10results about "Telephone data network interconnections" patented technology

The present disclosure provides a method for managing voice calls and gateway device. A gateway device according to the embodiment of the present disclosure includes a voice call module for being connected with a landline device and / or a mobile voice device and configured to provide a calling out service to the landline device and / or the mobile voice device; a cellular session initiation protocol (SIP) client module for being connected with a cellular network and configured to process voice signalling in a cellular network; an IP network SIP client module for being connected with an IP network and configured to process voice signalling in an IP network; and a voice call selection module configured to, when receiving a calling out from the voice call module, select either the cellular SIP client module or the IP network SIP client module for the calling out.

Fax device for conducting communication using a session established according to a Session Initiation Protocol (SIP), with: a transmission device for transmitting an INVITE signal, in which "image" is designated as a media type, in a case where an INVITE signal is received, in which "audio" is designated as a media type; a fax communication device for establishing an image session and executing a fax communication in the event that a successful response is received to the INVITE signal transmitted by the transmission device; and a call device for initiating call processing to call a user in the event that an error response is received to the INVITE signal transmitted by the transmission device.

A method for routing traffic for voice communication includes: receiving a request to establish a telecommunication link with a destination phone number; transmitting the received request to establish the telecommunication link to the destination phone number via a session initiated protocol (SIP) trunk; determining that the request to establish the telecommunication link via the SIP trunk failed; and transmitting, via a communication channel other than the SIP trunk, the received request to establish the telecommunication link to a separate destination based on a predetermined routing protocol.

A method for routing traffic of voice communications includes: receiving a request to establish a telecommunications link with a destination phone number; sending the received request to establish the telecommunications link to the destination phone number via a session initiation protocol (SIP) relay; determining that the request to establish the telecommunications link via the SIP relay fails; and sending the received request to establish the telecommunication link to a separate destination based on a predetermined routing protocol via a communication channel other than the SIP relay.

The present disclosure provides a method for managing voice calls and gateway device. A gateway device according to the embodiment of the present disclosure includes a voice call module for being connected with a landline device and / or a mobile voice device and configured to provide a calling out service to the landline device and / or the mobile voice device; a cellular session initiation protocol (SIP) client module for being connected with a cellular network and configured to process voice signalling in a cellular network; an IP network SIP client module for being connected with an IP network and configured to process voice signalling in an IP network; and a voice call selection module configured to, when receiving a calling out from the voice call module, select either the cellular SIP client module or the IP network SIP client module for the calling out.

Apparatus and methods concerning routing of calls in client-configured set of servers, such as a web-server configured to host a switchboard interface are disclosed. As an example, one apparatus includes such a hosting web-server configured to route VoIP calls for a plurality of end-users having respective VoIP-enabled devices communicatively coupled to the web-server via a network. A user-activity detection circuit is communicatively coupled to the web-server. The user-activity detection circuit is configured to receive activity data indicating activity of end-users in a virtual office environment and receive event messages indicating statuses of pending calls in the web-server. The user-activity detection circuit determines user-activity statuses of the end-users based on received activity data and event messages. A computing server is to provide a set of services for one or more of the plurality of users as a function of the determined user-activity statuses of the end-user.

A voice call management method is provided. The voice call management method may be applied to an apparatus. The voice call management method may include the following steps. The apparatus may determine the current scenario associated with the operation environment of the apparatus. Then, the apparatus may determine to perform a voice call through a modem (MD) voice engine or through an application processor (AP) voice engine according to the current scenario.

Embodiments are disclosed for securing internet-based transfer of voice call data. Embodiments may include: sending a primary-protocol-encoded communication, by a forwarding unit (FU), to a session border controller (SBC), when an indication is received at the FU that a secondary-protocol-encoded communication has been received at a gateway component, from a voice call data source, to propose transferring voice call data to the SBC; outputting by the SBC, in response to the primary-protocol-encoded communication an IP address encoded with the primary protocol; converting, by a gateway component, the IP address encoded with the primary protocol into an IP address encoded with the secondary protocol; and sending, by a gateway component, the IP address encoded with the secondary protocol to the voice call data source that is proposing to transfer the voice call data.

Aspects of the subject disclosure may include, for example, a method in which a processing system determines a plurality of combinations of settings for a session border controller (SBC) communicating with a first network and installed at a premises. The method also includes listing the combinations, and identifying a combination that coincides with settings of a primary rate interface (PRI) circuit. The method further includes enabling a first connection between a first port of the SBC and the PRI circuit and a second connection between a second port of the SBC and a second network; settings of the SBC correspond to the identified combination, and the SBC comprises a connection between the first and second ports. The method also includes subsequently disabling the second connection, resulting in the PRI equipment communicating with the first network without a change to the settings of the PRI equipment. Other embodiments are disclosed.

A voice call transfer method includes when a target device receives an incoming voice call as a called party, displaying, on a call application interface, a device identifier of a first device that can perform voice call transfer, receiving a selection operation performed on the device identifier, triggering the target device to enter a voice call transfer mode, and sending first voice data from a calling party device to the first device when it is determined that the target device performs a voice call with the calling party device and is in the voice call transfer mode, so that the first device plays the first voice data.