155 results about "Session Initiation Protocol" patented technology

A system of a telecommunications network that blocks spoofed calls. The system includes an interconnect session border controller (I-SBC) that can receive an inbound call over a network interface. The inbound call is associated with an identifier (e.g., a Session Initiation Protocol (SIP) identifier). The I-SBC causes a Spoofed Call Identification Function (SCIF) to compare the SIP identifier of the inbound call against a list of allowable SIP identifiers. In one example, the I-SBC receives an indication from the SCIF that the SIP identifier of the inbound call does not match any identifier on the list of allowable SIP identifiers. As such, the system rejects the inbound call as an illegitimate call. The illegitimate call is not forwarded to an IP Multimedia Subsystem (IMS) of the telecommunications network.

Techniques are described for sending an emergency related Short Message Service message (SM) from a user equipment (UE) to a Public Safety Answering Point (PSAP) with higher reliability and security and lower delay than with previous techniques. Following detection of a user input emergency SM, a UE establishes a Session Initiation Protocol (SIP) dialogue with a PSAP, where the SIP dialogue has no active SIP media, which can be compliant with SIP protocol rules. The emergency SM is sent to the PSAP in association with the SIP dialogue, as are any subsequent SMs, and PSAP SM replies are similarly returned to the UE. Media such as voice may be added and previous restrictions on support of UEs that are roaming or in limited service state can be overcome.

Various aspects of the present disclosure relate to a network system, user equipment (UE), method, and processor for wireless communication that supports rich data call establishment. The network system receives a message from an originating UE, and a first network entity obtains, from a second network entity, headers, payload, with token claims of a personal assertion token (PASSporT) extension for rich call data, and a private key certificate associated with a user identifier contained in the message. The first network entity creates a signature and at least one header of the PASSporT extension for rich call data, adds the signature in a first Session Initiation Protocol (SIP) header to the message, and transmits the message with the header to a third network entity for verification. When verified, the message with a second IP header is forwarded to a terminating UE to establish an IMS session with the originating UE.

The invention relates to the technical field of video return, in particular to an unmanned vehicle video return control method and system based on an SIP (Session Initiation Protocol) communication protocol. The control method comprises the steps that a home terminal serves as an SIP client and sends an SIP connection request to an unmanned vehicle terminal through TLS encryption, it is determined that the home terminal is a video only decoding terminal and the unmanned vehicle terminal is a video only coding terminal, meanwhile, the audio two-way communication capacity is negotiated, and encrypted signaling connection is established. As a mature multimedia communication protocol, the SIP has a perfect and safe session establishment, modification and termination mechanism. The method has good network adaptability, supports NAT traversal and firewall traversal, can stably work in different network environments, and is suitable for unmanned vehicle moving scenes. According to the invention, by constructing a systematized control scheme based on the SIP protocol, the problems of dynamic bandwidth adaptation and safe and reliable transmission control of unmanned vehicle video return in a complex network environment are solved.

A user equipment (UE) is configured to initiate a packet-switched call via a first radio access technology (RAT) and start a plurality of session initiation protocol (SIP) timers responsive to initiating the call. Responsive to at least one condition being present with initiation of a fallback process for the call, the UE is configured to restart a set of the plurality of SIP timers for the fallback process, wherein the set can include some or all of the plurality of SIP timers started with initiation of the call. Each SIP timer has a corresponding default timer duration, and in some instances restarting a SIP timer of the set includes restarting the SIP timer with a starting duration equal to a sum of the default timer duration and an offset so that the restarted SIP timer provides for a greater duration than when originally started to better accommodate the fallback process.

The invention discloses a low-loss vertical transmission transition structure, belongs to the technical field of microwave and millimeter wave circuits, and solves the problem of impedance mismatch in a signal transmission process of an SIP (Session Initiation Protocol) packaging device. A packaging substrate of the SIP packaging device is provided with a non-standard rectangular waveguide, a ball planting type waveguide structure constructed by using the idea that wide and narrow sides of the waveguide are slotted at the same time, a two-order waveguide cavity formed in a multi-layer printed board and a standard rectangular waveguide cavity in a metal waveguide are concentrically arranged from top to bottom; a vertical transmission transition structure is formed by the planting ball type waveguide structure and the two-order waveguide cavity and is used for realizing bidirectional low-loss transition transmission of signals; according to the invention, efficient vertical transmission of signals among the SIP packaging device, the multi-layer printed board and the metal waveguide is realized, the advantages of low transmission loss, simple structure and easy integration are realized, and key technical support is provided for wide application of W-band and terahertz-band SIP devices.

A first electronic device and a method for forwarding a call from a second electronic device are provided. A first electronic device includes a transceiver, memory storing instructions, and one or more processors configured to execute the instructions to receive from a server a first session initiation protocol (SIP) invite message generated at the second electronic device, through the transceiver, determine whether to forward the call from the second electronic device, in response to receiving the first SIP invite message, upon determining to forward the call from the second electronic device, obtain contact information related to the call forwarding from the second electronic device, generate a response message comprising the contact information related to the call forwarding, and transmit the response message to the server to provide the response message comprising the contact information from the server to the second electronic device through the transceiver, wherein the contact information in the response message transmitted to the server is used by the server to generate a second SIP invite message to be transmitted to the third electronic device.

The invention provides a network telephone system and a customer service scheduling method. The network telephone system comprises a client device, a private switch and at least one service extension set. The client device comprises a processing unit and a transmission unit. The private switch comprises a communication unit, an interactive voice response unit, a control unit and a distribution unit. In a call access procedure, a processing unit of the client device sets a push token to a first Session Initiation Protocol header of first Session Initiation Protocol information; if the service extension set is busy, the control unit of the private switch records a scheduling call from the push token and the service extension set to a queue list, and records a scheduling state of the scheduling call as to-be-pushed;

A method for handling a call using Sessions Initiation Protocol (SIP) includes sending an INVITE or a first media type by a network side to user equipment, sending an UPDATE of a second media type by the user equipment to the network side, receiving an SIP global failure response for the UPDATE by the user equipment from the network side, sending an SIP success response for the INVITE of the first media type by the user equipment to the network side, receiving an ACK of the first media type by the user equipment from the network side, and establishing a call session of the first media type between the user equipment and the network side.

Embodiments of the present application relate to the field of communications, provide a communication method and apparatus, and can reduce power consumption of terminal devices supporting a plurality of DC applications and enable free message and notification transmitting and receiving of the DC applications. The method comprises: when a terminal device registers to an IP multimedia subsystem (IMS) network on the basis of a session initiation protocol (SIP) message, a push server receives a first push message from a server of a DC application and triggers a first SIP message on the basis of the first push message, wherein the first push message is used for requesting to send a push notification to the DC application on the terminal device, and the first SIP message is used for sending the push notification to the DC application. Therefore, on the basis of an existing SIP persistent connection established based on a call, the SIP persistent connection is reused to send a push notification to a DC application in a closed state or tombstone state by means of an SIP message, such that power consumption is reduced, and free message and the notification transmitting and receiving can be achieved.

The application discloses a media negotiation method and device, related equipment, a program product and a storage medium. The method is applied to a network device provided with multiple service platforms, and includes the following steps: receiving a first message sent by a called terminal; triggering a first service platform in the multiple service platforms to send a second message to a second service platform based on the first message; the second message includes a session initiation protocol (SIP) message parameter; the SIP message parameter is used to indicate a time point of initiating media renegotiation in a call process and a service triggering the media renegotiation; determining a processing strategy of the second service platform according to the SIP message parameter; and sending the second message to the called terminal based on the processing strategy to complete the media renegotiation. By adding the SIP message parameter, the purpose of multiple media negotiations can be achieved in one SIP interaction, thereby reducing network signaling overhead.

Novel tools and techniques are provided for implementing fraud or distributed denial of service (“DDoS”) protection for session initiation protocol (“SIP”)-based communication. In various embodiments, a computing system may receive, from a first router, first SIP data indicating a request to initiate a SIP-based media communication session between a calling party at a source address and a called party at a destination address. The computing system may analyze the received first SIP data to determine whether the received first SIP data comprises any abnormalities indicative of potential fraudulent or malicious actions. If so, the computing system may reroute the first SIP data to a security deep packet inspection (“DPI”) engine, which may perform a deep scan of the received first SIP data to identify any known fraudulent or malicious attack vectors contained within the received first SIP data. If so, the security DPI engine may initiate mitigation actions.

A three-channel power supply module based on SIP (Session Initiation Protocol) technology relates to the technical field of integrated circuits and comprises a three-channel power supply circuit, an active chip bare core and a passive device of the three-channel power supply circuit are packaged in an integrated ceramic tube shell, and the top of the integrated ceramic tube shell is connected with a metal cover plate by adopting a parallel seam welding process; the bare core is connected with the ceramic substrate through a conductive adhesive, the bare core pad is connected with the ceramic substrate through 25 [mu] m gold wire ultrasonic pressure welding, and the passive device is connected with the ceramic substrate through a tin soldering process. According to the invention, the SIP system-level packaging technology is adopted, the wide voltage input and three-path power supply output functions are realized, power can be supplied to a digital circuit and an analog circuit at the same time, a certain power supply universality requirement is met, various power utilization scenes of avionic products can be met, high power density and high integration level can be realized, the space is greatly saved, and the cost is reduced. The device has obvious advantages in the aspects of miniaturization, modularization, light weight and the like of products.

The present disclosure generally relates to message processing in communication systems, more particularly to systems and methods for mapping Session Initiation Protocol (SIP) response with appropriate cause code. BTAS receives SIP response, and determines reason for error in call, and cause code for error. BTAS maps SIP and appropriate cause code. BTAS provides, in SIP response, appropriate reason for call failure or call error. Call is sent to E-SBC, routed from E-SBC to P-CSCF, and routed from P-CSCF to S-CSCF. A query is sent from S-CSCF to DNC server regarding DNC registration of B number. DNC server answers, if B number is registered / Non-registered for DNC. Call is forwarded to BTAS, and query is sent from BTAS to MNP. BTAS routes call to S-CSCF. Wireline calls to terminating BTAS, in which VOLTE calls to TAS, and offnet calls to MGCF via BGCF. Offnet calls is sent from BGCF to MGCF.

The application provides an abnormal call identification method, device, equipment, medium and program product, and relates to the technical field of communication, to solve the problem of low accuracy in identifying abnormal calls. The method comprises the following steps: obtaining a session initiation protocol (SIP) signaling of a voice over IP (VoIP) call based on a target interface, wherein the target interface is an interface between a mobile wireless network and a core network; analyzing the SIP signaling to obtain the bearing content corresponding to the SIP signaling; and determining whether the voice call corresponding to the SIP signaling is an abnormal call based on the bearing content. The application can improve the accuracy of identifying abnormal calls and accurately identify abnormal calls.

Certain aspects of the present disclosure provide techniques for managing loss of user equipment (UE) context information during internet protocol multimedia subsystem (IMS) procedures. An example method includes starting an internet protocol multimedia subsystem (IMS) procedure failure reset timer, transmitting, while the IMS Procedure Failure Reset Timer is running, one or more IMS session initiation protocol (SIP) messages to a network entity, detecting that the IMS Procedure Failure Reset Timer has expired without receiving an IMS response message from the network entity responding to the one or more IMS SIP messages, determining, in response to detecting that the IMS Procedure Failure Reset Timer has expired without receiving an IMS response message, that a synchronization timer associated with a connection between the UE and the network entity is running, and performing, based on the determination that the synchronization timer is running, a non-access stratum (NAS) procedure with the network entity.

The invention relates to a message transmission method and system based on a session initiation protocol and a storage medium, and the method comprises the steps: converting a service request distributed by a service gateway into a first session initiation protocol instruction, and transmitting the first session initiation protocol instruction to a session initiation protocol interaction object, meanwhile, the data are stored in the distributed cache middleware; after receiving a receiving response, moving the first session initial protocol instruction out of the distributed cache middleware, and after sequentially receiving a plurality of second session initial protocol instructions, respectively storing the second session initial protocol instructions into the distributed cache middleware; and reading the same batch of second session initial protocol instructions, converting the second session initial protocol instructions into request results, and sending the request results to the service gateway. According to the method, the strong binding relationship between the session initial protocol interaction object and the session initial protocol signaling service node can be relieved, the difficulty of session initial protocol signaling service lateral expansion is reduced, and the equipment bearing capacity of a single service node is improved.

The invention relates to an intelligent power module SIP (Session Initiation Protocol) integrated circuit applied to household appliances. A signal processing part is composed of an MCU (Microprogrammed Control Unit) chip and a peripheral circuit; each half-bridge driving chip and the bootstrap circuit and the power MOS tube connected with the half-bridge driving chip form a group of driving circuit parts, and the three groups of driving parts respectively output driving signals U, V and W to control a motor M; the signal processing part, the driving circuit part and the linear voltage regulator are integrated on a BT substrate together; the signal processing part is further provided with a passive element which is integrated on a semiconductor silicon wafer and then integrated on a BT substrate. The system has the beneficial effects that the MCU chip, the peripheral circuit, the driving circuit and other parts are integrated in SIP packaging, the design flexibility is enhanced, and a comprehensive integration scheme meeting various household appliance applications is realized; based on a peripheral circuit integration scheme of an IPD (passive component) technology, a functional circuit which needs to be realized by means of peripheral discrete components is realized at a wafer level.

Various aspects of the present disclosure generally relate to wireless communication. In some aspects, an authentication, authorization, and accounting (AAA) server may authorize a use of a third party specific user identity based at least in part on verifying a token associated with a session initiation protocol (SIP) INVITE message with the third party specific user identity. The (AAA) server may transmit, to a call session control function (CSCF) of an originating service provider via a network exposure function (NEF) of the originating service provider, an enterprise name card that indicates the third party specific user identity based at least in part on authorizing the use of the third party specific user identity. Numerous other aspects are described.

The invention discloses an SIP (Session Initiation Protocol) packaging ceramic substrate for a communication module. The SIP packaging ceramic substrate comprises a multi-layer ceramic substrate and a microstrip line structure arranged in the multi-layer ceramic substrate, the multi-layer ceramic substrate is provided with a TOP surface and a BTM surface, and the multi-layer ceramic substrate comprises a TOP layer, a middle layer and a BTM layer; the preparation method comprises the steps of tape casting slurry formula and precise thickness control, via hole laser manufacturing, inner electrode printing, raw ceramic lamination and lamination, raw ceramic warm water isostatic pressing, block segmentation and glue discharging and sintering. According to the high-density wiring technology HDI adopted by the invention, the line width and the spacing can be reduced to 0.05 mm, and the packaging density and the packaging performance are greatly improved. According to the technology, more I / O and grounding welding spot layouts can be realized in the ceramic substrate, and the requirement of high-density multi-chip packaging is met.

A system receives an electronic message over the internet at a WebRTC gateway (WRG) of a telecommunications network. The system forwards the electronic message to a call session control function (CSCF) using session initiation protocol (SIP). The system detects a failed delivery of the electronic message at a rich communication service configured to deliver and monitor delivery of the electronic message and report failed deliveries. The system generates a failed delivery notification indicating that the electronic message was not delivered. The system receives the failed delivery notification at the WRG and at the WRG converts the electronic message to either a short messaging service (SMS) or multimedia messaging service (MMS) message. The system sends the SMS or MMS message from the WRG to the CSCF using SIP. The system forwards the SMS or MMS message to a short message service center (SMSC), which delivers the SMS or MMS message.

Methods and systems for switching a push-to-talk (PTT) session to a voice over IP session are described herein. In one implementation, a device configured to support PTT service may receive an interaction performed by a user with the device. The device then transmits a request to establish a PTT session with a second user to a PTT server connected to an evolved packet core (EPC) network. Once the PTT session is established, the PTT server may monitor the real-time audio quality of the PTT session. When the audio quality is determined to be low, the PTT server may send a notification to the device to switch the PTT session to a two-way audio session (e.g., voice over IP). Upon receiving the user's confirmation, the device may send a session initiation protocol (SIP) invite to an IP multimedia subsystem (IMS) to establish the voice over IP session.

The present disclosure discloses methods and electronic device for managing a Multi-Device Environment (MDE) session for a client device in a communication network. The method includes transmitting, by the client device, a request to a MDE server of the communication network via one of a plurality of communication protocols. The request comprises a tag indicating the request for one of a call forking session and a text based session. The method further includes receiving by the client device, one or more notification messages from the MDE server using the communication protocol used for transmitting the request data packet. The one or more notification messages comprise one or more JavaScript Object Notation (JSON) wrapped Session Initiation Protocol (SIP) messages when the one or more notification messages correspond to the call forking session.

The present application relates to a determination method and apparatus for an on-board call service, and a network element device and a storage medium. The method comprises: when it is determined that a first device is in regenerative satellite access, sending a first session initiation protocol (SIP) message, wherein the first SIP message comprises first regenerative satellite access information of the first device.

The present invention relates to methods, systems, and apparatus for learning-based automated call flow testing. An exemplary method of testing a Session Initiation Protocol (SIP) call processing entity includes the steps of generating a first test case based on a first reference call flow record for a first SIP call or a trace record for the first SIP call; executing the first test case which includes sending one or more SIP messages to the first SIP call processing entity; generating a first test case call flow record for a first SIP test call corresponding to the first test case; and determining whether or not the first SIP call processing entity failed the first test case based on a comparison of the first test case call flow record and the first reference call flow record.

A method performed by an electronic device is provided. The method includes transmitting, to another electronic device through a server, a session initiation protocol (SIP) INVITE message for a connection request of a session. The SIP INVITE message includes information for indicating an authentication request. The method includes receiving, from the other electronic device through the server, a SIP information message for indicating a ringing including information for indicating an authentication support. The method includes identifying whether an SIP response message for indicating a success of the connection request is received from the other electronic device or a target electronic device for a call forwarding or not. The method includes establishing the session in response to a reception of the SIP response message. The reception of the SIP response message indicate an authentication success in the other electronic device or the target electronic device, according to the authentication request.

The present invention relates to methods, systems, and apparatus for learning-based automated call flow testing. An exemplary method of testing a first Session Initiation Protocol (SIP) call processing entity includes the steps of: performing a twinning configuration procedure on a second SIP call processing entity operating in a production network; generating configuration information for use in configuring the first SIP call processing entity in a lab environment for a plurality of test cases based on configuration information obtained during the twinning configuration procedure, the plurality of test cases including a first test case based on a first reference call flow record for a first SIP call processed by the second SIP call processing entity; and executing the first test case while the first SIP call processing entity is operating in the lab environment and has been configured using the generated configuration information.

The present disclosure relates to carrier integration through a user web interface proxy. A method (300) of carrier web integration through a user proxy interface includes receiving a request (130) from a mobile device (110) associated with a subscriber (10) of a carrier provider (301) to establish a voice service with the carrier provider for a companion device (120) linked to the mobile device. The request includes a subscriber identifier (132) of the mobile device. The method further includes authenticating the companion device based on the subscriber identifier, obtaining a session initiation protocol (SIP) credential (140) corresponding to the subscriber identifier, and registering the SIP credential for the companion device at a web interface proxy (200). The web interface proxy is in communication with an internet protocol multimedia subsystem (IMS) (320) of the carrier provider and the companion device. The method further includes establishing the voice service between the companion device and the IMS through the web interface proxy using the SIP credential.

The present disclosure relates to a 5G or 6G communication system for supporting higher data transmission rates. A method performed by a first Internet protocol multimedia core network subsystem (IMS) application server (AS) in a wireless communication system according to an embodiment of the present disclosure may comprise the steps of: receiving a first session initiation protocol (SIP) INVITE message including third party identification information from a first user equipment (UE); confirming rich call data (RCD)-related information from a home subscriber server (HSS) in response to the first SIP INVITE message; and receiving, from the HSS, address information of an RCD server included in the RCD-related information.

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.