wall plates for digital subscriber line (dsl) communication systems and methods to use the same | 20080304656 | 20081211 |
| Splitter wall plates for digital subscriber line (DSL) communication systems and methods to use the same are disclosed. An example apparatus comprises a splitter to separate a digital subscriber line (DSL) signal from a plain old telephone signal (POTS) signal, and a switch to selectively couple a VoIP signal received via a first jack or the POTS signal to a second jack.
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| Electronic directory system | 20080306905 | 20081211 |
| A reverse based Electronic directory system is disclosed in which a subscriber can record a number of individual contact details about themselves, such as their telephone numbers (all types, including business, office direct line, business mobile, VoIP, home, personal mobile, facsimile) and e-mail addresses. The system allows other users to use one of these personal details as a search term in order to obtain other personal details. The system further provides for the ability to set permissions on any detail (such as personal details), so that they are not released without its owner's explicit permission.
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| Methods, systems, and apparatus of providing qos and scalability in the deployment of real-time traffic services in packet-based networks | 20080298237 | 20081204 |
| The paper gives emphasis on broadband wireless because of its shared access mechanism.
... |
| Voip phone number discovery on pstns using two way fxo communication | 20080298343 | 20081204 |
| A method for placing a telephone call using a Voice over Internet Protocol (VoIP), the method including using a foreign exchange office (FXO) of a first VoIP system, making a public switching telephone network (PSTN) connection with an FXO of a second VoIP system; exchanging at least an Internet telephone address between the FXOs; terminating the PSTN connection; and placing the telephone call from the first VoIP system to the second VolP system over the Internet using the Internet telephone address.
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| System and method for providing audio cues in operation of a voip service | 20080298348 | 20081204 |
| An exemplary VoIP service provides call participants cues to indicate that an enhanced service is being employed. When calling, the standard dial tone may be replaced with a distinctive dial tone or sound that indicates to the call participant that enhanced service is active (e.g., a service active sound). In some embodiments, the person called by the VoIP user hears a viral sound that indicates that an enhanced telephone service is being used. Furthermore, communication audio cues may be provided during the communication to provide further information to the call participants.
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| System for transmitting high quality speech signals on a voice over internet protocol network | 20080298349 | 20081204 |
| The VoIP quality speech process is activated when a subscriber accesses a speech quality sensitive resource or in response to an activation of the feature by the subscriber, or when it is determined that the originating subscriber terminal device requires the transmission of high quality speech signals. A transmit buffer, associated with the port circuit that serves the originating device, stores a predetermined number of packets as they are transmitted from the originating device. In the case of lost or damaged packets, the VoIP quality speech system activates the transmit buffer to retransmit the missing or damaged packet to the destination device. Intelligent buffer management is provided, where the destination device can regulate the size of the transmit buffer as well as the size of its... |
| Web-call method | 20080298561 | 20081204 |
| The VoIP server will then connect “user A's telephone number” for user B.
... |
| Synergetic tandem pocket device | 20080300013 | 20081204 |
| A videophone device including two units 30, 34 is disclosed. When decoupled from each other, the first unit is a mobile phone 34 (for example, a cellular phone or a VOIP phone) that provides voice telephony services, and the second unit 30 is a video playback device (for example, an MP4 player) that provides media playback services. When coupled to each other to form a ‘tandem device’ 32, the tandem device 32 is operative to provide video telephony services using an onboard CODEC 224 of the media playback device, a screen 250 of the media playback device, and an RF transceiver 130 of the mobile phone. Various packaging features are disclosed. A tandem device including a screenless GPS device 330 and a mobile phone is also... |
| Voip wlan conference calling via home wireless routers | 20080291886 | 20081127 |
| A home wireless router establishes a Wireless Local Area Network (WLAN) and services a plurality of wireless terminals within the WLAN service area including at least one Voice over Internet Protocol (VoIP) wireless terminal. The home wireless router receives a request from the VoIP wireless terminal to establish a VoIP conference call with at least two other VoIP wireless terminals. The home wireless router interacts with the VoIP service accumulator to determine at least one other home wireless router required to service the VoIP conference call, queries the at least one other home wireless router to determine its ability to service the VoIP conference call for at least one serviced VoIP wireless terminal, and receives a response from the at least one other home wireless router.... |
| Integrated access device, voice over internet protocol system and backup method thereof | 20080291895 | 20081127 |
| An integrated access device, a voice over Internet protocol (VOIP) system and a backup method thereof. The VOIP system includes a first remote server, a second remote server and an integrated access device. The integrated access device includes a first Internet protocol interface, a second Internet protocol interface, a memory unit and a processing unit. A first network connection is built between the first Internet protocol interface and the first remote server, and a second network connection is built between the second Internet protocol interface and the second remote server. The memory unit is for storing a program, and the processing unit executes the program to judge whether a voice packet in the integrated access device can be transmitted through the first network connection or not.... |
| Voip communication control method and access point apparatus | 20070297368 | 20071227 |
| A critical threshold value of the number of connections of VoIP communication by a wireless LAN is set to an access point apparatus. The access point apparatus records identification information of terminals each of which has transmitted a request for connection of VoIP communication. After the number of terminals whose identification information have been recorded reaches the critical threshold value, the access point apparatus transmits a packet indicating release an association to a terminal whose identification information is newly recorded.
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| Method and apparatus for identifying and monitoring voip media plane security keys for service provider lawful intercept use | 20070297418 | 20071227 |
| A mechanism is described that enables encrypted end-point communications in a VoIP network to be accessed by a service provider. The mechanism includes a session information retrieval component which gathers session information such as encryption keys for each session that traverses a network element. The encryption keys may be used to decrypt data to make it available for lawful interception. A media stream monitoring component monitors media streams and verifies that the identified keys for each session are valid, to ensure continuity in compliance with LI regulations. Advantageously a security alert component may be used to controls further session operation for those sessions identified as potential security risks. With such an arrangement, the service provider can satisfy the legal requirement to provide interception, verify that the... |
| Public services access point (psap) designation of preferred emergency call routing method via internet or public switched telephone network (pstn) | 20070298765 | 20071227 |
| The use of the VoIP emergency network for routing wireless E911 calls to a designated PSAP. In this embodiment, a mobile positioning center (MPC) assigns an ESRK per existing prior art, but uses the invention to route the call to the PSAP via the VoIP server and an ESGW. This relieves wireless carriers of the obligation to install and maintain expensive dedicated SS7 or CAMA trunks from each MSC to each selective router in the areas served by that MSC. Instead, wireless 911 calls can be consolidated by ESGW vendors, maximizing the efficiency of the dedicated trunks to the selective router by sharing those trunks with multiple MSCs.
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| Providing content delivery during a call hold condition | 20070299939 | 20071227 |
| An approach for providing content transmission upon placement of a call on hold is disclosed. A data communications system includes a proxy server that is configured to receive a message from a first client indicating the hold condition of a Voice Over Internet Protocol (VOIP) call with a second client. The system also includes a content server (e.g., music server) that is configured to transmit the content stored therein to the second client in response to a request message from the server.
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| Voip voice quality remote for switch-to-switch connections | 20070291648 | 20071220 |
| A method to reroute a call with a first and second termination point based upon Quality of Service (QoS) is disclosed. The method comprises establishing a quality of service threshold, connecting a first communication path between the two termination points through the network elements, testing said quality of service along said first communication path and generating a quantitative value based at least in part on said quality of service. The method continues with establishing a second communication path between the network elements if the quantitative value is not up to the level of said quantitative quality of service threshold; and rerouting said data transmission through the newly established second communication path between the network elements while maintaining said termination points.
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| Waveform quality feedback for internet protocol traffic | 20070291655 | 20071220 |
| Techniques are disclosed for selecting transmission resources in a telecommunications system that comprises different networks, where the techniques are based on the quality of the waveform of transmitted media such as audio or video signals, in contrast to the quality of service of the network that transports the media. The problem with only using quality of service to determine which resources to allocate to a call is that quality of service does not guarantee that the quality experienced by the call's participants is satisfactory just because a component network's quality of service is satisfactory. For example, the end-to-end delay experienced in a VoIP network might be satisfactory for most data transfers, but might still be inadequate to control the echo experienced by telecommunications users. The disclosed... |
| Payment system and its method for supporting user verification in voip configuration | 20070291741 | 20071220 |
| Disclosed is a payment system and method for supporting user verification in VoIP configuration by receiving secret number or VoIP phone number of a user as payment information on Internet and completing the payment by user verification and occupancy verification. The payment system includes a user terminal for inputting secret number or VoIP phone number as payment information, a payment gateway server for receiving the payment information from the user terminal to intermediate user verification and approving payment by occupancy verification, and a central communication company server for receiving the payment information from the payment gateway server to verify the user and receiving a final payment approval to add a payment amount to communication charges of the user terminal and bill the charges. The payment system... |
| Method for compensating delays | 20070291744 | 20071220 |
| The present invention relates to a method and a user equipment arranged to communicate with at least a second user equipment in a VoIP service data transmission in a wireless communication system using a VoIP service, provided by an application server. The method comprises the steps of: receiving transmissions in form of a media stream from the at least one second user equipment; storing the data of the media stream; detecting whether an interruption of said transmissions having at least a minimum length has occurred during the VoIP service data transmission or expecting that an interruption of said transmissions having at least a minimum length will occur during the VoIP service data transmission, and if such an interruption is detected or expected, using a non-normative playout... |
| Waveform quality feedback for the selection of gateways | 20070291745 | 20071220 |
| Techniques are disclosed for selecting transmission resources in a telecommunications system that comprises different networks, where the techniques are based on the quality of the waveform of transmitted media such as audio or video signals, in contrast to the quality of service of the network that transports the media. The problem with only using quality of service to determine which resources to allocate to a call is that quality of service does not guarantee that the quality experienced by the call's participants is satisfactory just because a component network's quality of service is satisfactory. For example, the end-to-end delay experienced in a VoIP network might be satisfactory for most data transfers, but might still be inadequate to control the echo experienced by telecommunications users. The disclosed... |
| Voip network system for multicasting packets and a method of constructing the same | 20070291769 | 20071220 |
| In a VoIP network system, when a host terminal detects a link with a network, a connection-time determiner determines the time required for reconnection from a disconnection time when a link with the network was disconnected before to an establishment time when a link with the network is reestablished. If the time required for connection is shorter than a predetermined period of time, the connection-time determiner determines that address information is not to be transmitted since the time required for connection is too short to transmit the address information. Otherwise, the connection-time determiner determines that the address information is to be transmitted. According to a result of the determination, self-address information is transmitted by multicast communication to other terminals.
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| Method of packet data transmission | 20070291797 | 20071220 |
| A method of transmitting data packets using a staggered transmission technique by which some transmissions of data packets may be delayed to reduce periodicity inherently caused by synchronous retransmission schemes. The amount of delay should be minimal in order to not adversely effect applications with tight latency constraints, such as Voice over Internet Protocol (VoIP). The delay may be applied to initial transmissions or retransmissions of data packets.
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| Voice over internet protocol network test device and method | 20070291907 | 20071220 |
| The invention includes a device and a method for testing a communication network that includes an Internet Protocol (VoIP) network. The inventive device includes a user interface for communicating results of a test, a transceiver for allowing the test device to communicate with a telecommunications network, and a digital signal processor for reformatting a communication signal to be tested by the device. The digital signal processor may include at least one coder/decoder that uses various compression protocols including wherein the digital signal processor uses at least one of the following data compression techniques: G.711a-law, G711μ-law, G.720, G.723.1, G.726, G.728, G.729, G.729A, and G.729AB2. The transceiver also may be a power line modem. The inventive device may include a processor, random access memory, read only memory, a... |
| Method and apparatus for the purchase, sale and facilitation of voice over internet protocol (voip) consultations | 20070291923 | 20071220 |
| The present invention relates to a novel method and apparatus for facilitating voice-to-voice communication in an on-line environment between requestors seeking advice and helpers providing advice. The invention comprises a user-friendly website interface in which a plurality of helpers and requesters can provide personal data relating to their qualifications or information about the subject matter they wish to discuss through the posting of text or pictorial information. The information can be organized and categorized in a variety of ways such as, for example, to be accessible through a search engine. Visitors can review the information provided; identify a helper or requestor with whom they wish to establish communication; purchase or sell these services; and initiate VoIP voice-to-voice communication directly with an individual through the website.
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| System, method and handset for sharing a call in a voip system | 20070293220 | 20071220 |
| An embodiment generally relates a method of joining a call. The method includes establishing the call between an internal mobile terminal (MT), an external MT, and a network access point (NAP). The call comprises a connection between the internal MT and the NAP and a second connection between the NAP and the external MT. The method also includes sensing the call by a second internal MT and joining the call from the second internal MT by depressing a send key without entering a number on the second internal MT.
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| Computerized telephony accessibility provider (ctap) for provisioning a talking telephone device | 20070286098 | 20071213 |
| A computerized telephony accessibility provider (CTAP) for provisioning a talking telephone device coupled to a personal computer, laptop, mobile telephone, or other computing device. The CTAP functions to audiblize the results of the operation of a telephone feature, navigation and interaction with a telephone device and featured incoming telephony signals (Caller ID and Message Waiting) from an external telephony source. The CTAP function in combination with a telephone device or as a standalone VoIP telephone device.
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| Method and apparatus for processing session initiation protocol messages associated with a voice over ip terminal | 20070286160 | 20071213 |
| An apparatus and method are disclosed for processing session initiation protocol (SIP) messages associated with a Voice over IP (VoIP) terminal. An apparatus that incorporates teachings of the present disclosure may include, for example, a network proxy having a controller that manages a communications interface in a communication system. The controller can be programmed to receive from a VoIP terminal a SIP message comprising a request to communicate with a destination point, detect from the SIP message a lack of location information associated with the VoIP terminal, and transmit a location query to one among the VoIP terminal, and a gateway connecting the VoIP terminal to the network proxy. Additional embodiments are disclosed.
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| Method and apparatus for invoking multimodal interaction in a voip call | 20070286162 | 20071213 |
| Method and apparatus for invoking multimodal interaction in a VOIP call is described. In one example, handling a call at a first client device in a packet network is described. The first client device obtains device capabilities of a second client device in response to the call. The device capabilities are processed to determine whether the second client device is capable of graphical display or graphical interaction. If so, the first client device follows a multimodal call flow. Otherwise, the first client device follows a voice-only call flow.
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| Sip server | 20070286163 | 20071213 |
| There is provided a SIP server realizing a system capable of connecting a terminal in a local communication network to a carrier communication network without preparing a customer information database based on the communication company system specification or an IP call control device, thereby simplifying the system configuration and reducing the cost. The SIP server (30) controls connection between SIP terminals by the SIP extended network (10) built on the Internet network. A SIP mutual connection unit (33) containing one or more carrier identification numbers or representative numbers acquired from each carrier is provided in the extended network SIP server (30). The carrier identification number or the representative number contained in the SIP mutual connection unit (33) is used for reception from a carrier VoIP network... |
| Method for providing voip services for wireless terminals | 20070286165 | 20071213 |
| The present invention relates to a system and method for wireless telecommunication in a packet-based network comprising a Software Radio Port (SRP) which functions as a radio base station and a VoIP gateway to interconnect the wireless network with the VoIP packet network. Together with a Network Server Platform (NSP) and VoIP call-server, the SRP combines mobile call processing signaling with the VoIP call signaling to establish calls between the mobile and VoIP device or between mobiles. The SRP establishes the voice path to the mobile station over the air and the RTP media path to a party over a packet network for a call. These two paths are interconnected at the SRP so that an end-to-end voice path is established.
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| Method and apparatus for processing a communication request from a roaming voice over ip terminal | 20070286170 | 20071213 |
| An apparatus and method are disclosed for processing a communication request from a roaming VoIP terminal. An apparatus that incorporates teachings of the present disclosure may include, for example, a network proxy having a controller that manages a communications interface in a communication system. The controller can be programmed to receive from a Voice over IP (VoIP) terminal a request to communicate with a destination point, wherein the request comprises an identification (ID) of the destination point, an ID of the VoIP terminal, and a media access control (MAC) address of a gateway providing the VoIP terminal communication access to the network proxy, and determine from the MAC address of the gateway and the ID of the VoIP terminal whether the VoIP terminal is roaming outside... |
| Multi-mode voice instant messaging internet phone (vimip) handset device | 20070286171 | 20071213 |
| A multi-mode Voice Instant Messaging Internet Phone (VIMIP) handset device that can effectuate a VoIP phone call to a standard phone or other IM client device (a personal computer with voice IM client software or another VIMIP handset device) with a telephone number associated therewith through an Instant Messaging system which supports VoIP phone call with a telephone number. The VIMIP handset device can also effectuate a VoIP conversation with other IM client device with voice capability using a buddy list through an Instant Messaging system with voice communication (voice chat) capability. In a first mode, the VIMIP handset device communicates in a wireless computer network connection mode and in second mode, in a wired computer network connection mode.
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| Xdsl voip adapter device | 20070286172 | 20071213 |
| Described is an apparatus for permitting installation of voice over internet protocol (VoIP) service at a subscriber premises. A VoIP adapter is provided that is configured to enable application of sealing current to an outside line while permitting use of analog equipment on a VoIP enable premises wiring system. The VoIP adapter includes a high-pass filter portion that permits the use of existing premises analog equipment.
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| System and apparatus for incorporating voice over internet protocol (voip) adapter functionnality into a cordless telephone system | 20070286174 | 20071213 |
| A system (50) including a telephone device having incorporated therein voice over Internet protocol (VoIP) adapter functionality. The system includes a VoIP adapter for coupling to a network, such as an Internet protocol (IP) network, and a telephone device, such as a cordless telephone device, for coupling to the VoIP adapter. The telephone device includes a base station and a plurality of handsets, such as cordless handsets. The telephone device is configured to include therein VoIP adapter functionality, such as subscriber integrated circuit (SLIC) functionality that conventionally resides in a conventional VoIP adapter. The incorporation of SLIC functionality and other VoIP adapter functionality into the telephone device reduces the overall cost of the system, e.g., by eliminating relatively expensive hardware, such as SLICs, and also reduces... |
| System and method for responsive loss compensation in a voice over internet protocol communication environment | 20070286179 | 20071213 |
| A system and method for implementing dynamic end-to-end loss compensation in a VoIP communication system is provided. The invention utilizes standard signaling protocol to accommodate for the characteristics of various call endpoints, and in particular, provides for an SDP parameter that conveys terminal characteristics between endpoints of a VoIP connection.
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| Speech-based optimization of digital hearing devices | 20070286350 | 20071213 |
| A method of tuning a digital hearing device, such as a cellular phone, a mobile phone for communicating via a wireless ad hoc network, and a Voice-over-Internet-Protocol (VoIP) phone, can include playing portions of test audio, wherein each portion of test audio represents one or more distinctive features of speech. The method also can include receiving user responses to played portions of test audio heard through the digital hearing device and comparing the user responses with the portions of test audio. An operational parameter of the digital hearing device can be adjusted based on one or more of the distinctive features of speech that is associated with the operational parameter.
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| Calling party controlled ringback tones | 20070286402 | 20071213 |
| A computer readable medium stores a program for a calling party controlled ringback tone service. The service enables a calling party to control ringback tones, even overriding called party selected ringback tones. The service can operate in a wireless or wireline network, such as a voice over IP (VoIP) network or time division multiplexed (TDM) network. The calling party can be periodically billed for the ringback tone control service, as well as the ringback tones. The subscriber can configure the service via the web.
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| Voip adapter for network interface device | 20070286405 | 20071213 |
| An apparatus and methodology for permitting subscriber installation of voice over internet protocol (VoIP) service at a subscriber premises. A VoIP adapter is provided that is configured for snap-in mounting in a subscriber premises network interface device (NID). The adapter includes an input plug connector and an output jack connector, as well as VoIP electronics, and is configured such that the VoIP electronics may be easily coupled in the subscriber incoming service line. Methods for subscriber installation of the adapter are also disclosed.
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| Qos support for voip and streaming services | 20070287467 | 20071213 |
| In one embodiment, techniques provide quality of service (QoS) for services using a gateway. The techniques include receiving one or more control signaling messages from a subscriber station at the gateway. The gateway is in a signaling path for the subscriber station where the subscriber station communicates through a wireless network. The gateway determines a policy to apply for the service based on the one or more control signaling messages. QoS parameters may be determined for the service based on the policy. For example, QoS parameters are determined for a session that is created for a media stream. The media stream may flow from the subscriber station to a base station to the gateway, and vice versa. The gateway may communicate the QoS parameters to the... |
| Fax transmission over the packet network | 20070279699 | 20071206 |
| A FoIP server mediating fax communication between gateways and enterprise backend systems. The gateways are VoIP capable, connected to the PSTN and the enterprise network, and mediate between the PSTN fax call and fax over IP calls. The FoIP Server mediates between the FoIP calls and enterprise backend systems, such as message transfer agent (MTA), Customer Relationship Management CRM, document management, web services, mainframe, or a specialized application requiring faxing capabilities.
... |
| Voice over internet protocol system and related wireless local area network device | 20070280189 | 20071206 |
| A voice over Internet protocol (VoIP) system. The system includes a first device and a second device. The first device is disposed in a wireless local area network (WLAN) and connected to an Internet, for performing voice communication with a remote device through the Internet. Additionally, the first device processes audio information received during the voice communication to generate at least a playback audio packet, sends out the playback audio packet through the WLAN, receives at least a captured audio packet through the WLAN, and processes the captured audio packet to generate audio information to be sent out during the voice communication. The second device is disposed in the WLAN, for receiving the playback audio packet through the WLAN, playing audio information contained in the playback... |
| Voice over internet protocol (voip) ready computer system and method | 20060291442 | 20061228 |
| A VoIP ready computer system includes a housing having an integrated audio communications device holder adapted to receive an audio communications device. The housing includes a portion adapted to contain processing circuitry operable to execute programs and a storage device coupled to the processing circuitry. The storage device stores a VoIP program and other general purpose programs for execution by the processing circuitry. An audio communications device is removably positioned in the audio communications device holder and is coupled to the processing circuitry. The audio communications device is operable to communicate audio signals during execution of the VoIP program.
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| Methods and apparatus for forwarding ip calls through a proxy interface | 20060291450 | 20061228 |
| A session controller configured to be coupled to a communications network is disclosed. The session controller is configured to receive an indication of a Voice over Internet Protocol (VoIP) registration request at a first time from an endpoint having a third-party registration, and to receive a signaling message from the endpoint at a second time later than the first time. The session controller is further configured to forward an indication of the VoIP registration request and a signaling message to an application server, such that at least a portion of the topology of the communications network beyond the session controller is hidden to the endpoint.
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| Mobile telephone gateway apparatus, communication system, and gateway operating system | 20060291483 | 20061228 |
| A mobile telephone gateway and communication routing device (MPG) (121) disclosed by the present invention is coupled through an electrical signaling transmission medium with a mobile telephone (100) operative on a first communication network in accordance with a first communication protocol, for adding communication capabilities through at least a second communication network in accordance with a second communication protocol. A mobile communication system (150) is disclosed, comprising MPG (121) placed in-between mobile phone (100) and battery (118) and Subscriber Identification Module (SIM) card (117). MPG connects to the SIM interface of mobile phone (100), using which it wraps SIM functionalities and controls call management. It connects to the data/signaling interface of mobile phone (100), using which it communicates with it, while communicating with the second network... |
| Methods and apparatus for improved 911 support for voip service | 20060293024 | 20061228 |
| 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... |
| Portable voip wireless connector | 20060294262 | 20061228 |
| A portable VOIP wireless connector comprising: a USB connecting interface used as an interface for connecting a computer; a multimedia chip for controlling sound effect to transmit to a wireless module; a USB controller to control the ROM and the flash memory; the wireless module used to convert a sound effect into radio frequency signals for transmitting; the flash memory to store variable data; and the ROM to store invariable data (such as VOIP software). The wireless connector is inserted into a USB connecting port of the computer, the computer automatically executes the VOIP software in the ROM and inputs related parameters for setting up, and sends radio frequency signals via the wireless module to the wireless devices, so that a user can proceed to voice... |
| Voip music conferencing system | 20060283310 | 20061221 |
| A client device comprises a plurality of audio input ports, at least one audio output port, a MIDI input port, a MIDI output port, a video input port, a video output port, a processor and a network interface. The processor generates outgoing IP messages that carry at least one audio input signal received from the audio input ports, a first MIDI signal received from the MIDI input port and a first video signal received from the video input port. From incoming IP messages, the processor extracts a second MIDI signal for the MIDI output port, a second video signal for the video output port, and at least one audio output signal for the at least one audio output port. A network interface sends the outgoing IP... |
| Mobile voip phone communication system and method | 20060285492 | 20061221 |
| A mobile Voice-over-Internet-Protocol (VoIP) phone communication system and method allows users to make mobile communication via mobile VoIP phones. A local mobile VoIP server is established in at least one specific communication zone and at least a central mobile VoIP server for coordinating the local mobile VoIP server in each zone is established. The mobile VoIP phone logins to the local mobile VoIP server in a specific communication zone to which the mobile VoIP phone enters, so the local mobile VoIP server identifies the mobile VoIP server in the zone and stores identification information of the identified mobile VoIP phone. If the mobile VoIP phone moves from a first specific communication zone to a second specific communication zone, the identification information of the mobile VoIP phone... |
| Digital telecommunications call management and monitoring system | 20060285650 | 20061221 |
| The present invention discloses a centralized, digital, computer-based telephone call management system for authenticating users of a telephone system in an institutional facility. The system includes the capacity to allow an institution to control, record, monitor, and bill and report usage and access to a telephone network. The telephone call management system further includes both accounting and management software for use in controlling, monitoring, billing, recording, and reporting usage and access. Also, it can operate over both a Public Switch Telephone Network (PSTN) and a Voice over Internet Protocol (VOIP) infrastructure.
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| Multi-mode handset services | 20060286984 | 20061221 |
| Delivery of dual-mode GSM/WiFi voice and data services to a consumer with support for transparent network registrations and handovers between GSM and WiFi networks. Aspects include transparently and wirelessly automating call redirection procedures of calls and data (e.g., SMS messages) to legacy telephone devices, to GSM handsets via broadband, and to VoIP devices via broadband. An indoor network is treated as a pseudo visitor PLMN (VPLMN) by the home PLMN (HPLMN) which does not require GSM radio coverage in an indoor network. This innovation provides a means for automating a process of redirecting a subscriber's calls from the GSM network to the subscriber's home network, and can utilize a Bluetooth access point to provide a beacon signal for triggering redirection.
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| System and method for information provider communication with information requester | 20060287929 | 20061221 |
| The present invention provides a system and a method for an information provider communicating with an information requester. The system comprises a messenger server for managing instant messaging, identifying whether the information provider registers a timing available for communication, priority sequence of communication channel and communication means, searching communication means available for communication promptly connecting the communication means of the information provider from a higher priority to a lower priority sequentially; a proxy server, coupled to the messenger server, receiving a signal from the messenger server and deciding a communication channel to process this signal and transmitting a call; a redirect server administrating the position of the information requester and the information provider logged in the messenger server, based on Internet Protocol; and a VOIP gateway... |
| Proactive congestion control scheme for voip traffic on ip routers | 20060280162 | 20061214 |
| VoIP traffic bandwidth through a router port is controlled so that it does not exceed a predetermined bandwidth to ensure a desired quality of service for all VoIP calls routed through the port. Information of a packet received by the port is extracted from a packet header. The information includes source and destination addresses and port numbers as well as the bandwidth used by the packet. If the packet is from an existing call, the packet is routed and the time the packet was received is updated. If the packet is from a new call, it is determined whether the port has sufficient bandwidth to route the new call. If the port has sufficient bandwidth, the packet is routed, bandwidth is allotted for subsequent packets from... |
| Enhanced e911 location information using voice over internet protocol (voip) | 20060280164 | 20061214 |
| An E-9-1-1 voice-over-IP (VoIP) solution is provided wherein a 911 call from a wireless VoIP device is routed directly to the correct Public Safety Answer Point (PSAP) via dedicated trunks, together with correct location information and call-back number. VoIP gateways are implemented locally, at least one per LATA, and accept VoIP packetized data inbound, and convert it to standard wireline voice calls. Calls are routed to an IP address at the VoIP gateway, which then egresses the call to a voice port at a selective router. Dedicated voice trunks (CAMA, SS7, FG-D) are installed between each local VoIP gateway and appropriate selective routers. An Automatic Location Identification (ALI) database is provisioned with ESRKs dedicated for VoIP use. TCP/IP circuits may be established between some or all... |
| Systems and methods for supporting e911 emergency services in a data communications network | 20060281437 | 20061214 |
| The present invention provides systems and methods for supporting enhanced 911 (E911) emergency services in data communications networks that includes Voice over Internet Protocol (VoIP) telephones, by facilitating the identification of the geographic location of a VoIP telephone. The invention generally provides GPS-based geographic location information to the E911 emergency services network. In one embodiment, an E911-enablement system including an E911-enablement device coupled to a GPS receiver is provided. The E911-enablement device then inserts geographic location information obtained from the GPS receiver into 911 call packet streams to provide accurate location information to the E911 emergency services network.
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| Authentication of mobile stations | 20060281457 | 20061214 |
| A method and system for authenticating a mobile station, such as a dual mode cellular telephone, for access to a wireless local area network is disclosed. The mobile station can be authenticated by registering, and thus pre-authenticating, it with the wireless area network. Registration can comprise providing identifying information regarding the mobile station to the local area network. Access to the local area network facilitates the use of the local area network for communications by the mobile station, such as using VoIP over the Internet.
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| Characterization of voice over internet protocol (voip) network | 20060274664 | 20061207 |
| In one embodiment, a method for characterizing a VoIP network commences with the sending of a plurality of predefined messages from a calling end of the VoIP network. One of the messages is a call initiation message that is sent to a called end of the network. In response to sending the messages, the calling end of the VoIP network receives a plurality of predefined response messages. A plurality of delays are then calculated, with each delay being a function of at least two timing measurements selected from 1) times at which the messages are sent, and 2) times at which the response messages are received. The VoIP network is then characterized by comparing the delays to programmable reference values. Other embodiments are also disclosed.
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| Dynamic e911 updating in a voip telephony system | 20060274725 | 20061207 |
| A method in one example comprises the steps of establishing a location database for a first network that associates Internet Protocol address information for VOIP telephony devices with physical locations for the VOIP telephony devices; upon detecting registration of a VOIP telephony device through a network port of the first network, determining physical location of the VOIP telephony device by identifying physical location of the network port through which the VOIP telephony device is connected, and updating the location database; and, upon detecting IP telephony activity originating from a remote VOIP telephony device connected to the first network through a VPN gateway, determining physical location of the remote VOIP telephony device connected through the VPN gateway by identifying physical locations of network resources of a second... |
| Apparatus and method for connecting a voice over ip telephone subscriber to the 911 emergency network | 20060274729 | 20061207 |
| It is accordingly an object of the invention to provide a device to connect a VoIP telephone subscriber to connect a communication network, in order to place a telephone call, which may be an emergency call. The device includes determination circuitry to determine whether the telephone call is to be placed over a high speed IP data network, or via another communications network. Upon making a determination, the call is placed over the determined network.
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| Screening inbound calls in a packet-based communications network | 20060274733 | 20061207 |
| A method and system is provided for performing inbound call screening in a packet-based network, such as an H.323 Voice over IP (VoIP) network. The inbound gateways on the network are registered with inbound gatekeepers, and standard messages are used between an inbound gateway, an inbound gatekeeper and an inbound screening database to decide: whether an inbound call to a particular called number (DID) is to be allowed into the network; whether the called number should be translated into a different called number; and whether a routing index should be included in the called number to indicate the destination of the call.
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| Communicating apparatus, control method of communicating apparatus, and control program of communicating apparatus | 20060274892 | 20061207 |
| In a communicating apparatus which makes speech and image communication, a proper communication path is selected by a simple construction of a low cost, thereby enabling data communication of a high-speed and high reliability to be performed. If a telephone number of an opponent station corresponds to an VoIP network, a communicating apparatus obtains an IP address of the opponent station from an SIP proxy server and sends and receives communication data on an IP network to/from the opponent station by a file sending/receiving protocol such as FTP, HTTP, or the like. If the telephone number of an opponent station does not correspond to the VoIP, image data is facsimile-modulated, a digital encoding method (64 kbps PCM encoding) suitable for a facsimile modulating method is selected,... |
| Scheduling radio resources for symmetric service data connections | 20060268813 | 20061130 |
| Radio channel resources are allocated in a radio communications system based on whether a requested data communication requested is associated with a symmetric service or an asymmetric service. If the data communication is associated with a symmetric service, a symmetric connection between the mobile radio station and the radio network is established for communicating data packets associated with the data communication. Radio channel resources are assigned to the symmetric connection for transporting the data packets between the mobile radio station and the radio network. The symmetric connection is assigned substantially the same amount of radio channel resources in an uplink direction as in a downlink direction. In one example, the symmetric data communication is a voice over IP (VoIP) communication. One example environment is a GPRS/EDGE... |
| Communication path setting method and communication path setting system | 20060268826 | 20061130 |
| In accordance with the communication path setting system of the present invention, when it is determined that an ADSL modem 22 connected to a VoIP network 1 is not operating, the MGC 62 sets a communication path between a telephone terminal 11 and a telephone terminal 12 through the VoIP network 1, a public switched telephone network 2 and a voice band signal processing unit 230 of the ADSL modem 22. In the same manner, when it is determined that the MGC 62 is not operating, the MGC 61 sets a communication path between the telephone terminal 11 and the telephone terminal 12 through the VoIP network 1, the public switched telephone network 2 and the voice band signal processing unit 230 of the ADSL modem... |
| Selecting a network based on metrics for real time communication | 20060268828 | 20061130 |
| A system, method, and apparatus are directed towards routing a Voice over IP (VOIP) messages over a network. The VOIP messages are sent by a source client device to a destination client device through a portal service that has access to multiple routing services such as peering partners, carriers, etc. A VOIP system of the portal service aggregates call quality data after each VOIP call over each routing service. The call quality data is analyzed to determine a perception factor for each routing service at various times of day, days of the week, day of the year, geographic areas, and the like. When a VOIP call is requested through the portal service, the VOIP system determines a current cost, a current quality of service, and the... |
| Enhanced voip media flow quality by adapting speech encoding based on selected modulation and coding scheme (mcs) | 20060268837 | 20061130 |
| A voice-over-IP (VoIP) connection is established over a radio interface with a mobile radio station. A current radio condition for the VoIP connection is determined, and from that, a modulation and coding scheme (MCS) for a portion of the VoIP connection. A VoIP voice encoding mode for a portion of the VoIP connection is selected based on the determined modulation coding scheme. VoIP voice is then encoded into a number of VoIP encoded frames using the selected VoIP voice encoding mode which has an associated bit rate. An optimal number of VoIP encoded frames is included in a VoIP packet for transport over the VoIP connection given the selected voice encoding mode and the selected MCS. Other adjustments may be made to ensure robustness of the... |
| Authentication of an application layer media flow request for radio resources | 20060268838 | 20061130 |
| A radio access bearer authentication procedure prevents a service application running on a mobile station from obtaining a higher level of radio access bearer service than is authorized by the network operator. A secret identifier is determined both at the mobile station and at the radio network. When the mobile's service application requests a particular level of radio access bearer resources, the mobile sends its secret identifier to the radio network which compares the two. Such secret identifiers may be determined from a SIM associated with the mobile. If the secret identifiers match, the radio access network allocates the requested radio access bearer resources for the service application. One example service application is voice over IP (VoIP).
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| Voice over ip capturing | 20060268847 | 20061130 |
| An apparatus for Voice over IP capturing in a network, comprising: a data receiver, tapping the network and configured to receive at least one data packet transmitted in the network, a VoIP identifier, associated with the data receiver, configured to determine if the received data packet is a VoIP data packet qualifying for forwarding to a capture device, and a forwarder, associated with the VoIP identifier, configured to forward a copy of the qualifying VoIP data packet to at least one capture device, and to forward the received data packets back to the network.
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| Connection type handover of voice over internet protocol call based low-quality detection | 20060268848 | 20061130 |
| A telecommunications network comprises a base transceiver station node (28) and a packet control unit (PCU) 25. The base transceiver station node (28) serves, e.g., for providing radio transmission resources to a cell (C) for radio frequency communications. The packet control unit (PCU) 25 serves for allocating the radio transmission resources to respective voice over internet protocol (VoIP) calls handled as packet switched connections. In addition, for at least one VoIP call, the packet control unit (PCU) 25 is arranged for determining whether the at least one VoIP call should be changed from one connection type to another connection type, e.g., from a packet switched connection to a circuit switched connection. In an illustrated, example, non-limiting embodiment, the packet control unit (PCU) 25 determines whether the... |
| Method and apparatus for routing a call to a dual mode wireless device | 20060268928 | 20061130 |
| Disclosed is a system and method for routing a call to a dual mode wireless device. In accordance with an embodiment of the invention, a network node receives a call. The network node determines that the call is associated with a dual mode wireless device. Once this determination is made, the network node then selects one of a plurality of networks (e.g., cellular network or packet-based network, such as a VoIP network) for use in connecting the call to the dual mode wireless device. The network node then routes the call to the dual mode wireless device via the selected network. During the call, the network node re-routes the call to another network if the network node determines that this other network is now better suited... |
| Systems and methods for a fault tolerant voice-over-internet protocol (voip) architecture | 20060271811 | 20061130 |
| Systems and method for providing application layer fault tolerance in a VoIP architecture is shown and described. The method includes associating a virtual network address with one of a first communication device and second communication device, receiving a message from a network element, detecting a fault on an active one of the communication devices, and associating the virtual address with the other of the communication devices. Each of the first and second communication devices is coupled to a VoIP network. The virtual network address is associated with the active one of the communication devices. The detection of the fault occurs when the active communication device is at a first execution point of an application executing on the active communication device. When the virtual address is associated... |
| Systems and methods for providing redundant application servers | 20060271812 | 20061130 |
| Systems and methods for providing redundant application servers are described. A method of providing application server redundancy in a VoIP environment includes, receiving, at a standby server, application layer and signaling layer state information related to an active server and configuring the standby server to have substantially the same application layer and signaling layer state as the active server. The method also includes receiving, at the standby server, a copy of a message received by the active server and processing, by the standby server, the copy of the message to maintain synchronization between the state of the active server and the standby server.
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| Systems and methods for message handling among redunant application servers | 20060271813 | 20061130 |
| Systems and methods for messaging handling among redundant application servers are described. A method of providing application synchronization among a plurality of servers in an VoIP network environment includes pausing execution of an application on a standby server when the standby server encounters a checkpoint in the application and receiving a first message indicating that an active server reached the same checkpoint in a copy of the application executing on the active server. The method also includes transmitting, from the standby server, a second message to the active server indicating that the standby server received the first message and resuming execution of the application on the standby server.
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