Patent classifications
H04L65/10
Controller to determine a transmission right and control a timing of communication between nodes
A controller includes a real-time communication unit, an application communication unit, and a socket management unit. The real-time communication unit communicates with an external node by using a first socket on a basis of a transmission right map indicating a node possessing a transmission right out of multiple nodes. The application communication unit communicates, not on a basis of the transmission right map, with the external node by using a second socket differing from the first socket. The socket management unit restricts at least one of transmission of information by the application communication unit to the external node and reception of information by the application communication unit from the external node.
Making a Dialogue Available To an Autonomous Software Agent
A user terminal comprising a processor comprising one or more processing devices configured to run a communication client to establish a communication event with nodes in a communication network; a display on which contact identifiers are displayed, each contact identifier being selectable to initiate a communication event with a node addressed by the contact identifier. A user interface enabling a user to engage in an interaction with the user terminal, including communicating via an established communication events with at least one other node in the communication network associated with a human user, whereby messages in the communication event are available to an autonomous software agent (ASA) to convey an intent conveyed in a dialogue between the user terminal and the human user at the at least one other node, and the processor is configured to receive and present to the user a response to the intent received from the ASA.
Mute Call Detection in a Communication Network System
Information on a connection status between a user equipment which accesses a core network of a communication net work system via a radio access network of the communication network system, and the core network is acquired (S21). From the information on the connection status it is determined (S22) whether or not data packets should be communicated between the user equipment and the core network. In case it is determined that data packets should be communicated, it is detected (S23) whether or not data packets are present on a user plane path between the user equipment and the core network. In case it is detected that data packets are not present on the user plane path, a silence report is generated and information on the detection result is indicated in the silence report (S24). Based on the silence report, a mute call between the user equipment and the core network is detected (S25).
Method and apparatus for detecting application
Provided is a method for detecting an application in a wireless communication system. The method includes receiving and inspecting a packet; detecting flows from the packet using a predefined signature; granting a score to each of the detected flows, and summing the granted scores by integrating the detected flows for each application; comparing the summed score of the flows integrated for each application with a preset value; and determining that an application is detected, if the summed score is greater than the preset value.
Gateway device and monitoring method
Unexpected call disconnection during normal time is prevented. When a gateway device 1 installed on a POI border receives RTP packets even though reception of RTCP packets is stopped, the gateway device 1 generates RTCP packets and sends out the RTCP packets to the gateway device 1's own network side or the gateway device 1 generates a call control signal showing that media transfer is continued and sends out the call control signal to the gateway device 1's own network side. Thereby, even in the case of performing interruption monitoring of RTCP packets within the gateway device 1's own network, it is possible to prevent unexpected call disconnection during normal time accompanying change in RTCP packet sending-out conditions.
Electronic device providing IP multimedia subsystem (IMS) service in network environment supporting mobile edge computing (MEC)
An electronic device is provided. The electronic device includes a processor and a memory operatively connected to the processor. The memory stores instructions that, when executed, cause the processor to determine that IP multimedia subsystem (IMS) data to be transmitted and received between a first user equipment and a second user equipment can be processed by the same mobile edge computing (MEC) host based on location information received from the first user equipment and the second user equipment, instruct the MEC host to activate an IMS processing function, and transmit an address of the activated IMS processing function of the MEC host to the first user equipment and the second user equipment. Other embodiments are possible.
METHOD OF ENSURING VOICE OVER INTERNET PROTOCOL RELIABILITY AFTER ENTERING A POWER SAVING MODE
A method is provided for causing a networking device to enter a power saving mode, determining whether or not a Voice over Internet Protocol (VOIP) telephony interface is in service after the causing the networking device to enter the power saving mode, the networking device including the VOIP telephony interface, the VOIP telephony interface communicatively connected to a VOIP provider, and rebooting the networking device when the VOIP telephony interface is determined not to be in service.
Non-associative telephony and SMS messaging
Systems and methods for managing non-associative communications between devices is provided. A first call chain that indicates a routing between phone numbers is stored. A first phone call or a first SMS text is received from a first session initiation protocol (SIP) provider. Based on information provided by the first SIP provider, (i) a sender identity of the first phone call or the first SMS text; (ii) a receiver identity of the first phone call or the first SMS text; and (iii) an access mode of the call chain are determined. If the receiver identity corresponds to a first phone number in the first call chain, a second phone call or a second SMS text is initiated via a second SIP provider, from a second phone number in the first call chain, based on the sender identity and the access mode.
Method for performing codec adaptation in a UE operating in a communication network
A method, user equipment and chipset is used for a first user equipment (UE) having a codec in which at least one of encoding audio or video data and decoding of the audio or video data is performed. A first bit rate is acquired for the codec from a second user equipment that is a communication partner of the first user equipment, a base station a message is transmitted to a base station for checking whether the first bit rate can be provided by a base station. The message includes information indicating the first bit rate.
Method for performing codec adaptation in a UE operating in a communication network
A method, user equipment and chipset is used for a first user equipment (UE) having a codec in which at least one of encoding audio or video data and decoding of the audio or video data is performed. A first bit rate is acquired for the codec from a second user equipment that is a communication partner of the first user equipment, a base station a message is transmitted to a base station for checking whether the first bit rate can be provided by a base station. The message includes information indicating the first bit rate.