Switching communications between different networks based on device capabilities

Methods and systems provided herein allow a communication session to be established on an optimal communication network and subsequently transferred to alternative networks if conditions dictate that such a change is necessary or desirable. The types of conditions which may be considered when identifying the optimal communication network or the alternative network include conditions related to the operation of a user device, communication profiles of the user, resources available on other networks, and the like.

Switching communications between different networks based on device capabilities

CAMERA-BASED FACIAL RECOGNITION AS A METHOD OF EFFECTING TELECOM DEVICE ALERTING

A camera can be associated with each conference participant endpoint. The camera, either frame or video-based, can monitor and detect one or more of gestures, facial recognition, emotions, and movements of the conference participant. Based on the detection of one or more of these triggering events, a correlation to an action corresponding the triggering event can be evoked. For example, if a participant raises their hand, e.g., a triggering event, the system can recognize that this is a request to speak. The participant can then be queued in the system based, for example, relative to other participants' requests. When the other participants have finished speaking, and it is the time for the user who raised their hand to speak, the system can optionally queue the user by modifying the endpoint with which they are associated.

A system for managing software versions in multitenant cloud IP video-telephony services

Internet protocol (IP) serverless Page Party (SPP) station and systems and methods for deploying multiple SPP stations

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station ...

TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME

An outlet (75) for a Local Area Network (80) (LAN), containing an integrated adapter (21) that converts VoIP to and from analog telephony, and a standard telephone jack (72) (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub (31a, 31c) that allows connecting both an analog telephone set (22a, 22c) via an adapter (21a, 21c), as well as retaining the data network connection, which may be accessed by a network jack (73a, 73c). The invention may also be applied to a telephone line- based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet ...

IP telephone device, IP telephone system, and setting confirmation method

An IP telephone system comprises a main device that manages outgoing and incoming calls of an IP telephone device, an external storage device storing network configuration information and telephone device configuration information, and an IP telephone device comprising a first interface section that uses in connection to the external storage device and a second interface section that uses in connection to a network. When automatically carrying out internal setting by connecting the external storage device to the first interface section, the IP telephone device obtains the network configuration information and the telephone device configuration information from the external storage device and, based on the obtained network configuration information and telephone device configuration information, carries out network setting and telephone device setting. The IP telephone device accesses the main device through the second interface section based on the setting and performs confirmation of the ...

CAMERA-BASED FACIAL RECOGNITION AS A METHOD OF EFFECTING TELECOM DEVICE ALERTING

Telephony discovery mashup and presence

Embodiments of the present invention are directed to a communication system that provides various automated operations, including searches of public and private resources for third party information and presentation of the located third party information by a mashup and preparation of a mashup to combine different communication modes or modes in a common session.

Jitter buffer

TELEPHONE CONNECTION WITH MORE PAKETTELEPHONIEADAPTER AND A NET OF WHICH THE TELEPHONE CONNECTION USES

TELEPHONE CONNECTION WITH MORE PAKETTELEFONIEADAPTER AND A NET, WHICH USES SELBIGEN

IP TELEPHONE SYSTEM

An IP telephone system equipped with a plurality of telephone terminals and a file server, wherein the file server comprises notification unit notifies an occurrence of update data of the software to the telephone terminals and transfer unit transfers the update data to a telephone terminal of a request source in response to a request from the update data from the telephone terminal, and the telephone terminal comprises reporting unit reports a fact of such to user when the occurrence of the update data is notified from the file server, a input unit used for inputting the update timing of the software by the user reported, request unit requests a transfer of the update data to the file server at the timing, and software update unit switches old software to new software based on the update data transferred from the file server in response to the request.

SYSTEM ENABLING COMMUNICATION BETWEEN AN AIRCRAFT-BASED COMPUTER NETWORK AND A GROUND-BASED COMPUTER NETWORK

The invention relates to a system enabling communication between an aircraft-based computer network and a ground-based computer network, characterised in that the system includes means for establishing a network communication between a ground-based computer network and an aircraft-based computer network via at least one communication medium using a synchronous communication mode. The invention can be used to simplify the procedures involved in maintaining a fleet of aircraft connected to a maintenance control center (MCC) and, in particular, to facilitate and secure the updating of aircraft logbooks (e.g. the cabin logbook and the technical logbook).

Method and system of controlling state of devices operable to support calls between endpoints

A method and system of controlling state for devices operable to support signaling or sessions between endpoints, such as the type used to support phone, video, messaging, and other calls. The state control may include transitioning the device between in-service and out-of-service states as a function of current, past or future operating demands on the devices.

Processing SMS messages

A first telephony identifier and a second telephony identifier are used for communication with a first, multi-identity user. Each of the first and second telephony identifiers are configured for use as different destination addresses in different SMSmessages sent to the first user. At least one SMS message is processed in SMS processing equipment in the network, and the SMS processing equipment performs identity management, in relation to the use of the first and second telephony identifiers for the first user, during processing of the at least one SMS message. The SMS processing equipment comprises an IP short message gateway (IP-SM-GW), where identity management takes place. The telephone network comprises a circuit-switched core and a packet-switched core, the gateway equipment processing SMS messages transmitted between both cores.

Seamless movement between phone pc with regard to applications, display, information transfer or swapping active device

TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME

An outlet (75) for a Local Area Network (80) (LAN), containing an integrated adapter (21) that converts VoIP to and from analog telephony, and a standard telephone jack (72) (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub (31a, 31c) that allows connecting both an analog telephone set (22a, 22c) via an adapter (21a, 21c), as well as retaining the data network connection, which may be accessed by a network jack (73a, 73c). The invention may also be applied to a telephone line- based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet ...

SIP Protocol-based IP telephone routing inspection system, method and server

REDUCED HIERARCHY KEY MANAGEMENT SYSTEM AND METHOD

Automated booting of a client for a subscriber is provided for clients that are for use in interactive user sessions that involve multimedia. A subscribe message is sent from the client to a proxy server. The proxy server authenticates the subscribe message, and sends the subscribe message to the configuration server. A notify message is sent to the client from the configuration server. The notify message is sent through the proxy server, and contains a location of a profile for the client. The profile is downloaded to the clients. This arrangement allows the persistence of profiles in a centralized place.

Method for Registering Multi-Contact Devices

A technique is described allowing an IMS enabled network to terminate a communication request directed towards a user device in a well-defined way, even for cases in which multiple registrations for IMS are present that are related to the same physical user device. A method embodiment of this technique comprises registering multiple user agents for one and the same user device, wherein from each user agent contact information including a device ID identifying the user device is provided; and processing a communication request directed towards the user device, the processing comprising identifying multiple identical device IDs of registered contacts, and treating the communication request according to a predetermined handling scheme related to the detection of identical device IDs.

METHOD AND APPARATUS FOR CONTROLLING CALL VOLUME

A method and apparatus for controlling a call volume for an office that serves as a protecting site for another office in a packet network are disclosed. For example, the method collects one or more customer registration counts from one or more session border controllers located in a first office, determines if the one or more customer registration counts have reached or exceeded a threshold. The method directs all of said one or more session border controllers located in said first office to enact one or more throttling rules if the one or more customer registration counts have reached or exceeded the threshold. 1. A method for controlling a call volume in a network , the method comprising:determining, by a processor, a number of calls traversing a first access office and that utilize a media gateway in a second access office of the network have exceeded a threshold; anddirecting, by the processor, at least one session border controller located in the first access office to enact a throttling, wherein the throttling comprises refusing to setup an inter office call, wherein the inter office call comprises a call traversing different access offices, wherein the different access offices comprise the first access office and at least the second access office.2. The method of claim 1 , wherein the throttling implements a throttling rule that is based on a number of inter office calls that the network is able to support.3. The method of claim 1 , wherein the throttling is based on a number of inter office calls that are allowed at any given time.4. The method of claim 1 , wherein the inter office call is determined from a numbering plan area code.5. The method of claim 1 , wherein the network is a packet network.6. The method of claim 1 , further comprising:disabling the throttling when the number of calls traversing the first access office and that utilize the media gateway in the second access office of the network have fallen below the threshold.7. The method of claim ...

Multiple user GUI

Mechanisms are provided herein which accommodate the sharing of a user device among multiple users. In particular, when multiple users are detected as sharing a user device, the user interface of the user device can be automatically altered to facilitate the sharing of the user device. Data associated with each sharing user may also be presented according to each user's presentation preferences and to accommodate presentation needs of each user.

A system for managing software versions in multitenant cloud IP video-telephony services

A shared computing system (200) for use in telecommunications between end-point telecommunication devices comprises a plurality of instances (212, 222; 214, 224; 216, 226) of a telecommunications service component and a version control gateway (232, 234) configured to route a communication connection from an end-point telecommunication device to an appropriate instance, of the plurality of instances, of the service component. Thus, different end-point telecommunication devices may use different instances of the service component. Different instances may run different software versions for compatibility with different end-point devices. Also disclosed is a method of migrating a user from one software version of a telecommunications service to another by creating instances of the new software version, suspending connections to and from an end-point device corresponding to a user, updating configuration data for the user and resuming connections to and from the end-point device before destroying ...

Telephony discovery mashup and presence

PROCEDURE FOR REGISTERING MULTI-CONTACT ARRANGEMENTS

Smarter business thinking mobile device

The present invention relates to providing a device for use in corporate systems and methods thereof, and during emergency situations.

ENHANCED SYSTEM OPERATION BY VIRTUALIZATION

A system and method for enhanced operation of internet protocol (IP) telephony is disclosed. The system comprises a call server. The call server includes a shared resources module, a virtual machine environment in communication with the shared resources module the virtual machine having a plurality of virtual machines. A plurality of telecommunication function applications operate on separate instances. Each instance is operable to be connected to a customer through a network connection in communication with at least one of the plurality of virtual machines.

COMMUNICATION APPARATUS, COMMUNICATION DATA RECORDING SYSTEM, COMMUNICATION METHOD, AND PROGRAM

Provided is an IP telephone and the like in which missing of speech start is suppressed. A communication device according to the present invention is provided with: a memory (110) in which data of communication with a communication destination are stored; and a control unit (120) which: performs transmission and reception of the data of communication with the communication destination at a first period; causes the communication data that have been transmitted or received to be stored in the memory (110); and, after a communication session is established with an external recording device for recording the communication data, causes the communication data stored in the memory (110) to be output to the external recording device at a second period shorter than the first period.

TELEPHONE SYSTEM, EXCHANGE, TELEPHONE EXCHANGING METHOD, AND TELEPHONE EXCHANGING PROGRAM

The present invention is provided with a first telephone, a repeater for managing first accommodation location information on the first telephone, and an exchange having a management table for managing as a set, for each of a plurality of second telephones, second telephone identification information, second accommodation location information on a second telephone connected to the repeater, and extension numbers. If the exchange receives a registration request containing first telephone identification information and first accommodation location information on the first telephone, and the first telephone identification information does not match any of the second telephone identification information registered in the management table but the first accommodation location information does match any of the second accommodation location information registered in the management table, then the second telephone identification information in the same set with the matching second accommodation ...

Voice call processing method and device

COMMUNICATION SYSTEM ENTERS a COMPUTER NETWORK an AIRCRAFT AND a COMPUTER NETWORK ON the GROUND

METHOD FOR AUTHENTICATING CALLER OF VOICE NETWORK BY USING DATA NETWORK

The present invention relates to a method for authenticating a caller of a voice network by using a data network. According to an embodiment of the present invention, the method for authenticating a caller of a voice network by using a data network comprises: a first step of matching identification information of at least one of outgoing information and incoming information with an authentication key value; a second step of detecting the outgoing information and the incoming information; a third step of confirming the authentication key value matched with the identification information of the at least one of the outgoing information and the incoming information; a fourth step of generating a verification key value by using the outgoing information and the incoming information; a fifth step of authenticating whether or not an outgoing device is valid; and a sixth step of outputting caller authentication data to a receiver. The method for authenticating a caller of a voice network by using ...

인터넷 프로토콜(IP) 서버리스 페이지 파티(SPP) 스테이션 및 다수의 SPP 스테이션들을 배치하기 위한 시스템들 및 방법들

... 서버리스 페이지 파티(SPP) 스테이션은 페이지 어나운싱 및 파티 라인 컨퍼런싱을 위해 제공되며, 이는 시스템 내의 모든 SPP 스테이션들이 런타임 시에 그들의 구성들을 상호 유지관리하게 할 수 있는 상호 프로비저닝 및 다중 마스터 스테이션 협상 및 마스터 페일오버를 구현하도록 구성된다. 복수의 SPP 스테이션들은 네트워크에 이더넷 케이블 또는 WiFi에 의해 접속되므로, 스테이션들 간에 값 비싸고 번거로운 케이블링 및 데이지 체인 구성이 필요하지 않게 된다. SPP 스테이션들은 멀티캐스트 기술을 통해, 그리고 IP-PBX 또는 스테이션 간 접속들을 제어하기 위한 다른 서버(예를 들어, SIP 서버)를 필요로 하지 않고, 서로 간에 패킷들을 라우팅한다. 상호 프로비저닝 모드에서, SPP 스테이션들은 커맨드 채널을 사용하여 이미 네트워크에 있는 다른 SPP 스테이션들로부터 획득되는 시스템 구성을 사용하여 동작한다. 시스템 구성은 마스터 스테이션으로서 지정된 SPP 스테이션에 의해 유지관리된다. 복수의 SPP 스테이션들은 새로운 마스터 스테이션을 지정하기 위해 그들 간에 협상할 수 있다.

station-PAGE swelling without server (sp) Internet Protocol (PI) and systems and methods for deploying multiple stations

VOIP VERIFIER

VoIP verification is initiated by subscriber-side test VoIP appliance, or VoIP proxy agent. Test VoIP appliance is coupled to test node over a communications internetwork by upstream and downstream datapaths. VoIP verification request is transmitted to test node. Responsively, test node transmits VoIP verification signal to test VoIP appliance. Perceivable indication of verification signal received by VoIP appliance is effected by visual display, audible indication, or both. Operational state verifier apparatus is disposed in a digital telephony device, a computer terminal, or with an ATA interface coupled to analog telephony device used in VoIP communications system.

Communication System and Method

A method of controlling a connection between a user terminal and an access node connected to a communication network is provided. The user terminal establishes a data connection with the access node, periodically generates a message at predetermined intervals and transmits the periodic message to at least one network node via the access node over the communication network. Responses to the periodic messages are received from the at least one network node. The responses are analysed to determine whether to terminate the connection to the access node, and in the case that the connection to the access node should be terminated, a disconnect message is transmitted to the access node from the user terminal.

Alternate routing of voice calls in a heavily loaded SIP network

Methods and systems for supporting the alternate routing of calls between endpoints, such as session initiation protocol (SIP) endpoints, are provided. An inner gateway alternate route (IGAR) application is invoked when the primary, Internet protocol, communication network is overloaded or otherwise unable to handle a call. The IGAR application assigns incoming and outgoing trunks, and serves as a contact point for session managers in routing the trunk call.

Providing a unified communication history of a multi-modal communication

Methods are provided for providing a unified communication history of a multi-modal communication. In one aspect, a method includes receiving a plurality of event logs related to the multi-modal communication. The multi-modal communication comprises a first communication modality and a second communication modality that is different from the first communication modality. The method includes aggregating the one or more events from the event logs as a plurality of aggregated events and resolving an identity of each user from among the plurality of users in the plurality of aggregated events. The method includes storing the aggregated events into a communication history of the multi-modal communication. Systems and machine-readable media are also provided.

RELIABILITY OF A CONNECTION DURING A COMMUNICATION SESSION ON A NETWORK DEVICE

The present application details methods and systems for improving the reliability of a connection during a communication session on a network device. In one or more embodiments, the network device maps to and connects with multiple datacenters. For example, the network device connects with a primary datacenter and participates in a communication session via a primary media bridge connection while jointly being connected with a secondary datacenter. In addition, the network device can switch from the primary media bridge connection to a secondary media bridge connection if a fault is detected on the primary media bridge connection.

Seamless movement between phone pc with regard to applications, display, information transfer or swapping active device

It would be beneficial to allow one or more of information, applications, desktops, and in general any information to be shared between a computer and a communications device. For example, it would be useful to be able to move a presentation from a computer to a mobile communications device. Some solutions attempt to address this by requiring a user to load the information on a laptop, USB key, burn it to an optical disk or the like, and then "open" the information on another device located at the remote destination. In one embodiment, through the use of one or more of virtualization and content translation, any information and/or applications can be exchanged between a computer and a communications device. More specifically, one embodiment utilizes metadata in conjunction with a virtualization environment to allow the exchange of information and/or applications between a personal computer and a communications device, or vice versa.

IP Telephone system

Jitter buffer

Multiple user gui

TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME

An outlet (75) for a Local Area Network (80) (LAN), containing an integrated adapter (21) that converts VoIP to and from analog telephony, and a standard telephone jack (72) (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub (31a, 31c) that allows connecting both an analog telephone set (22a, 22c) via an adapter (21a, 21c), as well as retaining the data network connection, which may be accessed by a network jack (73a, 73c). The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation.

TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME

SYSTEM ENABLING COMMUNICATION BETWEEN AN AIRCRAFT-BASED COMPUTER NETWORK AND A GROUND-BASED COMPUTER NETWORK

The invention relates to a system enabling communication between an aircraft-based computer network and a ground-based computer network, characterised in that the system includes means for establishing a network communication between a ground-based computer network and an aircraft-based computer network via at least one communication medium using a synchronous communication mode.

METHOD FOR AUTOMATICALLY REGISTERING A USER IN A DESK-SHARE ENVIRONMENT AND IP TELEPHONE

The present invention relates to a method for automatically registering a user in a desk-share environment comprising a plurality of desks, each desk being equipped with an IP telephone connected to a communication network, in particular, to a local area network, wherein the IP telephone holds a data base comprising data of all desk-share users of the desk-share environment, the data comprising at least a user ID and a MAC address of a terminal device for each user assigned to a user profile, the method comprising the steps of: receiving, at the IP telephone, an IP data packet from a first terminal device via the communication network; verifying the MAC address in the IP data packet received from the first terminal device; and when the MAC address corresponds to a MAC address in the data of the data base, activating the user profile assigned to the MAC address in the IP telephone. Further, the invention relates to an IP telephone which is adapted to carry out the method for automatically ...

Customizing network-level settings for a network device on a communication network

The present application describes methods and systems for improving user communication in a network-based communication system. In one or more embodiments, a datacenter provides a unique connection address network device that connects to the datacenter. By providing a unique connection address, the datacenter can provide customized network-level settings on a per network device basis. In addition, the datacenter can also provide customized network identification on a per network device basis.

СИСТЕМА СВЯЗИ МЕЖДУ СЕТЬЮ КОМПЬЮТЕРОВ В ЛЕТАТЕЛЬНОМ АППАРАТЕ И СЕТЬЮ КОМПЬЮТЕРОВ НА ЗЕМЛЕ

FIELD: radio engineering, communication. SUBSTANCE: system includes means of establishing a network connection between a ground-based computer network and an aircraft-based computer network via at least one communication medium using a synchronous communication mode. EFFECT: high efficiency of using communication channels. 8 cl, 7 dwg

СИСТЕМА СВЯЗИ МЕЖДУ СЕТЬЮ КОМПЬЮТЕРОВ В ЛЕТАТЕЛЬНОМ АППАРАТЕ И СЕТЬЮ КОМПЬЮТЕРОВ НА ЗЕМЛЕ

... 1. Система связи между сетью компьютеров в летательном аппарате и сетью компьютеров на земле, отличающаяся тем, что содержит средства, выполненные с возможностью установления сетевого соединения между сетью компьютеров на земле и сетью компьютеров в летательном аппарате через, по меньшей мере, один носитель связи в режиме синхронной связи. ! 2. Система связи по п.1, отличающаяся тем, что содержит средства обновления данных, хранящихся в памяти в сети компьютеров летательного аппарата, на основании данных, хранящихся в памяти в сети компьютеров на земле. ! 3. Система связи по п.1, отличающаяся тем, что содержит средства передачи данных, хранящихся в памяти в сети компьютеров летательного аппарата, в сеть компьютеров на земле. ! 4. Система связи по п.1, отличающаяся тем, что, по меньшей мере, один носитель связи является сетью мобильной телефонии. ! 5. Система связи по п.1, отличающаяся тем, что, по меньшей мере, один носитель связи является сетью беспроводной связи. ! 6. Система связи по ...

Processing SMS messages

There is provided a method for processing communications in a telephony network. The method includes configuring a first telephony identifier for communication with a first user and a second user, and a second telephony identity for communication withthe second user, each of the telephony identifiers being configured for use as different destination addresses in different SMS messages sent to the second user. The method includes processing at least one SMS message in SMS processing equipment in the network, and during this processing, performing identity management, in relation to the use of the first and second telephony identifiers for the second user. In this way, the second user equipment may be used to send SMS messages independently of the shared identity. This is useful if the user equipment is associated with a business identity and also a personal identity, each identity being associated with a different telephony identifier and only the business identity being a shared identity ...

Communication system and method

Controlling a connection between a user terminal and an access node connected to a communication network

TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME

An outlet (75) for a Local Area Network (80) (LAN), containing an integrated adapter (21) that converts VoIP to and from analog telephony, and a standard telephone jack (72) (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub (31a, 31c) that allows connecting both an analog telephone set (22a, 22c) via an adapter (21a, 21c), as well as retaining the data network connection, which may be accessed by a network jack (73a, 73c). The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation.

Method and system of controlling state devices operable to support calls between endpoints

A method and system of controlling state for devices operable to support signaling or sessions between endpoints, such as the type used to support phone, video, messaging, and other calls. The state control may include transitioning the device between in-service and out-of-service states as a function of current, past or future operating demands on the devices.

Internet protocol (IP) serverless page party (SPP) station and systems and methods for deploying multiple SPP stations

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station ...

Method and system for managing routing across multiple networks with separate routing masters

Novel tools and techniques are provided for implementing management of routing across multiple voice or data networks with separate routing masters. In various embodiments, in response to receiving a request to establish a call between a calling party in a first network and a called party in a second network, a computing system might receive a first set of network information from a first routing database(s) that is operated by a first service provider and a second set of network information from a second routing database(s) that is operated by a second service provider separate from the first service provider; might analyze the received first and second sets of network information to generate a unified routing model for optimizing routing of the call through the first and second networks; and might establish the call through a selected optimized route based on the generated unified routing model.

SYSTEM ENABLING COMMUNICATION BETWEEN AN AIRCRAFT-BASED COMPUTER NETWORK AND A GROUND-BASED COMPUTER NETWORK

IP telephone system

Controlling a connection between a user terminal and an access node based on remaining user credit

A method of controlling a connection between a user terminal and an access node connected to a communication network is provided. The user terminal establishes a data connection with the access node, periodically generates a message at predetermined intervals and transmits the periodic message to at least one network node via the access node over the communication network. Responses to the periodic messages are received from the at least one network node. The responses are analysed to determine whether to terminate the connection to the access node, and in the case that the connection to the access node should be terminated, a disconnect message is transmitted to the access node from the user terminal. Analysing the response could indicate a command to terminate the connection or involve determining that a response has not been received within a predetermined time interval. The periodic messages may be tick messages. The connection may be terminated in response to the at least one network ...

Processing SMS messages

apparatus and method for automatic repeat request for allocating transmission overhead-features, reduced, in a communication system a VoIP wireless

Smarter Business Thinking Mobile Device

The present invention relates to providing a device for use in corporate systems and methods thereof, and during emergency situations.

Backup server architecture in a VoIP system

A method for providing telephony services in a distributed VoIP system includes determining that a first switch is unable to communicate with a first server, where the first switch is configured to provide telephony services to a first plurality of communication devices, and the first server is configured to provide applications and data associated with the first plurality of communication devices to the first switch. The method also includes sending an adoption request from the first switch to a second server, where the second server is configured to provide applications and data associated with a second plurality of communication devices to a second switch and also configured to provide the applications and data associated with the first plurality of communication devices to the first switch. The method also includes receiving at the first switch the applications and data associated with the first plurality of communication devices from the second server.

INTERNET PROTOCOL (IP) SERVERLESS PAGE PARTY (SPP) STATION AND SYSTEMS AND METHODS FOR DEPLOYING MULTIPLE SPP STATIONS

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station ...

Multiple user identity and bridge appearance

A call control system is provided that can determine the location of a person. From the location of the person, the call control system can temporarily associate a different phone with the person. A bridge appearance may then be changed to include the temporary association such that calls from the different phone show the person as a party to the call. Further, a routing algorithm for routing calls to phones within an enterprise may then be temporarily modified to route calls to the different phone for the person. Thus, a person may travel through the facilities of the enterprise and have important calls routed to the person wherever they may be.

Switching communications between different networks based on device capabilities

Telephone call terminal

TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME

An outlet (75) for a Local Area Network (80) (LAN), containing an integrated adapter (21) that converts VoIP to and from analog telephony, and a standard telephone jack (72) (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub (31a, 31c) that allows connecting both an analog telephone set (22a, 22c) via an adapter (21a, 21c), as well as retaining the data network connection, which may be accessed by a network jack (73a, 73c). The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet ...

IP telephone software update allowing a user to specify the time of update

An IP telephone system equipped with a plurality of telephone terminals and a file server, wherein the file server notifies an occurrence of update data of the software to the telephone terminals and transferring the update data to a telephone terminal in response to a request from the update data from the telephone terminal, and the telephone terminal reporting to a user when the occurrence of the update data is notified from the file server, an input unit used for inputting the update timing of the software by the user reported, and a request unit requesting transfer of the update data to the file server at the timing, and software update unit switching old software to new software based on the update data transferred from the file server in response to the request. The present invention allows a user to input the time of update of the software program. In an alternative embodiment, a status management unit manages the user status including login/logout information of the user and presence ...

IP Telephone system

Processing SMS messages

TELEPHONE SYSTEM, EXCHANGE, TELEPHONE EXCHANGING METHOD, AND TELEPHONE EXCHANGING PROGRAM

The present invention is provided with a first telephone, a repeater for managing first accommodation location information on the first telephone, and an exchange having a management table for managing as a set, for each of a plurality of second telephones, second telephone identification information, second accommodation location information on a second telephone connected to the repeater, and extension numbers. If the exchange receives a registration request containing first telephone identification information and first accommodation location information on the first telephone, and the first telephone identification information does not match any of the second telephone identification information registered in the management table but the first accommodation location information does match any of the second accommodation location information registered in the management table, then the second telephone identification information in the same set with the matching second accommodation location information is changed to the first telephone identification information in the management table, and an extension number in the same set with the matching second accommodation location information is used as the extension number of the first telephone.

SYSTEM ENABLING COMMUNICATION BETWEEN AN AIRCRAFT-BASED COMPUTER NETWORK AND A GROUND-BASED COMPUTER NETWORK

The invention relates to a system enabling communication between an aircr aft-based computer network and a ground-based computer network, characterise d in that the system includes means for establishing a network communication between a ground-based computer network and an aircraft-based computer netw ork via at least one communication medium using a synchronous communication mode. The invention can be used to simplify the procedures involved in maint aining a fleet of aircraft connected to a maintenance control center (MCC) a nd, in particular, to facilitate and secure the updating of aircraft logbook s (e.g. the cabin logbook and the technical logbook).

INTERNET PROTOCOL (IP) SERVERLESS PAGE PARTY (SPP) STATION AND SYSTEMS AND METHODS FOR DEPLOYING MULTIPLE SPP STATIONS

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station ...

Internet protocol (IP) serverless page party (SPP) station and systems and methods for deploying multiple SPP stations

COMMUNICATION SYSTEM ENTERS a COMPUTER NETWORK an AIRCRAFT AND a COMPUTER NETWORK ON the GROUND

L'invention est relative à un système de communication entre un réseau d'ordinateurs dans un aéronef et un réseau d'ordinateurs au sol, caractérisé en ce qu'il comprend des moyens aptes à établir une connexion réseau entre un réseau d'ordinateurs au sol et un réseau d'ordinateurs dans l'aéronef via au moins un support de communication selon un mode de communication synchrone. The invention relates to a communication system between a computer network in an aircraft and a computer network on the ground, characterized in that it comprises means capable of establishing a network connection between a computer network at ground and a computer network in the aircraft via at least one communication medium according to a synchronous communication mode.

VoIP verifier

VoIP verification is initiated by subscriber-side test VoIP appliance, or VoIP proxy agent. Test VoIP appliance is coupled to test node over a communications internetwork by upstream and downstream datapaths. VoIP verification request is transmitted to test node. Responsively, test node transmits VoIP verification signal to test VoIP appliance. Perceivable indication of verification signal received by VoIP appliance is effected by visual display, audible indication, or both. Operational state verifier apparatus is disposed in a digital telephony device, a computer terminal, or with an ATA interface coupled to analog telephony device used in VoIP communications system.

BANDWIDTH UTILIZATION DURING SILENCE FRAMES

Briefly, in accordance with one or more embodiments, an apparatus of a user equipment (UE) comprises one or more baseband processors to generate a voice call invite message for a remote UE, and a memory to store the voice call invite message. The voice call invite message includes a header p-use-silence-period to indicate that the UE is configured to transmit or receive non-voice data during a silence period of the voice call. 130-. (canceled)31. An apparatus of a user equipment (UE) , comprising:one or more baseband processors to generate a voice call invite message for a remote UE; anda memory to store the voice call invite message;wherein the voice call invite message includes a header p-use-silence-period to indicate that the UE is configured to transmit or receive non-voice data during a silence period of the voice call.32. The apparatus of claim 31 , further comprising a radio-frequency (RF) transceiver to transmit the voice call invite message to the remote UE via a core network.33. The apparatus of claim 31 , wherein the header indicates a type of non-voice data to be transmitted or received during a silence period.34. The apparatus of claim 31 , wherein the one or more baseband processor are to generate one or more non-voice data packets to be sent to the remote UE during a silence period of the UE.35. The apparatus of claim 31 , wherein the one or more baseband processors are to decode one or more non-voice data packets received from the remote UE during a silence period of the remote UE.36. The apparatus of claim 31 , wherein the one or more baseband processors are to generate a Silence Indicator Description (SID) frame to be sent from the UE to the remote UE prior to sending one or more non-voice data packets to the remote UE during a silence period of the UE.37. The apparatus of claim 31 , wherein the voice call comprises a Voice over Long Term Evolution (VoLTE) call.38. The apparatus of claim 31 , wherein the non-voice data comprises message data claim ...

Multiple user identity and bridge appearance

Internet protocol (IP) serverless Page Party (SPP) station and systems and methods for deploying multiple SPP stations

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station ...

TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME

An outlet (75) for a Local Area Network (80) (LAN), containing an integrated adapter (21) that converts VoIP to and from analog telephony, and a standard telephone jack (72) (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub (31a, 31c) that allows connecting both an analog telephone set (22a, 22c) via an adapter (21a, 21c), as well as retaining the data network connection, which may be accessed by a network jack (73a, 73c). The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet ...

Method for multi-interface registration

A technique is described allowing an IMS enabled network to terminate a communication request directed towards a user device in a well-defined way, even for cases in which multiple registrations for IMS are present that are related to the same physical user device. A method embodiment of this technique comprises registering multiple user agents for one and the same user device, wherein from each user agent contact information including a device ID identifying the user device is provided; and processing a communication request directed towards the user device, the processing comprising identifying multiple identical device IDs of registered contacts, and treating the communication request according to a predetermined handling scheme related to the detection of identical device IDs.

Graphical user interface (GUI) for administering a voice over internet protocol (VOIP) network implementing media aggregation

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 all communication packets communicated between the residents to travel along the selected path.

Processing SMS messages

Transmitter processing control according to jitter buffer status information of the receiver

Transmitting data of a real-time communication event from the transmitter to a jitter buffer of a receiver is described, such as during a video telephone call. Jitter buffer state information is received at the transmitter from the receiver, the jitter buffer state information indicating a state of the jitter buffer. At least one processing parameter is controlled based on the received jitter buffer state information, the at least one processing parameter describing how data is to be processed for transmission from the transmitter to the jitter buffer in the real-time communication event. Data is processed for transmission from the transmitter to the jitter buffer in accordance with the determined at least one processing parameter. The processed data is transmitted from the transmitter to the jitter buffer of the receiver in the real-time communication event. Example embodiments include changing a quality parameter of the transmitted data according to the jitter buffer status information ...

Multiple user GUI

Method and apparatus for controlling call volume

A method and apparatus for controlling a call volume for an office that serves as a protecting site for another office in a packet network are disclosed. For example, the method collects one or more customer registration counts from one or more session border controllers located in a first office, determines if the one or more customer registration counts have reached or exceeded a threshold. The method directs all of said one or more session border controllers located in said first office to enact one or more throttling rules if the one or more customer registration counts have reached or exceeded the threshold.

A SYSTEM AND METHOD FOR ALERTING A PARTY IN A CALL OF A CALL DISCONNECTION OF ANOTHER PARTY

Social network virtual private network

A system and method create an interconnection among users of different social networks. Each user can provide information about the social networks they use (e.g., username, passwords, etc.). An application may register with the social network and can inter- dispose the application between the user and the social network. Then, the system can interconnect the users into a "virtual private network" (VPN) where communications between the users and the social networks can be replicated to other social networks for the other users. As such, different users using different social networks can communicate.

Seamless movement between phone and PC with regard to applications, display, information transfer or swapping active device

IP telephone system

A system for managing software versions in multitenant cloud IP video-telephny services

Multiple user identity and bridge appearance

APPARATUS AND METHOD FOR MANAGING THE NETWORK RESOURCE

Telephone outlet with packet telephony adaptor, and a network using same

An outlet for a Local Area Network (LAN), containing an integrated adapter that converts VoIP to and from analog telephony, and a standard telephone jack (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub that allows connecting both an analog telephone set via an adapter, as well as retaining the data network connection, which may be accessed by a network jack. The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking ...

IP telephony network using a configuration map for organizing sites in a tree-like hierarchy

A system and method for managing an IP telephony network that uses a configuration map for organizing sites in a tree-like hierarchy. The system is capable of managing the addition, removal and movement of IP telephony elements within an IP telephony network and locating services and IP telephony elements within the IP telephony network in an efficient manner.

METHOD AND SYSTEM OF CONTROLLING STATE OF DEVICES OPERABLE TO SUPPORT CALLS BETWEEN ENDPOINTS

A method and system of controlling state for devices operable to support signaling or sessions between endpoints, such as the type used to support phone, video, messaging, and other calls. The state control may include transitioning the device between in-service and out-of-service states as a function of current, past or future operating demands on the devices. 1. A gateway operable to service calls between a plurality of endpoints , the gateway being operable to:an in-service (IS) state where the gateway is enabled to service calls between the plurality of endpoints;an out-of-service (OOS) state where the gateway is disabled from servicing calls; anda pending state where the gateway monitors one or more operating characteristics to determine occurrence of a trigger event, including automatically a transitioning from the IS state to the OOS state upon occurrence of the trigger event.2. The gateway of further being operable to determine the triggering event as a function of at least one of a number and type of calls actively being serviced through the gateway.3. The gateway of further being operable to determine the triggering event to occur when the number of active calls is zero.4. The gateway of further being operable to notify participants that each call active will be dropped at a time of transitioning to the OOS state.5. The gateway of further being operable claim 1 , as at least part of the pending state claim 1 , to an out-of-service pending new calls blocked state wherein existing calls are permitted to complete and new calls are blocked.6. The gateway of further being operable to define the triggering event according to instructions received from a service scheduler.7. The gateway of further being operable to transition to the OOS state prior to notifying the service scheduler of occurrence of the triggering event.8. The gateway of further operable to prevent automatically transitioning to the OOS state upon occurrence of the trigger event if the triggering ...

INTERNET PROTOCOL (IP) SERVERLESS PAGE PARTY (SPP) STATION AND SYSTEMS AND METHODS FOR DEPLOYING MULTIPLE SPP STATIONS

A serverless Page Party (SPP) station is provided for page announcing and party line, conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e,g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station. 1. An Internet Protocol (IP)-based station that provides at least one of page announcing and party line conferencing with respect to other IP-based stations , the IP-based station connected to other IP-based stations in a network that supports multicast and point-to-point communication , the IP-based station comprising:a network interface to connect the IP-based station to the network;a selector input configured to select from a plurality of party lines when the IP-based station is configured to transmit and receive audio in party line conferencing with other IP-based stations and transmit and a plurality of page lines when the IP-based station is configured to transmit and receive page announcements with respect to the other IP-based stations, the party lines being assigned respective party line sockets and the page lines being assigned respective page sockets in accordance with a system configuration; anda processing device that operates in a mutual provisioning mode to configure ...

Method and system for booting, provisioning and activating hardware and software clients

Automated booting of a client for a subscriber is provided for clients that are for use in interactive user sessions that involve multimedia. A subscribe message is sent from the client to a proxy server. The proxy server authenticates the subscribe message, and sends the subscribe message to the configuration server. A notify message is sent to the client from the configuration server. The notify message is sent through the proxy server, and contains a location of a profile for the client. The profile is downloaded to the clients. This arrangement allows the persistence of profiles in a centralized place.

Seamless movement between phone and pc with regard to applications, display, information transfer or swapping active device

It would be beneficial to allow one or more of information, applications, desktops, and in general any information to be shared between a computer and a communications device. For example, it would be useful to be able to move a presentation from a computer to a mobile communications device. Some solutions attempt to address this by requiring a user to load the information on a laptop, USB key, burn it to an optical disk or the like, and then “open” the information on another device located at the remote destination. In one embodiment, through the use of one or more of virtualization and content translation, any information and/or applications can be exchanged between a computer and a communications device. More specifically, one embodiment utilizes metadata in conjunction with a virtualization environment to allow the exchange of information and/or applications between a personal computer and a communications device, or vice versa.

ELECTRONIC DEVICE FOR DISPLAYING AND UPDATING MOBILE NETWORK OPERATION INFORMATION AND CONTROLLING METHOD THEREOF

An electronic device which displays and updates Mobile Network Operator (MNO) information includes a memory, a display, a communicator, a user inputter, and a processor. The processor is configured to control the display to display a contact list including telephone number information of a plurality of external devices and Mobile Network Operator (MNO) information of the plurality of external devices, and in response to a user command for selecting one of the plurality of external devices included in the contact list being input, establish a communication connection with the selected external device. The processor is further configured to control the communicator to receive information regarding the MNO information of the external device from the selected external device; and update the contact list based on the MNO information of the external device and store the updated contact list in the memory. 1. An electronic device comprising:a memory;a display;a communicator;a user inputter; anda processor operably connected to the memory, the display, the communicator, and the user inputter, and configured to control the electronic device,wherein the processor is configured to:control the display to display a contact list including telephone number information of a plurality of external devices and Mobile Network Operator (MNO) information of the plurality of external devices;in response to a user command for selecting one of the plurality of external devices included in the contact list being input, establish a communication connection with the selected external device;in response to the communication connection being established, control the communicator to receive information regarding the MNO information of the external device from the selected external device; andin response to the received MNO information of the external device being different from MNO information of a pre-stored external device, update the contact list based on the MNO information of the external ...

MOBILE DEVICE AUDIO TUNING

Various of the disclosed embodiments improve encoding during a network call, such as a Voice Over Internet Protocol (VOIP) connection, by correlating various contextual parameters from previous calls, with appropriate settings for a current call. For example, the system may take note of the model of cell phone used during a communication, the carrier, the presence of a WiFi connection, the user rating, the codecs employed, etc. During a subsequent call, the system may compare these past calling parameters with the current situation, and may select call settings (e.g., codec selections) based thereon. Machine learning methods may be applied using the past data to inform the selection of the settings for the present call. 1. A computer-implemented method , comprising:determining a grouping associated with a user device;determining a first configuration profile associated with the grouping;applying at least a portion of the configuration profile to the user device during a communication by the user device;receiving feedback from the user regarding the quality of the communication; andadjusting the first configuration profile based upon the feedback to create a second configuration profile.2. The computer-implemented method of claim 1 , wherein determining a grouping associated with a user device comprises determining a location of the user device claim 1 , a bandwidth associated with a network in communication with the user device.3. The computer-implemented method of claim 1 , wherein determining a grouping comprises assessing each of features associated with the user claim 1 , the user device hardware claim 1 , and a network environment associated with the user device.4. The computer-implemented method of claim 1 , wherein determining a first configuration profile comprises interpolating portions from a plurality of configuration profiles to generate the first configuration profile.5. The computer-implemented method of claim 1 , further comprising adjusting the first ...

TELEPHONE OUTLET WITH PACKET TELEPHONY ADAPTER, AND A NETWORK USING SAME

An outlet for a Local Area Network (LAN), containing an integrated adapter that converts VoIP to and from analog telephony, and a standard telephone jack (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub that allows connecting both an analog telephone set via an adapter, as well as retaining the data network connection, which may be accessed by a network jack. The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation.

Method and Apparatus for Performing Wireless Communications

Configuration information is received for a first bearer for a first machine-type communication service type and a second bearer for at least one second machine-type communication service type, wherein the first service type includes a voice over internet protocol service. A base station communicates using said first bearer for said first service type and the second bearer for said at least one second service type. 1. A method , comprising:receiving configuration information for a first bearer for a first machine-type communication service type and a second bearer for at least one second machine-type communication service type, said first service type comprising a voice over internet protocol service; andcommunicating with a base station using said first bearer for said first service type and the second bearer for said at least one second service type.2. A method comprising:transmitting, to an apparatus, configuration information for a first bearer for a first machine-type communication service type and a second bearer for at least one second machine-type communication service type, said first service type comprising a voice over Internet protocol service; andcommunicating with the apparatus using said first bearer for said first service type and the second bearer for said at least one second service type.3. The method as claimed in claim 2 , comprising:determining a metric of a signal quality for communications with the apparatus;comparing the metric to a threshold value; anddetermining the configuration information in dependence on the comparison.4. The method as claimed in claim 3 , comprising determining the threshold value dynamically.5. The method as claimed in claim 2 , comprising:determining a metric of a signal quality for communications with the apparatus; andusing the metric to determine a number of repetitions a channel transmitted to the apparatus is to have.6. The method as claimed in claim 3 , comprising:receiving an indication of the signal quality ...

Internet protocol (ip) serverless page party (spp) station and systems and methods for deploying multiple spp stations

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station.

IP TELEPHONY NETWORK USING A CONFIGURATION MAP FOR ORGANIZING SITES

A system and method for managing an IP telephony network that uses a configuration map for organizing sites in a tree-like hierarchy. The system is capable of managing the addition, removal and movement of IP telephony elements within an IP telephony network and locating services and IP telephony elements within the IP telephony network in an efficient manner. 123-. (canceled)24. A method comprising:providing, by a processor, services and a router to a server that is being added to an IP telephony network having sites and other servers;updating, by the processor, a configuration map as to a location of the server, the configuration map including a location of each of the sites relative to each other site, and the configuration map organizing the sites according to a tree-like hierarchy; andupdating, by the processor, the other servers as to the location of the server based on the configuration map.25. The method of claim 24 , wherein:the IP telephony network further has switches at a site of the server,the site of the server is one of the sites; andthe method further comprises updating, by the processor, the switches at the site of the server as to the location of the server based on the configuration map.26. The method of claim 25 , wherein:the switches include respective site maps with respective server assignments and respective backup server assignments associated with the switches; andthe updating of the switches further comprises updating, by the processor, the respective site maps as to the location of the server based on the configuration map.27. The method of claim 24 , further comprising:determining, by the processor, that a site of the server is connected to at least one younger generation site in the tree-like hierarchy based on the configuration map, the site of the server being one of the sites, and the at least one younger generation site being another one of the sites and including IP telephony elements; andupdating, by the processor, the IP ...

METHOD AND APPARATUS FOR CONTROLLING CALL VOLUME

A method and apparatus for controlling a call volume for an office that serves as a protecting site for another office in a packet network are disclosed. For example, the method collects one or more customer registration counts from one or more session border controllers located in a first office, determines if the one or more customer registration counts have reached or exceeded a threshold. The method directs all of said one or more session border controllers located in said first office to enact one or more throttling rules if the one or more customer registration counts have reached or exceeded the threshold. 1. A method for controlling a call volume in a network , comprising:collecting, by a processor, customer registration counts from session border controllers located in a first office;determining, by the processor, whether the customer registration counts have exceeded a threshold; anddirecting, by the processor, the session border controllers located in the first office to enact a throttling when the customer registration counts have exceeded the threshold, wherein the throttling comprises refusing to setup an inter office call, wherein the inter office call comprises a call traversing different access offices.2. The method of claim 1 , wherein the throttling implements a throttling rule that is based on a number of inter office calls that the network is able to support.3. The method of claim 1 , wherein the throttling is based on a number of inter office calls that are allowed at any given time.4. The method of claim 1 , wherein the inter office call is determined from a numbering plan area code.5. The method of claim 1 , wherein an increase of the customer registration counts is due to a fail-over event associated with another office.6. The method of claim 1 , wherein the network is a packet network.7. The method of claim 1 , further comprising:disabling the throttling if the customer registration counts have fallen below the threshold.8. The method of ...

PROCESSING SMS MESSAGES

A first telephony identifier and a second telephony identifier are used for communication with a first, multi-identity user. Each of the first and second telephony identifiers are configured for use as different destination addresses in different SMS messages sent to the first user. At least one SMS message is processed in SMS processing equipment in the network, and the SMS processing equipment performs identity management, in relation to the use of the first and second telephony identifiers for the first user, during processing of the at least one SMS message. 1. A method for processing communications in a telephony network , the method comprising:processing at least one SMS message in SMS processing equipment in the telephony network, wherein first and second telephony identifiers are associated with a first, multi-identity user, and wherein each of the first and second telephony identifiers is useable as a destination address in an SMS message sent to the first user; and accessing stored data indicative of an association between at least one characteristic derivable from an SMS message and at least one of the first and second telephony identifiers; and', 'selecting one of the first telephony identifier and the second telephony identifier on the basis of the stored data,', 'wherein the telephony network comprises a circuit-switched core, a packet-switched core and gateway equipment connecting the circuit-switched core and the packet-switched core, wherein the gateway equipment implements transport layer interworking for SMS messages transmitted between the packet-switched core and the circuit-switched core, and wherein the identity management is performed at the gateway equipment., 'performing identity management, in relation to the use of the first and second telephony identifiers for the first user, during said processing of the at least one SMS message at the SMS processing equipment, wherein said performing said identity management comprises2. The method of ...

PROCESSING SMS MESSAGES

There is provided a method for processing communications in a telephony network. The method includes configuring a first telephony identifier for communication with a first user and a second user, and a second telephony identity for communication with the second user, each of the telephony identifiers being configured for use as different destination addresses in different SMS messages sent to the second user. The method includes processing at least one SMS message in SMS processing equipment in the network, and during this processing, performing identity management, in relation to the use of the first and second telephony identifiers for the second user. 1. A method for processing communications in a telephony network , the method comprising:processing at least one SMS message in SMS processing equipment in the telephony network, wherein a first telephony identifier is associated with a first user and a second user, wherein a second telephony identifier is associated with the second user, and wherein each of the first and second telephony identifiers is useable as a destination address in an SMS message sent to the second user; and accessing group messaging indicator data indicative of an association between the first user and the second user; and', 'selecting the first telephony identifier on the basis of the group messaging indicator data., 'performing identity management, in relation to the use of the first and second telephony identifiers for the second user, during said processing of the at least one SMS message at the SMS processing equipment, wherein said performing identity management comprises2. The method of claim 1 , wherein the telephony network comprises a circuit-switched core claim 1 , a packet-switched core and gateway equipment connecting the circuit-switched core and the packet-switched core claim 1 , the gateway equipment processing SMS messages transmitted between the packet-switched core and the circuit-switched core claim 1 , and wherein the ...

COMMUNICATION APPARATUS, COMMUNICATION DATA RECORDING SYSTEM, COMMUNICATION METHOD, AND PROGRAM

The present invention provides an IP telephone and the like capable of preventing a loss of the beginning part of a talk. A communication apparatus according to the present invention includes: a memory () configured to store communication data with a communication destination; and a control unit () configured to transmit and receive the communication data to and from the communication destination in each first duration and store the transmitted and received communication data in the memory (), then output, after a communication session with an external recording apparatus that records the communication data is established, the communication data stored in the memory () to the external recording apparatus in each second duration that is shorter than the first duration. 1. A communication apparatus comprising:a memory configured to store communication data with a communication destination; anda control unit configured to transmit and receive the communication data to and from the communication destination in each first duration and store the transmitted and received communication data in the memory, then output, after a communication session with an external recording apparatus that records the communication data is established, the communication data stored in the memory to the external recording apparatus in each second duration that is shorter than the first duration, whereinthe control unit outputs, after a communication session with the communication destination is established, a request signal for establishing a communication session to the external recording apparatus.2. (canceled)3. The communication apparatus according to claim 1 , wherein the control unit determines a length of a second duration after receiving claim 1 , from the external recording apparatus claim 1 , a response signal to the request signal output for establishing the communication session with the external recording apparatus.4. The communication apparatus according to claim 1 , wherein the ...

Method and System for Managing Routing Across Multiple Networks with Separate Routing Masters

Novel tools and techniques are provided for implementing management of routing across multiple voice or data networks with separate routing masters. In various embodiments, in response to receiving a request to establish a call between a calling party in a first network and a called party in a second network, a computing system might receive a first set of network information from a first routing database(s) that is operated by a first service provider and a second set of network information from a second routing database(s) that is operated by a second service provider separate from the first service provider; might analyze the received first and second sets of network information to generate a unified routing model for optimizing routing of the call through the first and second networks; and might establish the call through a selected optimized route based on the generated unified routing model. 1. A method , comprising:receiving, with a computing system from a calling party at an originating point in at least one first network, a request to establish a call between the calling party and a called party at a destination point in at least one second network;receiving, with the computing system, a first set of network information from one or more first routing databases associated with the at least one first network that is operated by a first service provider;receiving, with the computing system, a second set of network information from one or more second routing databases associated with the at least one second network that is operated by a second service provider that is separate from the first service provider;analyzing, with the computing system, the received first set of network information and the received second set of network information to generate a unified routing model for optimizing routing of the call between the calling party and the called party through the at least one first network and the at least one second network;selecting, with the computing ...

Delivering a toll-free caller id on toll-free calls

A computer device may include logic configured to receive a Session Initiation Protocol (SIP) telephone call and identify the SIP telephone call as originating from a first toll-free telephone number and destined for a second toll-free telephone number. The computer device may further determine a screened telephone number associated with the first toll-free telephone number; generate a diversion header that includes the determined screened telephone number for the SIP telephone call, in response to identifying the SIP telephone call as originating from the first toll-free telephone number and destined for the second toll-free telephone number; and send the SIP telephone call to a local exchange carrier network associated with the second toll-free telephone number.

METHOD FOR CONTROLLING BITRATE IN REALTIME AND ELECTRONIC DEVICE THEREOF

Disclosed is an electronic device including a sound receiving circuit, a communication circuit, a processor, and a memory. The memory may store one or more instructions that, when executed, cause the processor to control the electronic device to: make a packet based call with an external electronic device using the communication circuit, to obtain communication environment information of the electronic device, to determine a change in a bitrate of the packet based call based on the obtained communication environment information, to identify whether a voice is received through the sound receiving circuit, and to change the bitrate of the packet based call from a first bitrate to a second bitrate based on a silent section in which the voice is not received through the sound receiving circuit being identified. 1. An electronic device comprising:a sound receiving circuit;a communication circuit;a processor operatively connected to the sound receiving circuit and the communication circuit; anda memory operatively connected to the processor,wherein the memory stores one or more instructions that, when executed, cause the processor to control the electronic device to:make a packet based call to an external electronic device using the communication circuit;obtain communication environment information of the electronic device using the communication circuit;determine a change in a bitrate of the packet based call based on the obtained communication environment information;identify whether a voice is received through the sound receiving circuit in response to the determination of the change in the bitrate; andchange the bitrate of the packet based call from a first bitrate to a second bitrate based on a silent section in which the voice is not received through the sound receiving circuit being identified.2. The electronic device of claim 1 , wherein the one or more instructions claim 1 , when executed claim 1 , cause the processor to control the electronic device to:encode a ...

Telephone system, exchange, telephone exchanging method, and telephone exchanging program

A system includes: a first telephone; a relay that manages first accommodation position information of the first telephone; and an exchange that includes a management table that manages second telephone identification information, second accommodation position information of a second telephone connected to the relay, and an extension number of each of a plurality of the second telephones, receives first telephone identification information and the first accommodation position information, when the first telephone identification information does not match with any one of the second telephone identification information registered, and the first accommodation position information matches with one of the second accommodation position information in the management table, changes second telephone identification information corresponding to the matched second accommodation position information to the first telephone identification information, and uses an extension number corresponding to the matched second accommodation position information as an extension number of the first telephone.

DETECTING AND REPORTING USER TRIGGERED CALL DROPS

A method for detecting user triggered call drops includes identifying one or more user terminated calls from a plurality of monitored calls. Signaling information associated with the identified user terminated calls is correlated with media channel information associated with the identified user terminated calls. A determination is made if termination of the one or more of the identified user terminated calls is related to quality of media across corresponding media channels. A predefined cause code is assigned to the one or more of the identified user terminated calls, in response to determining that the termination of the one or more of the identified user terminated calls is related to the quality of media across the corresponding media channels. 1. A method for detecting user triggered call drops , the method comprising:identifying one or more user terminated calls from a plurality of monitored calls;correlating signaling information associated with the identified user terminated calls with media channel information associated with the identified user terminated calls;determining if termination of the one or more of the identified user terminated calls is related to quality of media across corresponding media channels; andassigning a predefined cause code to the one or more of the identified terminated calls, in response to determining that the termination of the one or more of the user identified terminated calls is related to the quality of media across the corresponding media channels.2. The method of claim 1 , wherein the plurality of monitored calls comprises a plurality of Voice over IP (VOIP) calls.3. The method of claim 1 , wherein determining if termination of the one or more of the identified terminated calls is related to quality of media across corresponding media channels further comprises determining media quality of one or more uplink media channels and determining media quality of one or more downlink media channels.4. The method of claim 1 , ...

METHOD FOR REGISTERING MULTI-CONTACT DEVICES

A technique is described allowing an IMS enabled network to terminate a communication request directed towards a user device in a well-defined way, even for cases in which multiple registrations for IMS are present that are related to the same physical user device. A method embodiment of this technique comprises registering multiple user agents for one and the same user device, wherein from each user agent contact information including a device ID identifying the user device is provided; and processing a communication request directed towards the user device, the processing comprising identifying multiple identical device IDs of registered contacts, and treating the communication request according to a predetermined handling scheme related to the detection of identical device IDs. 1. A method of registering in an IP multimedia Subsystem (IMS)-enabled network , two or more user agents associated with a multi-contact user device , the method comprising the steps of:associating with each of the two or more user agents, a unique user device identification (ID) that is unique to the multi-contact user device, wherein the unique user device ID is derived from a global permanent identifier associated with the multi-contact user device; andproviding, by each of the two or more user agents, the device ID to the IMS-enabled network.2. The method according to claim 1 , wherein at least one of the user agents is located in the user device.3. The method according to claim 1 , wherein at least one of the user agents is located in the IMS-enabled network.4. The method according to claim 3 , wherein the at least one user agent located in the IMS-enabled network acts for a Circuit-Switched (CS) client in the multi-contact user device.5. The method according to claim 1 , wherein the user agents additionally provide a predetermined user capability value suitable for delivery of Circuit-Switched (CS) services via the IMS-enabled network.6. A method of controlling a communication ...

SUPPORTING HITLESS UPGRADE OF CALL PROCESSING NODES IN CLOUD-HOSTED TELEPHONY SYSTEM

A method is provided in which a call agent process that supports one or more Internet Protocol (IP) calls, stores to persistent memory a set of data associated with the one or more IP calls. An outage is detected affecting the one or more IP calls. Using the data retrieved from the persistent memory, the one or more IP calls are resynthesized using a device simulator process to simulate connectivity with endpoints that were participating in the one or more IP calls prior to the outage. After resynthesizing, depending on activity detected from devices associated with the one or more IP calls, the one or more IP calls are internally re-stitched/re-establishing (without signaling to endpoints) with the endpoints involved in the one or more IP calls, or the one or more calls are fully re-stitched/re-established by signaling an endpoint that was participating in the one or more IP calls. 1. A method comprising:at a call agent process running on a server to support one or more Internet Protocol (IP) calls, storing to persistent memory a set of data associated with the one or more IP calls;detecting an outage affecting the one or more IP calls;using the data retrieved from the persistent memory, resynthesizing the one or more IP calls using a device simulator process to simulate connectivity with endpoints that were participating in the one or more IP calls prior to the outage; and internally re-establishing the one or more IP calls with the endpoints involved in the one or more IP calls; or', 'fully re-establishing the one or more IP calls by signaling an endpoint that was participating in the one or more IP calls., 'after resynthesizing, depending on activity detected from devices associated with the one or more IP calls2. The method of claim 1 , wherein detecting comprises detecting an unplanned outage or a planned outage associated with an upgrade of the call agent process.3. The method of claim 1 , wherein internally re-establishing the one or more IP calls is ...

INTERNET PROTOCOL (IP) SERVERLESS PAGE PARTY (SPP) STATION AND SYSTEMS AND METHODS FOR DEPLOYING MULTIPLE SPP STATIONS

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station. 1. A page party station that provides page announcing and party line conferencing with respect to other page party stations , the page party station connected to other page party stations in a network that supports multicast and point-to-point communication , the page party station comprising:a network interface to connect the page party station to the network;a plurality of configurable multicast addresses to transmit and receive audio in party line conferencing with other page party stations;a plurality of configurable multicast addresses to transmit announcements to other page party stations;a party line selector to select one of the multicast addresses for party line conferencing;a page line selector to select one of the multicast addresses for page announcing; anda processing device that operates in a mutual configuration mode to configure the station to provide page announcing and party line conferencing operations using voice over Internet Protocol (VoIP) on the network, ...

Method and system for connecting between terminals in multimedia communication

Disclosed is a terminal connection method for multimedia communication. The terminal connection method may reduce or minimize the use of an access network by bridging a connection between access networks of two terminals through an internal application server (iAS).

GATEWAY AND METHOD FOR MANAGING VOIP TELEPHONE SERVICE

The method performed by a gateway between a LAN and a WAN that includes a platform of the VoIP telephone service, is characterized in that it includes a step of activating the VoIP telephone service on detecting a connected state of a terminal connected to the gateway. 1. A method of managing a VoIP telephone service performed by a gateway between a LAN and a WAN that includes a platform of the VoIP telephone service , the method including activating the VoIP telephone service on detecting a connected state of a terminal connected to the gateway.2. The method of claim 1 , wherein the connected state of the terminal is detected when an off-hook state of the terminal is detected.3. The method of claim 1 , wherein activating uthe VoIP telephone service depends on detecting both the connected state and the off-hook state of the terminal.4. The method of claim 1 , further including deactivating the VoIP telephone service on detecting that a countdown timer initialized on the activation has expired.5. The method of claim 4 , further including reinitializing the countdown on detecting a hung-up state following detecting the connected and off-hook states of the terminal.6. (canceled)7. A non-transitory computer readable medium having stored thereon instructions which claim 1 , when executed by a processor claim 1 , cause the processor to implement the method of .8. A gateway between a LAN and a WAN including a VoIP telephone service platform claim 1 , the gateway comprising:at least one detector configured to detect a connected state of a terminal connected to the gateway; anda manager of the VoIP telephone service, configured to activate the VoIP telephone service on detecting the connected state.9. The gateway of claim 8 , wherein the detector is configured to detect an off-hook state of the terminal connected to the gateway.10. The gateway according to of claim 9 , wherein the manager of the VoIP telephone service is configured to activate the VoIP telephone service on ...

CALL CONNECTING METHOD AND SYSTEM BASED ON MOBILE-VoIP

본 발명은 모바일 인터넷 전화(mVoIP) 기반의 호 처리 시스템 및 방법에 대한 것이다. 가입자서버와 호 처리장치를 포함하는 mVoIP(Mobile-Voice over Internet Protocol) 호 연결 시스템에서 수행되는 mVoIP 호 연결 방법에 있어서, 가입자서버에서 제1 사용자 단말로부터 제2 사용자 단말로의 호 연결 요청을 수신하고, 제2 사용자 단말로 호 연결 요청을 알리는 단계; 호 처리장치가 제1 사용자 단말로부터 호 시도 메시지를 수신하는 단계; 호 처리장치가 제2 사용자 단말로부터 호 수신 준비를 알리는 호 대기 알림을 수신하는 단계; 및 호 처리장치가 제2 사용자 단말로 호 시도 메시지를 전달하여 제1 사용자 단말과 제2 사용자 단말을 호 연결하는 단계를 포함하는 호 연결 방법이 제공될 수 있다.

VoIP terminal having a function of credit card settlement, settlement processing system using a VoIP terminal and settlement processing method using a VoIP terminal

본 발명은 카드 결제 기능을 구비한 VoIP 단말, VoIP 단말을 이용한 결제 처리 시스템 및 VoIP 단말을 이용한 결제 처리 방법에 관한 것으로, ARS 서버가 앱 실행 이벤트 코드와, 상품 코드를 포함하는 결제 요청 메시지를 VoIP 단말로 전송하고, VoIP 단말은 ARS 서버로부터 전송된 결제 요청 메시지에 포함된 앱 실행 이벤트 코드를 감지하여 결제용 앱을 실행하고, 결제용 앱을 통해 결제 요청 메시지에 포함된 상품 코드에 대응하는 결제 인터페이스를 결제 대행 서버에 요청하여 결제를 수행함으로써 홈쇼핑 방송 채널을 통해 방송되는 상품에 대한 ARS 주문 결제시 사용자 조작을 최소화할 수 있도록 한 것이다. The present invention relates to a VoIP terminal having a card payment function, a payment processing system using a VoIP terminal, and a payment processing method using a VoIP terminal. The ARS server transmits a payment request message including an application execution event code and a product code to VoIP The VoIP terminal detects an application execution event code included in the payment request message transmitted from the ARS server, executes the payment application, and transmits a payment corresponding to the product code included in the payment request message through the payment application Interface to the settlement agent server to perform settlement, thereby minimizing the user's manipulation during ARS order settlement for a product broadcast through a home shopping broadcast channel.

Method and apparatus for transmitting/receiving for packet decoding information of packet service in mobile telecommunication system

본 발명은 이동통신 시스템에서 패킷 서비스 제공 시 단말에게 패킷 디코딩 정보를 효율적으로 시그날링 하는 방법 및 장치에 관한 것이다. 본 발명에 따른 패킷 디코딩 정보의 송신 방법은, 전송하고자 하는 패킷의 패킷 디코딩 정보를 결정하고 상기 결정된 패킷 디코딩 정보를 설정된 기본 패킷 디코딩 정보와 비교하는 과정과, 상기 결정된 패킷 디코딩 정보가 상기 기본 패킷 디코딩 정보와 동일하면, 상기 결정된 패킷 디코딩 정보를 전송하지 않고 상기 전송하고자 하는 패킷을 설정된 기본 전송 자원을 사용하여 공용 채널을 통해 전송하는 과정과, 상기 결정된 패킷 디코딩 정보가 상기 기본 패킷 디코딩 정보와 상이하면, 상기 전송하고자 하는 패킷과 상기 결정된 패킷 디코딩 정보를 함께, 상기 전송하고자 하는 패킷을 위해 할당된 전송 자원을 사용하여 상기 공용 채널을 통해 전송하는 과정을 포함한다. 이러한 본 발명은 패킷 디코딩 정보의 전송으로 인한 시그널링 오버헤드를 감소시킨다. Packet Decoding Information, Grant Channel, VoIP

Telephone outlet with packet telephony adapter, and a network using same

지역 네트워크(80; LAN)용 인출구는 아날로그 전화 통신으로 그리고 이로부터 VoIP를 변환시키는 통합 어댑터(21)와 통상의 아날로그(POTS) 전화 세트에 연결하기 위한 표준 전화 잭(72; 예로써, 북아메리카에서 RJ 11)을 포함한다. 이러한 인출구는 VoIP 환경에서의 아날로그 전화 세트를 사용하는 것을 허용하여 IP 전화 세트 또는 외부 어댑터에 대한 필요성을 제거한다. 상기 인출구는 어댑터(21a, 21c)를 통해 아날로그 전화 세트(22a, 22c) 양자 모두에 연결하고 네트워크 잭(73a, 73c)에 의해 어세스될 수 있는 데이터 네트워크 연결을 유지하는 것을 허용하는 허브(31a, 31c)를 포함할 수 있다. 본 발명은 전화 라인 기반 데이터 네트워킹 시스템에 적용될 수 있다. 이러한 환경에서, 데이터 네트워킹 회로와 VoIP/POTS 어댑터는 전화 인출구로 통합되어 아날로그 전화 세트를 사용하여 통상의 아날로그 서비스와 VoIP 전화 통신 서비스 및 데이터 네트워킹을 제공하다. 이러한 환경에서, 인출구는 두 개의 표준 전화 잭과 데이터 네트워킹 잭을 필요로한다. 본 발명에 따른 인출구는 새로운 설비뿐만 아니라 현존하는 LAN과 건물의 전화선을 개장하는데 사용될 수 있다. 인출구, 커넥터, 로우 패스 필터, 하이 패스 필터, 허브 The outlet for the local area network 80 (LAN) is an integrated adapter 21 for converting VoIP to and from analog telephony and a standard telephone jack 72 for connecting to a conventional analog (POTS) telephone set (e.g., in North America). RJ 11). This outlet allows the use of analog telephone sets in a VoIP environment, eliminating the need for IP telephone sets or external adapters. The outlet is connected to both analog telephone sets 22a and 22c via adapters 21a and 21c and to a hub 31a allowing to maintain a data network connection that can be accessed by network jacks 73a and 73c. , 31c). The present invention can be applied to a telephone line based data networking system. In this environment, data networking circuitry and VoIP / POTS adapters are integrated into the telephone outlet to provide conventional analog services, VoIP telephony services, and data networking using analog telephone sets. In this environment, the outlet requires two standard telephone jacks and a data networking jack. The outlet according to the invention can be used to retrofit existing LANs and telephone lines of buildings as well as new installations. Outlet, Connector, Low Pass Filter, High Pass Filter, Hub

Call mode control method and device, storage medium and electronic equipment

本公开提供了一种通话方式控制方法、通话方式控制装置、存储介质和电子设备，涉及终端控制技术领域。该通话方式控制方法包括：获取WiFi可用时长；其中，WiFi可用时长为终端设备持续连接WiFi热点的时间上限；如果WiFi可用时长大于第一时间阈值，则将通话方式配置为VoWiFi通话方式；监测WiFi剩余时长，如果WiFi剩余时长小于第二时间阈值，则将通话方式由VoWiFi通话方式切换为VoLTE通话方式。本公开可以避免出现通话中断的问题。

Method and apparatus for reconnecting voice call at picture communication of dual mode mobile communication set

A method and an apparatus for re-connecting a voice call during video call communication in a dual mode mobile terminal are provided to enable a dual mode mobile terminal to minimize a waste of time and provide a basic call service without user's manipulation in a WCDMA shadow area. A video call communication unit(110) performs video call communication. A voice call communication unit(120) performs voice call communication. A comparing unit(130) compares a BLER(Block Error Ratio) with respect to the video call communication with the first or second reference value. When the BLER with respect to the video call communication is higher than the first reference value, a controller(140) controls the voice call communication unit(120) to be ready, and when the BLER with respect to the video call communication is higher than the second reference value, the controller(140) controls the video call communication unit(110) to terminate the video call communication and controls the voice call communication unit(120) to re-originate a voice call.

Debugging apparatus and method of the distributed private exchange box system

본원 발명은 분산 사설교환기 시스템에 관한 것으로서, 더욱 상세하게는 분산 사설교환기 시스템의 디버깅 성능을 개선 시킬 수 있도록 하는 분산 사설교환기 시스템의 디버깅 장치 및 그 방법에 관한 것이다. The present invention relates to a distributed private exchange system, and more particularly, to a debugging apparatus and a method of a distributed private exchange system to improve the debugging performance of the distributed private exchange system. 상술한 본원 발명의 분산 사설교환기 시스템의 디버깅 장치는, 음성 통신망에 연결된 단말 장치로부터 요청되는 음성 통신 서비스를 제공하기 위하여 프로그램과 데이터를 관리하고 로컬사설교환기들의 디버깅 정보를 수집하여 저장하는 중앙교환기와; 상기 중앙교환기로 네트워크 통신을 통해 접속하여 디버깅 정보를 출력하고 상기 중앙교환기를 통해 분산 사설교환기 시스템을 제어하는 유지보수장치;를 포함하여 이루어지는 것을 특징으로 한다. The above-described debugging device of the distributed private exchange system of the present invention includes a central exchanger that manages programs and data and collects and stores debugging information of local private exchanges in order to provide a voice communication service requested from a terminal device connected to a voice communication network. ; And a maintenance apparatus for accessing the central exchange through network communication to output debugging information and controlling a distributed private exchange system through the central exchange. 분산 사설교환기시스템, 디버깅, 다중링크접속, 텔넷, 네트워크 Distributed Private Switching System, Debugging, Multiple Link Access, Telnet, Network

IP phone cruising inspection system, method and server based on Session Initiation Protocol

本发明公开了基于SIP协议的IP电话巡检系统、方法及服务器，包括IP电话巡检服务器，一端与IMS网络所在核心交换机及IMS网络各网元相连，另一端与需要查看巡检结果的管理计算机相连；IP电话巡检服务器采用SIP中继与IMS核心网通信，通过该中继构建IP话机巡检消息检测IP话机工作状态的方法，通过向被检测IP话机发送检测指令，通过对端的回应消息判定对端的状态；IP电话巡检服务器采用内部数据接口调用方法，通过本地数据库巡检任务查询请求，IP电话巡检服务器将符合条件的查询巡检任务返回到巡检模块，由巡检模块进行巡检，当IP电话巡检服务器查询到巡检结论后，传输至故障查询计算机进行显示巡检结论信息。

Apparatus and method for automatic repeat request with reduced resource allocation overhead in a wireless VoIP communication system

A base station ( 103 ) assigns a set of mobile stations ( 101 ) to a group wherein the group will share a set of radio resources ( 770 ). A shared control channel information element ( 501 ) is sent to the group of mobile stations ( 101 ) and provides a bitmap having fields for group ordering ( 511 ), resource allocations ( 530 ), continuation resources ( 540 ) for HARQ, and an ordering pattern ( 513 ). If a mobile station requires retransmission it will access the resources indicated by the continuation resources field ( 54 ) in order to receive data. The HARQ blocks may be assigned to a mobile station based upon an index ( 601 ) which may correspond to the mobile station vocoder rate. Further, HARQ subgroups may be defined to associate subgroups of mobile stations with specific HARQ transmission opportunities on the super-frame and allocated by a rotating bitmap.

Apparatus and method for providing quality of service to a wireless access point accomodating session initiaition protocol(sip) based multimedia over internet protocol(moip) service

A device for supplying QoS(Quality of Service) to a wireless AP(Access Point) which receives an SIP(Session Initiation Protocol)-based MoIP(Multimedia over Internet Protocol) service and a method therefor are provided to optimize transmission rate control of a physical layer by considering session characteristics, thereby increasing the number of subscribers who receive a service as well as MoIP service quality control at a wireless AP. A receiver interface(202) receives an SIP/SDP(Service Discovery Protocol) message passing through an AP(Access Point). An SIP signal analyzer(204) analyzes a protocol of the received SIP/SDP message, and detects RTP(Real-time Transport Protocol) flow information of the AP. A QoS manager(206) decides whether an SIP-based RTP flow is accepted, by using the RTP flow information and the currently received RTP flow information, stores information on the accepted RTP flow, controls MAC(Medium Access Control) parameters based on the decided results and the received RTP flow information, and converts acceptance control results for an SIP call request into an SIP message type.

Social network virtual private network

A system and method create an interconnection among users of different social networks. Each user can provide information about the social networks they use (e.g., username, passwords, etc.). An application may register with the social network and can inter- dispose the application between the user and the social network. Then, the system can interconnect the users into a "virtual private network" (VPN) where communications between the users and the social networks can be replicated to other social networks for the other users. As such, different users using different social networks can communicate.

Communication system and method

A method of controlling a connection between a user terminal and an access node connected to a communication network is provided. The user terminal establishes a data connection with the access node, periodically generates a message at predetermined intervals and transmits the periodic message to at least one network node via the access node over the communication network. Responses to the periodic messages are received from the at least one network node. The responses are analyzed to determine whether to terminate the connection to the access node, and in the case that the connection to the access node should be terminated, a disconnect message is transmitted to the access node from the user terminal.

Voip verifier

Audio data coding method, device, equipment and storage medium

本发明实施例公开了一种音频数据的编码方法、装置、设备和存储介质，所述方法包括：获取音频数据传输网络的网络带宽值；确定所述音频数据当前设置的原始语音编码码率和原始包头编码码率；根据所述网络带宽值分别对所述原始语音编码码率和所述包头编码码率进行调整，得到目标语音编码码率和目标包头编码码率；采用所述目标语音编码码率和所述目标包头编码码率对所述音频数据进行编码，得到目标音频数据。本发明实施例根据网络的网络带宽值分别对原始语音编码码率和包头编码码率进行调整，在低网络带宽下可以降低包头编码码率以降低总编码码率，以使得总编码码率适配低网络带宽，保证了低网络带宽下的通话质量。

Telephone outlet with packet telephony adapter, and a network using same

An outlet (75) for a Local Area Network (80) (LAN), containing an integrated adapter (21) that converts VoIP to and from analog telephony, and a standard telephone jack (72) (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub (31a, 31c) that allows connecting both an analog telephone set (22a, 22c) via an adapter (21a, 21c), as well as retaining the data network connection, which may be accessed by a network jack (73a, 73c). The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation.

Method and device for redirecting a communication request

Procédé et dispositif de redirection d'une requête de communication L'invention concerne un procédé de redirection d'une requête de communication émise par un terminal (T1) dit terminal appelant, à destination d'un terminal dit terminal appelé, un identifiant du terminal appelant étant absent d'une liste d'identifiants de terminaux appelants pour lesquels l'établissement d'une communication avec le terminal appelé est autorisé ou interdit, respectivement dite liste blanche ou liste noire, le procédé comprenant : la redirection (24) de la requête de communication vers un terminal (T22) associé au terminal appelé, distinct du terminal appelé, dit terminal de redirection, la réception (25) d'un message d'instruction, comprenant l'identifiant du terminal appelant et une instruction relative à la liste blanche ou à la liste noire, en fonction de l'instruction reçue, l'ajout (26) de l'identifiant du terminal appelant à la liste blanche ou à la liste noire. Figure pour l'abrégé: Figure 2 Method and device for redirecting a communication request The invention relates to a method of redirecting a communication request sent by a terminal (T1) called calling terminal, to a terminal called called terminal, an identifier of the calling terminal being absent from a list of identifiers of the calling terminal. calling terminals for which the establishment of a communication with the called terminal is authorized or prohibited, respectively called a white list or black list, the method comprising: the redirection (24) of the communication request to a terminal (T22) associated with the called terminal, distinct from the called terminal, called a redirection terminal, the reception (25) of an instruction message, comprising the identifier of the calling terminal and an instruction relating to the white list or to the black list, depending on the instruction received, adding (26) the identifier of the calling terminal to the ...

MULTIPLE USER IDENTITY AND BRIDGING PRESENTATION

Es wird ein Anrufsteuerungssystem bereitgestellt, das den Standort einer Person feststellen kann. Von dem Standort der Person aus kann das Anrufsteuerungssystem zeitweilig ein anderes Telefon mit der Person verknüpfen. Es kann eine Brückendarstellung verändert werden, so dass sie die zeitweilige Verknüpfung einschließt derart, dass Anrufe von dem anderen Telefon die Person als einen Teilnehmer an dem Anruf zeigen. Ferner kann ein Leitweglenkungsalgorithmus für das Leiten von Anrufen zu Telefonen innerhalb des Unternehmens dann zeitweilig so modifiziert werden, dass Anrufe zu dem anderen Telefon für die Person geleitet werden. Folglich kann sich eine Person durch die Einrichtungen des Unternehmens bewegen und wichtige Anrufe zu der Person geleitet bekommen, wo immer sie sich befinden mag. A call control system is provided which can determine the location of a person. From the person's location, the call control system may temporarily associate another telephone with the person. A bridge representation may be altered to include the temporary association such that calls from the other telephone show the person as a participant in the call. Further, a routing algorithm for routing calls to telephones within the enterprise may then be temporarily modified to route calls to the other telephone for the person. As a result, a person can move through the company's facilities and receive important calls to the person wherever they are.

A kind of cloud telephone network communication system and communication modes

本发明提供一种云电话网络通讯系统及通讯方式，包括用于接入电话或呼出电话的电话转接盒子终端、用于与电话转接盒子终端进行网络通讯连接的服务器终端及手机移动终端应用；在电话转接盒子终端内部设置有用于进行呼入呼出控制、号码读取输出控制及网络传输控制的语音接入输出控制单元及PCM语音与VOIP互转单元；还包括设置于电话转接盒子终端内部、用于插入SIM卡的多个SIM卡卡槽及用于进行连接的网络接口，实际应用过程中，通过建立电话转接盒子终端、服务器终端以及手机移动终端应用之间的通讯连接，即可通过数据格式的转换进行双向通话，且实现有效降低漫游费，无任何漫游费用，只需要支付当地网络的数据费用，即可实现漫游等诸多优势，使用效果突出。

Telephone outlet with packet telephony adapter, and a network using same

지역 네트워크(80; LAN)용 인출구는 아날로그 전화 통신으로 그리고 이로부터 VoIP를 변환시키는 통합 어댑터(21)와 통상의 아날로그(POTS) 전화 세트에 연결하기 위한 표준 전화 잭(72; 예로써, 북아메리카에서 RJ 11)을 포함한다. 이러한 인출구는 VoIP 환경에서의 아날로그 전화 세트를 사용하는 것을 허용하여 IP 전화 세트 또는 외부 어댑터에 대한 필요성을 제거한다. 상기 인출구는 어댑터(21a, 21c)를 통해 아날로그 전화 세트(22a, 22c) 양자 모두에 연결하고 네트워크 잭(73a, 73c)에 의해 어세스될 수 있는 데이터 네트워크 연결을 유지하는 것을 허용하는 허브(31a, 31c)를 포함할 수 있다. 본 발명은 전화 라인 기반 데이터 네트워킹 시스템에 적용될 수 있다. 이러한 환경에서, 데이터 네트워킹 회로와 VoIP/POTS 어댑터는 전화 인출구로 통합되어 아날로그 전화 세트를 사용하여 통상의 아날로그 서비스와 VoIP 전화 통신 서비스 및 데이터 네트워킹을 제공하다. 이러한 환경에서, 인출구는 두 개의 표준 전화 잭과 데이터 네트워킹 잭을 필요로한다. 본 발명에 따른 인출구는 새로운 설비뿐만 아니라 현존하는 LAN과 건물의 전화선을 개장하는데 사용될 수 있다. 인출구, 커넥터, 로우 패스 필터, 하이 패스 필터, 허브

A kind of voip communicationses method and system

本发明实施例公开了一种网络电话通信方法及系统，其中方法包括：客户端若检测到用户的网络电话通信请求，则向服务端发送网络电话通道请求消息；所述服务端根据所述网络电话通道请求消息，从当前可用的网络电话供应商中选择通话质量满足预设要求的目标网络电话供应商；所述服务端将所述目标网络电话供应商的信息发送给所述客户端；所述客户端根据所述目标网络电话供应商的信息发起网络电话呼叫。本发明实施例通过在当前可用的网络电话供应商中选择通话质量满足预设要求的目标网络电话供应商进行网络电话通话，使得网络电话应用不再受限于单个网络电话供应商部署的网络电话服务器的处理性能，从而提高网络电话的通话质量。

Method and apparatus for controlling call volume

A method and apparatus for controlling a call volume for an office that serves as a protecting site for another office in a packet network are disclosed. For example, the method collects one or more customer registration counts from one or more session border controllers located in a first office, determines if the one or more customer registration counts have reached or exceeded a threshold. The method directs all of said one or more session border controllers located in said first office to enact one or more throttling rules if the one or more customer registration counts have reached or exceeded the threshold.

Communication management platform for prison family telephone integration

本发明公开了一种监狱亲情电话融合通讯管理平台，包括用户操作功能模块以及平台底层音频处理传输控制模块，用户操作功能模块包括亲情电话拨打子系统、亲情电话控制子系统、亲情电话分析子系统和平台控制子系统，平台底层音频处理传输控制模块包括录音单元、提示音播放单元、DTMF拨号单元和混音桥单元。把犯人、犯人亲友、监管人员三方汇接在一起，解决了亲情电话系统和监仓可视对讲系统间不能互通的问题。

Communication system and method

A method of controlling a connection between a user terminal and an access node connected to a communication network is provided. The user terminal establishes a data connection with the access node, periodically generates a message at predetermined intervals and transmits the periodic message to at least one network node via the access node over the communication network. Responses to the periodic messages are received from the at least one network node. The responses are analysed to determine whether to terminate the connection to the access node, and in the case that the connection to the access node should be terminated, a disconnect message is transmitted to the access node from the user terminal.

Voice call method and device and electronic equipment

本申请公开了一种语音通话方法、装置和电子设备，属于通信技术领域。其中，语音通话方法包括：根据当前语音通话所处的第一网络的当前网络参数信息，确定所述第一网络的当前网络环境等级信息；根据所述当前网络环境等级信息，为所述当前语音通话分配带宽；其中，所述当前网络参数信息包括当前网速信息；所述带宽为所述第一网络的带宽或者第二网络的带宽，所述第一网络与所述第二网络为不同的网络。本方案能够实现基于网络带宽分配进行语音通话，从而避免进行语音通话时由于网速或网络资源分配等因素影响导致通话质量降低，保障通话的质量和连续性，增加用户体验，很好的解决了现有的VoWiFi通话方案通话质量不稳定的问题。

Camera-based facial recognition or other single/multiparty presence detection as a method of effecting telecom device alerting

A camera can be associated with each conference participant endpoint. The camera, either frame or video-based, can monitor and detect one or more of gestures, facial recognition, emotions, and movements of the conference participant. Based on the detection of one or more of these triggering events, a correlation to an action corresponding the triggering event can be evoked. For example, if a participant raises their hand, e.g., a triggering event, the system can recognize that this is a request to speak. The participant can then be queued in the system based, for example, relative to other participants' requests. When the other participants have finished speaking, and it is the time for the user who raised their hand to speak, the system can optionally queue the user by modifying the endpoint with which they are associated.

handset and method for requesting auto repeat with reduced resource allocation overhead in a wireless voice over ip (voip) communication system

APARELHO E MÉTODO PARA SOLICITAÇçO DE REPETIÇçO AUTOMÁTICA COM OVERHEAD DE ALOCAÇçO DE RECURSOS REDUZIDO, EM UM SISTEMA DE COMUNICAÇçO DE COMUNICAÇçO DE VOZ SOBRE IP (VOIP) SEM FIO Uma estação base (103) atribui um conjunto de estações móveis (101) a um grupo em que o grupo compartilhará um conjunto de recursos de rádio (770). Um elemento de informação de canal de controle compartilhado (501) é enviado para o grupo de estações móveis (101) e provê um mapa de bits tendo campos para ordenamento de grupo (511), alocações de recursos (530) , recursos de continuação (540) para HARQ, e um padrão de ordenamento (513). Se uma estação móvel exigir retransmissão ela acessará os recursos indicados pelo campo de recursos de continuação (540) para receber os dados. Os blocos HARQ podem ser atribuídos a uma estação móvel com base em um índice (601) o qual pode corresponder à taxa de codificador de voz de estação móvel. Adicionalmente, subgrupos HARQ podem ser definidos para associar subgrupos de estações móveis com oportunidades de transmissão HARQ específicas no superquadro e alocados por intermédio de um mapa de bits rotativo. APPLIANCE AND METHOD FOR AUTOMATIC REPEAT REQUEST WITH REDUCED RESOURCE ALLOCATION IN A WIRELESS VOICE COMMUNICATION COMMUNICATION SYSTEM A base station (103) assigns a set of mobile stations (101) to a group of mobile stations where the group will share a radio feature set (770). A shared control channel information element (501) is sent to the mobile station group (101) and provides a bitmap having fields for group ordering (511), resource allocations (530), continuation resources ( 540) for HARQ, and an ordering pattern (513). If a mobile station requires relaying it will access the resources indicated by the continuation resource field (540) to receive the data. HARQ blocks may be assigned to a mobile station based on an index (601) which may correspond to the mobile station voice encoder rate. In addition, HARQ subgroups may be defined to associate ...

EQUIPMENT AND METHOD FOR REQUESTING AUTOMATIC REPEAT WITH REDUCED RESOURCE ALLOCATION OVERHEAD IN A WIRELESS VOIP COMMUNICATION SYSTEM

aparelho e método para solicitação de repetição automática com overhead de alocação de recursos reduzido, em um sistema de comunicação de comunicação de voz sobre ip (voip) sem fio uma estação base (103) atribui um conjunto de estações móveis (101) a um grupo em que o grupo compartilhará um conjunto de recursos de rádio (770). um elemento de informação de canal de controle compartilhado (501) é enviado para o grupo de estações móveis (101) e provê um mapa de bits tendo campos para ordenamento de grupo (511), alocações de recursos (530) , recursos de continuação (540) para harq, e um padrão de ordenamento (513). se uma estação móvel exigir retransmissão ela acessará os recursos indicados pelo campo de recursos de continuação (540) para receber os dados. os blocos harq podem ser atribuídos a uma estação móvel com base em um índice (601) o qual pode corresponder à taxa de codificador de voz de estação móvel. adicionalmente, subgrupos harq podem ser definidos para associar subgrupos de estações móveis com oportunidades de transmissão harq específicas no superquadro e alocados por intermédio de um mapa de bits rotativo. apparatus and method for requesting automatic repetition with reduced resource allocation overhead, in a wireless voice over ip (voip) communication system a base station (103) assigns a set of mobile stations (101) to a group where the group will share a set of radio resources (770). a shared control channel information element (501) is sent to the group of mobile stations (101) and provides a bitmap having fields for group ordering (511), resource allocations (530), continuation resources ( 540) for harq, and an ordering pattern (513). if a mobile station requires retransmission it will access the resources indicated by the continuation resource field (540) to receive the data. harq blocks can be assigned to a mobile station based on an index (601) which can correspond to the mobile station's voice encoder rate. in addition, harq subgroups can be ...

Processing sms messages

There is provided a method for processing communications in a telephony network. The method includes configuring a first telephony identifier for communication with a first user and a second user, and a second telephony identity for communication with the second user, each of the telephony identifiers being configured for use as different destination addresses in different SMS messages sent to the second user. The method includes processing at least one SMS message in SMS processing equipment in the network, and during this processing, performing identity management, in relation to the use of the first and second telephony identifiers for the second user.

Voice communication method and system

本发明公开了一种语音通话方法及系统：所述方法包括以下内容：注册请求步骤，将各智能家居设备在语音插件上进行注册；心跳连接步骤，所述智能家居设备向所述语音插件发送心跳请求建立连接；语音通话建立步骤，所述智能家居设备根据不同呼叫模式与外部终端建立语音通话。利用光猫内部本身提供的语音模块，通过在光猫中另外加一个语音插件，将传统的协议包装成专门的协议提供给其它网络终端来使用，避免将电信运营商的核心系统暴露到公共环境中。光猫上的语音模块本身就是为固定电话提供通话能力的，通话质量是受运营商保证的，因此，在加入语音插件后语音通话质量区别传统的软电话质量更加可靠。

Telephone outlet with packet telephony adaptor, and a network using same

An outlet for a Local Area Network (LAN), containing an integrated adapter that converts VoIP to and from analog telephony, and a standard telephone jack (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub that allows connecting both an analog telephone set via an adapter, as well as retaining the data network connection, which may be accessed by a network jack. The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP/POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation.

IP telephone system

An IP telephone system equipped with a plurality of telephone terminals and a file server, wherein the file server comprises notification unit notifies an occurrence of update data of the software to the telephone terminals and transfer unit transfers the update data to a telephone terminal of a request source in response to a request from the update data from the telephone terminal, and the telephone terminal comprises reporting unit reports a fact of such to user when the occurrence of the update data is notified from the file server, a input unit used for inputting the update timing of the software by the user reported, request unit requests a transfer of the update data to the file server at the timing, and software update unit switches old software to new software based on the update data transferred from the file server in response to the request.

Voice-over-internet protocol device load profiling

Virtual private network for social networks

Ein System und ein Verfahren schaffen eine wechselseitige Verbindung unter Anwendern von unterschiedlichen sozialen Netzen. Jeder Anwender kann Informationen über das soziale Netz, das er verwendet (z. B. Anwendername, Passwörter usw.), bereitstellen. Eine Anwendung kann sich bei dem sozialen Netz anmelden und kann die Anwendung zwischen den Anwender und das soziale Netz dazwischenschalten. Danach kann das System die Anwender wechselseitig zu einem „virtuellen privaten Netz” (VPN) verbinden, wobei Mitteilungen zwischen den Anwendern und den sozialen Netzen für die anderen Anwender zu anderen sozialen Netzen repliziert werden können. Daher können unterschiedliche Anwender, die unterschiedliche soziale Netze verwenden, kommunizieren.

Voice call methods, devices and electronic equipment

본 출원은 통신 기술 분야에 속하며 음성 통화 방법, 장치 및 전자 장비를 개시한다. 그중, 음성 통화 방법은, 현재 음성 통화가 이루어지고 있는 제1 네트워크의 현재 네트워크 파라미터 정보에 따라, 상기 제1 네트워크의 현재 네트워크 환경 등급 정보를 결정하는 단계; 및 상기 현재 네트워크 환경 등급 정보에 따라, 상기 현재 음성 통화에 대한 대역폭을 할당하는 단계를 포함하며, 상기 현재 네트워크 파라미터 정보는 현재 네트워크 속도 정보를 포함하고, 및 상기 대역폭은 상기 제1 네트워크의 대역폭 또는 제2 네트워크의 대역폭이고, 상기 제1 네트워크와 상기 제2 네트워크는 서로 다른 네트워크이다.

Camera-based facial recognition or other single/multiparty presence detection as a method of effecting telecom device alerting

A camera can be associated with each conference participant endpoint. The camera, either frame or video-based, can monitor and detect one or more of gestures, facial recognition, emotions, and movements of the conference participant. Based on the detection of one or more of these triggering events, a correlation to an action corresponding the triggering event can be evoked. For example, if a participant raises their hand, e.g., a triggering event, the system can recognize that this is a request to speak. The participant can then be queued in the system based, for example, relative to other participants' requests. When the other participants have finished speaking, and it is the time for the user who raised their hand to speak, the system can optionally queue the user by modifying the endpoint with which they are associated.

Call method and system, and related device

Disclosed is a call method, which relates to the field of communications, and is applied to a system at least comprising a first electronic device and a second electronic device. The method comprises: in response to a first call request, enabling a first electronic device to make a call to at least a third electronic device; the second electronic device initiating a second call request to the first electronic device, wherein the second call request is used for requesting to make a call to at least the first electronic device; in response to the second call request, an interface of the first electronic device displaying first prompt information, wherein the first prompt information indicates a call request of the second electronic device; and in response to an input acting on the first prompt information, enabling the second electronic device to make a call with at least the first electronic device and the third electronic device, so that a calling combination requirement of a called user and a multi-party callback requirement of a calling user are met.

Communication apparatus, communication data recording system, communication method, and program

Provided is an IP telephone and the like in which missing of speech start is suppressed. A communication device according to the present invention is provided with: a memory (110) in which data of communication with a communication destination are stored; and a control unit (120) which: performs transmission and reception of the data of communication with the communication destination at a first period; causes the communication data that have been transmitted or received to be stored in the memory (110); and, after a communication session is established with an external recording device for recording the communication data, causes the communication data stored in the memory (110) to be output to the external recording device at a second period shorter than the first period.

Call method and device, intelligent sound equipment and controller and medium

本发明涉及一种通话方法和装置、智能语音设备以及控制器和介质，所述方法包括将智能语音设备接入运营商局域网中；所述智能语音设备获取语音指令；根据所述语音指令执行对应的拨打电话或接听电话过程。本发明使智能语音设备实现高质量的如固话、手机一样的通话功能，提升了用户体验。

Drag and drop importation of content

시스템은 사용자가 한 애플리케이션으로부터 콘텐츠를 드래그하고 별도의 애플리케이션으로 콘텐츠를 드롭할 수 있게 한다. 콘텐츠를 드래그하고 드롭하는 사용자 인터페이스 기능에 응답하여, 임포테이션 시스템은 데이터 주제를 생성할 수 있다. 주제는 드래그되는 애플리케이션 또는 데이터에 기초하여 정의되는 필드를 가질 수 있다. 주제 필드는 드래그 기능 동안에 하이라이트될 수 있는 애플리케이션의 데이터를 검색하는 검색 기능을 사용하여 자동적으로 배치될 수 있다. 드롭이 인터셉트되고 정보는 수신 애플리케이션을 위해 포맷된다. 정보는 수신 애플리케이션의 사양을 만족시키도록 포맷되어 정보의 저장 및 사용을 최적화한다. 따라서, 시스템은 정보를 본래 포맷으로부터 수신 애플리케이션에 의해 이해 가능한 포맷으로 변환할 수 있다. The system allows a user to drag content from one application and drop content into a separate application. In response to a user interface function for dragging and dropping content, the assignment system can create a data subject. The subject may have fields defined based on the application or data being dragged. The subject field can be automatically placed using a search function to retrieve data of the application that can be highlighted during the drag function. The drop is intercepted and the information is formatted for the receiving application. The information is formatted to meet the specifications of the receiving application to optimize the storage and use of the information. Thus, the system can transform information from the original format into a format understandable by the receiving application.

A kind of information input method and its call terminal

本发明公开了一种信息录入方法，其特点是：包括如下步骤，电话呼叫终端键入客户电话号码，并将该客户的电话号码发送至云端服务器；云端服务器根据此电话号码在数据库中进行匹配；与客户建立通话呼叫，在通话呼叫结束后，在电话呼叫终端的显示单元中提示是否修改状态标签并进行语音播报；云端服务器判断新的状态标签是否与数据库中已存的状态标签组相同，若相同，则将客户信息以及新的状态标签存入到对应的状态标签组中；若不相同，则新建一状态标签组，并将客户信息以及新的状态标签存入到新建的状态标签组中。本发明具有客户信息及时录入和客户信息语音播报的特点。

Apparatus and method for automatic repeat request with reduced resource allocation overhead in a wireless voip communication system

A base station (103) assigns a set of mobile stations (101) to a group wherein the group will share a set of radio resources (770). A shared control channel information element (501) is sent to the group of mobile stations (101) and provides a bitmap having fields for group ordering (511), resource allocations (530), continuation resources (540) for HARQ, and an ordering pattern (513). If a mobile station requires retransmission it will access the resources indicated by the continuation resources field (540) in order to receive data. The HARQ blocks may be assigned to a mobile station based upon an index (601) which may correspond to the mobile station vocoder rate. Further, HARQ subgroups may be defined to associate subgroups of mobile stations with specific HARQ transmission opportunities on the super-frame and allocated by a rotating bitmap.

Main and standby framework VoIP radio station gateway based on ED137 protocol

本发明提供了一种基于ED137协议的主备架构VoIP电台网关，该VoIP电台网关设备采用VoIP设计方法，严格依照ED137协议，设计VoIP电台接入接口、VCCS接入方案、主备冗余功能。增加工作模式、增益、延时等多种附加功能。实现CLIMAX、BSS、动态延时补偿等特定功能，极大的体现了VoIP电台优势。该VoIP电台网关可接入VoIP电台接收机、VoIP电台发射机、VoIP电台收发一体机和VoIP电台远控设备等。该VoIP电台网关配备友好的前端管理终端，所有功能都可通过web访问进行管理配置。本发明既满足了空管语音系统对VoIP电台的需求，又满足了空管安全冗余需求。

Relay server, relay method, and relay program

A relay server 31 comprises: a detection unit 31A that detects a failure in a subscriber accommodation server 21; and a relay unit 31B that, when the failure in the subscriber accommodation server 21 is detected by the detection unit 31A, relays a call connection request 51, 52, 53 to separate networks in which the subscriber could be accommodated, in accordance with connection attempt network information indicating the network in which a call connection is attempted.

Method and device for realizing sharing of office telephone system

本发明提供了一种共享办公电系统的实现方法与装置。所述方法包括话机终端通过IP网络连接公司的PBX系统，用户在话机终端输入分机号及密码登录账户，话机终端校验密码，校验成功后话机终端从PBX系统中下载配置文件，话机终端获取配置文件后自动重启，重启后用户再次登录账户，话机终端信息共享成功。以此方式，用户利用PBX系统，使用户能够选择任一话机终端作为自己的分机，并拥有自己之前的电话号码和相应的权限与资源，且能不受地域影响，减少了资源利用，提高了沟通效率。

A kind of lightweight Media Gateway phonetic function implementation method

本发明公开了一种轻量级媒体网关语音功能实现方法，包括：媒体网关向局端发心跳消息；局端向媒体网关发语音通道激活命令；媒体网关将语音通道置为可用，设置用于RTP流传输的IP地址；用户摘机，媒体网关放拨号音；用户拨打电话，向局端报号；局端收到号码中第一位，指示媒体网关停止放拨号音；局端接收完号码后，转被叫，指示媒体网关放回铃音；被叫摘机，局端指示媒体网关停止放回铃音；局端发DSP内容到媒体网关，媒体网关设置DSP语音通道；媒体网关打开DSP语音通道；挂机，向局端报挂机事件；局端指示媒体网关关闭DSP语音通道；RTP流断开，通话结束。本发明，减小了语音功能的复杂度，大大降低了媒体网关设备的故障率。

System apparatus and method for updating communication services

A system for updating communication services includes user equipment configured to communicate using a first communication service standard and a second communication service standard, a subscriber database, an application server, and a network device. The network device detects an update trigger corresponding to a user equipment, requests that the user equipment discontinue using the first communication service standard, and flags the user equipment as updated by communicating with the application server and the subscriber database. The network device also receives a communication from the user equipment and determines whether the communication corresponds to the first communication service standard or the second communication service standard. If the communication corresponds to the first communication service standard the network device communicates an error message to the user equipment. However, if the communication corresponds to a second communication service standard, the network device processes the communication.

Method and system for booting, provisioning and activating hardware and software clients

Automated booting of a client for a subscriber is provided for clients that are for use in interactive user sessions that involve multimedia. A subscribe message is sent from the client to a proxy server. The proxy server authenticates the subscribe message, and sends the subscribe message to the configuration server. A notify message is sent to the client from the configuration server. The notify message is sent through the proxy server, and contains a location of a profile for the client. The profile is downloaded to the clients. This arrangement allows the persistence of profiles in a centralized place.

RTP Media Stream traversing method, sip server and SIP communication system

本发明公开了一种RTP媒体流穿越方法、SIP服务器和SIP通信系统，涉及通信领域，用于解决当互相传输RTP媒体流的SIP用户代理之间存在NAT设备时，RTP媒体流无法穿越NAT设备的问题。RTP媒体流穿越方法包括：所述SIP服务器接收主叫用户代理和被叫用户代理的SIP注册报文或注册保活报文，获取公网IP地址和公网端口号；根据主叫用户代理和被叫用户代理的公网IP地址和公网端口号向所述主叫用户代理和被叫用户代理发送报文以等待进行确认；判定用户代理类型；如果二者均为媒体代理类型，则将RTP媒体信息直接透传给另一方；如果有一方为媒体中继类型，则进行替换并将替换后的SIP报文转发给另一方。本发明的实施例应用于SIP通信。

Method and apparatus for device management

本公开的实施例涉及用于设备管理的方法和装置。一种电话设备包括存储器，该存储器存储用以操作电话设备的固件。该电话设备包括网络接口，该网络接口传输第一命令以建立与电话服务器的连接。该电话设备包括接收器，该接收器从设备管理器接收第二命令。第二命令包括用以升级电话设备的固件的指令。该电话设备包括处理器，该处理器基于第二命令来升级电话设备的固件。处理器基于经升级的固件来建立到虚拟会议室平台的连接。

Updating configuration parameters of clients

Automated booting of a client for a subscriber is provided for clients that are for use in interactive user sessions that involve multimedia. A subscribe message is sent from the client to a proxy server. The proxy server authenticates the subscribe message, and sends the subscribe message to the configuration server. A notify message is sent to the client from the configuration server. The notify message is sent through the proxy server, and contains a location of a profile for the client. The profile is downloaded to the clients. This arrangement allows the persistence of profiles in a centralized place.

Method for Authenticating Caller of Voice Network using Data Network

본 발명의 데이터망을 이용한 전화망의 발신자 인증 방법에 따르면, 교환기를 경유하는 전화망과 교환기를 경유하지 않는 데이터망을 포함하는 서로 다른 이종 망에 다중 연결 가능한 발신측 장치와 데이터망을 통해 통신 가능하며 상기 발신측 장치로부터 전화망을 통해 송출된 호 연결 신호를 확인 가능한 인증시스템을 통해 실행되는 방법에 있어서, 상기 발신측 장치에서 호 연결 요청을 착신할 착신정보를 확인한 경우 상기 발신측 장치로부터 데이터망을 경유하여 상기 호 연결 요청의 발신정보와 착신정보 중 적어도 하나의 식별정보와 상기 발신측 장치에 구비된 인증키 생성모듈이 상기 발신정보와 착신정보를 이용하여 생성한 인증키 값을 수신하여 매칭 유지하고, 상기 발신측 장치에서 전화망을 경유하여 송출한 호 연결 신호를 판독하여 발신정보와 착신정보를 검출하고, 상기 전화망을 통해 검출된 발신정보와 착신정보 중 적어도 하나의 식별정보와 매칭된 인증키 값을 확인하고, 상기 매칭된 인증키 값을 확인한 경우 상기 전화망을 통해 검출된 발신정보와 착신정보를 이용하여 검증키 값을 생성하고, 상기 전화망을 통해 생성된 검증키 값과 상기 데이터망을 통해 수신된 인증키 값을 비교 인증하여 상기 전화망을 통해 상기 착신정보로 호 연결을 요청한 발신측 장치가 상기 인증키 값을 생성한 인증키 생성모듈을 구비한 유효한 발신측 장치인지 인증하며, 상기 전화망을 통해 호 연결을 요청한 발신측 장치를 상기 데이터망을 통해 인증한 경우 상기 데이터망을 통해 전화망의 발신측을 인증한 결과에 대응하는 발신자 인증 데이터를 구성하여 착신자에게 출력되도록 처리한다.

Telephone outlet with packet telephony adapter, and a network using same

Automatically registering a user in a desk-share environment and ip telephone

Supporting hitless upgrade of call processing nodes in cloud-hosted telephony system

A method is provided in which a call agent process that supports one or more Internet Protocol (IP) calls, stores to persistent memory a set of data associated with the one or more IP calls. An outage is detected affecting the one or more IP calls. Using the data retrieved from the persistent memory, the one or more IP calls are resynthesized using a device simulator process to simulate connectivity with endpoints that were participating in the one or more IP calls prior to the outage. After resynthesizing, depending on activity detected from devices associated with the one or more IP calls, the one or more IP calls are internally re-stitched/re-establishing (without signaling to endpoints) with the endpoints involved in the one or more IP calls, or the one or more calls are fully re-stitched/re-established by signaling an endpoint that was participating in the one or more IP calls.

Administering a communication network

Methods and apparatus are provided for administering a communication network. In one embodiment, a first interface screen depicts nodes within a network. The nodes include a pair of media aggregation managers providing multiplexing/demultiplexing of media traffic associated with multiple application sessions between a pair of communities onto a preallocated reservation protocol session between the media aggregation managers. The media aggregation managers are visually distinguishable from other nodes. A second interface screen depicts potential paths through the network. Each potential path is capable of transferring media packets between the media aggregation managers. Via the second user interface screen, a network administrator is capable of initiating (1) path-level configuration of routers that are part of a selected potential path to cause the routers to route media packets exchanged between the pair of communities over the selected path, and (2) establishment of the preallocated reservation protocol session between the media aggregation managers.

Communication system, service processing method and device

本发明公开了一种通信系统、业务处理方法及装置，所述系统包括：包括互联网协议多媒体子系统IMS、与所述IMS通讯的至少一个演进型分组核心网EPC；所述多个EPC中的任两个EPC的IP地址的网段不同；所述多个EPC中的任一EPC用于为附着于所述EPC下的UE分配IP地址，所述UE的IP地址与所述EPC的IP地址的网段相同。

Telephony discovery mashup and presence

Embodiments of the present invention are directed to a communication system that provides various automated operations, including searches of public and private resources for third party information and presentation of the located third party information by a mashup and preparation of a mashup to combine different communication modes or modes in a common session.

Voice communication method, device, equipment and storage medium

本公开是关于一种语音通信方法、装置、设备及存储介质，所述方法包括：响应于建立与第一设备之间的第一语音通信连接，并在用户界面上预设的正在通话区域显示所述第一设备的第一标识信息；响应于在基于所述第一语音通信连接与所述第一设备进行语音通信的过程中接收到第二设备发送的第二语音通信请求，在所述用户界面上预设的等待通信区域显示所述第二设备的第二标识信息。本公开可以为用户展示当前正在进行语音通信连接的设备的标识信息，并且可以展示当前等待进行语音通信连接的设备的标识信息，并且无需当前等待进行语音通信的用户自行估算时间进行再次呼叫，可以提高通信效率，提升用户体验。

Communication device, communication data recording system, communication method, and program

Telephony discovery mashup and presence

Embodiments of the present invention are directed to a communication system that provides various automated operations, including searches of public and private resources for third party information and presentation of the located third party information by a mashup and preparation of a mashup to combine different communication modes or modes in a common session.

The method of measurement of audio transmission time delay and system

本发明公开了音频传输延时的测量方法及系统。其中，该方法包括：在发送端和接收端之间对待测的原始音频码本的传输进行同步操作，得到原始音频码本的发送开始指示信息、发送结束指示信息、接收开始指示信息、接收结束指示信息；发送端响应发送开始指示信息开始向接收端发送待测的原始音频码本，响应发送结束指示信息停止向接收端发送原始音频码本，接收端响应接收开始指示信息开始对发送端发送的原始音频码本进行采集，并响应接收结束指示信息停止对发送端发送的原始音频码本进行采集；根据接收端采集得到的测试音频码本以及接收端上预存的原始音频码本得到音频传输延时。本发明解决了现有技术中音频传输延时计算不准确的问题。

Social network virtual private network

A system and method create an interconnection among users of different social networks. Each user can provide information about the social networks they use (e.g., username, passwords, etc.). An application may register with the social network and can inter-dispose the application between the user and the social network. Then, the system can interconnect the users into a “virtual private network” (VPN) where communications between the users and the social networks can be replicated to other social networks for the other users. As such, different users using different social networks can communicate.

Internet protocol (ip) serverless page party (spp) station and systems and methods for deploying multiple spp stations

A serverless Page Party (SPP) station is provided for page announcing and party line conferencing that is configured to implement mutual provisioning to allow all SPP stations within a system to mutually maintain their configurations in runtime, and multiple master station negotiation and master failover. Plural SPP stations are connected with Ethernet cable or WIFi to a network and therefore without need for expensive and cumbersome cabling and daisy-chain configuration among stations. SPP stations route packets between each other via multicast technology and without need for a IP-PBX or other server (e.g., a SIP server) for controlling inter-station connections. In Mutual Provisioning Mode, SPP stations operate using a system configuration obtained from other SPP stations already on the network using a command channel. System configuration is maintained by a SPP station designated as a master station. Plural SPP stations can negotiate among themselves to designate a new master station.

Method and system for reducing load of IMSSIP server based on VOIP equipment polling group

本发明属于VOIP设备轮询组技术领域，具体涉及一种基于VOIP设备轮询组减少IMSSIP服务器负荷方法、系统及平台。本发明通过方法配置轮询组参数数据；根据不同呼叫类型进行相应的轮询组处理；拨打来话时的被叫号码为轮询组主号码，根据优先级进行相应的振铃序列；被叫摘机后停止振铃序列，然后双方通话，任何一方挂机后结束通话；拨打去话时的主叫号码设为轮询组主号码，被叫振铃主叫听回铃音；被叫摘机后双方通话，任何一方挂机后结束通话；以及与所述方法相应的系统、平台，本方案可以实现快速组网；资源共享，节约成本；减少IMS/SIP服务器的负荷；当IMS/SIP服务器出现故障时，还可以继续拨打内部电话。