381results about "Interconnection arrangements" patented technology

The method and systems for automatically identified currently viewing website domain name or a uniform resource locator (URL), receiving, by the server, said automatically identified website domain name or the uniform resource locator (URL), a request or an invitation for initiating a communication including live chat, puts the request for initiating a communication including live chat from the first user and the second user in a communication including live chat queue of the website or the uniform resource locator (URL) associated account, for available or identified or relevant agents to pick up, routing request or invitation for initiating a communication including live chat from the first user and the second user according to a pre-set rules and start a communication including live chat by available agent by selecting particular request for initiating a communication including live chat from the queued communication including live chats.

Systems and methods for facilitating electronic safety alert communications by a safety alert management system are disclosed herein. A method includes receiving an electronic safety alert for a specific safety event from a user electronic device via a safety alert application, the safety alert including at least one user message from a user associated with the user device. The method includes initiating an electronic chat session between the user and the safety agent attending the safety management application and, for at least one user message received at the safety management application, determining a reply message to send to the user device in response to the at least one user message. In embodiments, a machine learning model is used to analyze the user message and determine one or more recommended reply messages to display to the agent. A method for training a safety chat language model in a safety alert management system is also disclosed.

A jitter compensation during receipt of voice content (1, 2, 3, 4) via IP-based networks in voice packets (110, 120; 210) with a header (H) and payload (PL) is described, wherein a piece of time information of a transmitter time of a transmitter, which indicates the time of sending (ts) of the voice packet (110, 120; 210), is contained in the voice packet (110, 120; 210). It is provided that the receiver initializes a receiver clock using a transmitter time by means of the time information in the initial packet, that the receiver determines a minimum relative packet transmission duration (delta) after the initialization of the receiver clock, adjusts the receiver time as a function of the minimum relative packet transmission duration (delta), during the receipt of a first voice packet (111) with voice content, determines the actual time of receipt of this voice packet and determines the buffer (DJB) as a function of the actual time of receipt

Systems and methods detect and prevent vishing attacks through an integrated framework combining SIP header customization, STIR / SHAKEN frameworks, AI / ML analysis, and real-time speech analysis using the Viterbi algorithm. The system begins with call initiation, embedding authentication information in the SIP header. The SIP data is transmitted and verified using STIR / SHAKEN frameworks, ensuring the authenticity of the caller's identity. Verified data is cross-referenced with third-party databases and analyzed by an AI / ML engine to detect anomalies. If potential fraud is detected, the call is blocked, and the customer is notified. Calls that pass initial checks are further analyzed using the Viterbi algorithm, which converts speech to text and identifies suspicious patterns. An anomaly pattern detector processes the converted text to detect vishing indicators, terminating the call if a match is found. This multi-layered approach ensures robust protection against vishing, enhancing the security and reliability of voice communications while safeguarding users from fraud.

An example vehicle audio system includes a vehicle speaker configured to generate audio, multiple microphones, and a vehicle control module configured to receive an input signal via the multiple microphones, split the input signal into a parallel domain signal and an orthogonal domain signal, select a constant step size value for an orthogonal domain filter weight and a variable step size value for a parallel domain filter weight, adapt the orthogonal domain filter weight according to the orthogonal domain signal and the constant step size value, adapt the parallel domain filter weight according to the parallel domain signal and the variable step size value, combine the adapted orthogonal domain filter weight and the adapted parallel domain filter weight to define a total filter weight, and apply the total filter weight to the received input signal to perform a signal processing operation on the received input signal.

Disclosed are example methods, systems, and devices for allowing caller computing devices to authenticate calls via a service provider computing system. Users may opt to have entities register to contact the user with a positive ID, icon, or other notification on the user's computing device transmitted by the service provider computing system. A caller computing device may use a unique security token of the user to activate the notification on the user's device. The user device may be used to exert control over the security token via a service provider client application running on the user device. The caller computing device may initiate authentication via an API call to the service provider computing system. The caller computing device is able to have items (text, images, documents, etc.) delivered to the user computing device if authenticated.

To provide an intercom system capable of setting, by operation on a wireless master unit, the wireless master unit itself to a sleep mode.SOLUTION: An intercom system has a slave unit 1 with a function to call a resident and a master unit 2 with a function to respond to calls from the slave unit 1, and the master unit 2 includes a portable wireless master unit 31 in addition to a fixedly installed main master unit 21, and the wireless master unit 31 can respond to calls from the slave unit 1 by wirelessly communicating with the main master unit 21. The wireless master unit 31 has a sleep button 33a that sets the wireless master unit to a sleep mode in which no calling operation is performed. When the wireless master unit 31 is set to the sleep mode, no calling operation is performed even if the slave unit 1 is operated to call and a call signal is sent to the wireless master unit 31 via the main master unit 21.SELECTED DRAWING: Figure 1

Even when a speech is small, the speech is allowed to be detected and an echo is allowed to be appropriately suppressed. Whenever a sample point of a reception signal transmitted through a receiving signal path that transmits a signal to a speaker is acquired, an optimal mask is sequentially generated or selected from base masks as one or a plurality of masks generated based on a learning signal based on a reception signal acquired within a predetermined period before a time point at which the sample point was acquired. Whenever the optimal mask is selected, whether a double-talk state is present is sequentially detected based on a result of comparing an input signal with the optimal mask. When detecting that a speech is not input to a microphone and the reception signal includes a speech, a process of suppressing an echo is sequentially performed on the input signal.

Systems and processes are disclosed for real-time fraud detection and prevention in in-person transactions. The invention utilizes an AI / ML engine to analyze customer application data for inconsistencies and unusual requests indicative of potential fraud. Concurrently, a real-time conversation analysis engine with speech recognition algorithms monitors interactions between bank associates and customers, identifying suspicious speech patterns, hesitations, and keywords associated with scams. By combining insights from application data and conversational analysis, the system generates a comprehensive risk assessment. When a high probability of fraud is detected, an alert notifies the bank associate, security personnel, and other relevant individuals. This proactive approach enables immediate action to prevent fraudulent transactions, reducing manipulation risks and minimizing financial losses. The system continuously learns from new data, adapting to evolving fraud tactics, thus providing robust, long-term protection for financial institutions and their customers.

The invention relates to an MCX group call service penetration packet implementation method. The MCX server comprises a call control module, a speaking right control module and a media distribution module; a plurality of MCX clients, wherein each MCX client comprises a call processing module, a speaking right processing module and a media processing module; the speaking right control module and the media distribution module are configured to receive and process a UDP penetration packet and update address mapping; the speaking right processing module and the media processing module are configured to send UDP penetration packets and process speaking right and voice streams. According to the invention, the MCX client is enabled to acquire the extranet address and the port of the MCX server for processing the speaking right and the voice stream by using the call signaling established by MCX group call, the speaking right SSRC and the voice SSRC are respectively used for identifying the speaking right of the MCX client and the voice stream of the MCX client according to the MCX protocol, the MCX client is enabled to send the fixed format UDP penetration packet with the SSRC identification to the MCX server, and the UDP penetration packet is enabled to be transmitted to the MCX server. And an intranet penetration function of speaking right and voice between the MCX client and the MCX server is realized.

This application provides a call method, electronic device, and system; relating to the field of terminal technology, it can improve the call efficiency between terminals. The method includes: conducting a VoIP call with a second terminal; during the VoIP call with the second terminal, receiving a call request from a third terminal; and, if it is determined that the third terminal supports VoIP calls, sending a VoIP call request to the third terminal, wherein the VoIP call request is used to request the third terminal to join the VoIP call.

The present application discloses an active echo canceller for a hearing device, where the hearing device comprises an output interface configured to output a first sound based on a far-end signal, an input interface configured to provide a plurality of input audio signals indicative of the first sound and a near-end sound, and one or more processors, a filter configured to: filter a first filter input signal based on a far-end signal using a first filter to provide a first filtered signal, the first filter configured to estimate an echo signal originating from a first sound; combining the first filtered output signal with a first input signal to provide a first output signal; filtering a second filter input signal based on the plurality of input audio signals using a second filter to provide a second filtered signal, the second filter configured to attenuate the near-end sound; combining the second filtered output signal with the first filtered output signal to provide an update signal; one or more first filter coefficients of the first filter are updated based on the update signal.

Embodiments of the present disclosure disclose apparatuses, methods, and computer programs for acoustic echo cancellation (AEC) based on one or more diagonal regularized correlation matrices. At least some example embodiments may allow for an intelligent and efficient solution for multi-channel AEC and / or stereo AEC, providing improved performance and simpler implementation.

Methods and systems related to supporting real time communications are disclosed. In one type of a real time communication session, prior to an offer and acceptance to perform real time communications, a communication channel may be established between a caller device and a callee device. Information related to addresses of the caller and callee devices can be contained in, or pointed to, by a Universal Resource Locators (URLs) for the caller and callee devices. A network computing device may communicate messages relating to establishing and maintaining the communication channel between the caller and callee devices. In a scenario where the network computing device becomes unavailable, another network computing device can use the URLs and other channel information that may be stored in the database, to handle messages.

Model-based updating of triage data includes receiving problem analysis data associated with a computing system and determining one or more reference codes in the problem analysis data that are associated with one or more defects. A portion of the problem analysis data to be used as training data for a model is determined based on data usage associated with the one or more defects. An indication of each of the one or more reference codes within the portion of the problem analysis data is input to the model. The model is configured to output one or more data confidence scores based on the one or more reference codes. An association of the one or more reference codes to the one or more defects is updated based on the one or more data confidence scores.

This invention provides a nurse call system, a nurse call method, and a nurse call program that can reduce patient waiting times. [Solution] The nurse call system according to this disclosure includes: a display means that, when a nurse call button located in a patient's room is pressed, displays the name of the patient who pressed the nurse call button and the elapsed time on a nurse's terminal and a display terminal; and a notification means that, when the nurse call button located in the patient's room is pressed, first sounds a notification tone on a nurse's terminal held by the nurse in charge of the patient; if the receiver button on the nurse's terminal held by the nurse in charge is not pressed within a predetermined time, sounds a notification tone on a display terminal located in the nurse's station; and if the receiver button on the display terminal is not pressed within a predetermined time, sounds a notification tone on a nurse's terminal held by each nurse in the ward.

This application relates to the field of communication technology, specifically disclosing a SIP-based IPPBX voice switching management system and method. The method includes: authenticating users and assigning internal contact numbers through a voice management terminal; generating call paths based on a calling and called number matching routing strategy, and optimizing the SIP paths using a machine learning model; dynamically allocating media resources according to the call path and network conditions, initiating RTP stream transmission of voice data, and dynamically adjusting voice compression parameters; continuously monitoring the session status during the call and triggering corresponding processing mechanisms based on the voice session status. This application provides a secure, efficient, and intelligent IPPBX voice switching management method.

To provide an entrance system capable of remotely controlling the operation of equipment connected to the entrance system by using a voice messaging circuit.SOLUTION: An entrance system according to the present invention includes a communication circuit and a control circuit. The communication circuit includes a SIM (SubscriberIdentityModule) and obtains out-of-band DTMF (DualToneMulti-Frequency) signals from the communication terminal and converts the out-of-band DTMF signals into control signals having a format suitable for operation of the control circuit. The control circuit operates the entrance system or operates a device connected to the entrance system based on the control signal.SELECTED DRAWING: Figure 1

Disclosed herein are embodiments of a hearing device. The hearing device can filter a first filter input signal to provide a first filtered signal, where filtering is performed to estimate an echo in a first input signal. The hearing device can combine a first filtered output signal with the first input signal to provide a first output signal, the first filtered output signal is based on the first filtered signal. The hearing device can filter a second filter input signal to provide a second filtered signal, where the filtering is performed to attenuate a near-end sound. The hearing device can combine the second filtered output signal with the first filtered output signal to provide an update signal, where the second filtered output signal is based on the second filtered signal. The hearing device can update one or more filter coefficients based on the update signal.

The application discloses a system and method for non-invasively upgrading a customer service voice telephone into a video telephone, which comprises a video-assisted client module and a video-assisted server module; the video-assisted client module is non-invasively arranged on a customer service platform and is used as an entrance for a customer service to initiate a request for upgrading a current voice call into a video call and a terminal for the customer service to receive a video call; the video-assisted server module is arranged on an enterprise call center and is used for voice-upgrading a video call, and is connected with an SBC call module of the enterprise call center at an upper end and is connected with a PBX module of the enterprise call center at a lower end; during a voice call process, a call is sent to the PBX module, and a call flow remains unchanged with an original flow of the call center. The application retains an original call center architecture and a voice call flow, realizes a video call function and a scene, and maintains an original customer service usage habit unchanged, and only initiates a video service scene when the customer service triggers actively.

A method for rerouting communication in a unified communications network. The method comprises: receiving, at a disaster plan server, a mapping of an original telephony number to a substitute telephony number; storing, at the disaster plan server, the mapping between the original telephony number and the substitute telephone number as part of a disaster plan; receiving, at a disaster plan server, a request to activate the disaster plan; transmitting, from the disaster plan server to a routing server, a message to activate the disaster plan based on receiving the request to activate the disaster plan; revising, at the routing server, routing details to the original telephony number based on the disaster plan; and routing, at a session border controller, a call to the original telephony number to a new destination based on the revised routing details associated with the disaster plan.

Systems, methods, and other embodiments associated with concurrently joining a voice channel and a web channel are described. In one embodiment, a method includes establishing a voice session to communicate over an audio channel, where an artificial customer service communicates audio voice signals with a user. In response to identifying the question from the user, a navigation link is transmitted, where the navigation link, when activated, navigates the browser to a web page associated with the question. A web session is established to communicate between a browser and a web page. A voice session and a web session associated with a user are linked together. Since the two channels have been connected, the call controller may then communicate with the two channels simultaneously, allowing a manual customer service to disconnect from the audio channel.

The invention discloses a method (500) for a cellular communications system (100), in which traffic is sent in frames (200), each frame comprising a first number of subframes (201-210), with a second number of said subframes being available for at least either uplink or downlink traffic. At least one of said second number of subframes is made to comprise at least three parts (515), as follows: • One part (520) which is utilized for uplink traffic, • One part (525) which is utilized for downlink traffic, • One part (530) which is utilized as a guard period, with said guard period part (525) being scheduled between the uplink and the downlink parts. The duration of at least two of said three parts (520, 525,530) may be varied to fit the current system need.

A user of a user device initiates a contact center engagement with a first contact center. The first contact center processes a prompt received, by a first contact center software service associated with a first entity. Determine, by the first contact center software service and based on stored data associated with the prompt or an account of the contact center user device, that the prompt relates to a second contact center software service associated with a second entity. Train, using online learning techniques, an artificial intelligence agent of the first contact center software service based on prior responses received from the second contact center software service during a merged contact session. The artificial intelligence agent is capable of providing a resolution to a user of the contact center user device without contacting the second contact center software service after a response is provided to questions in the merged contact session.

To provide an inexpensive telephone facility that can perform advanced call control including calling a designated telephone handset by utilizing the transmission function of the transmission line of an R-type fire alarm facility for call control. [Solution] A fire detector with a unique address is connected to a transmission line drawn from the receiver to the security zone, and a telephone line drawn from the receiver connects a telephone base unit installed in the receiver to a telephone handset installed in the security zone. The receiver is equipped with a telephone base unit control unit that controls the call connection of the telephone base unit, and a telephone repeater with a unique address that is connected to the transmission line and the telephone line and controls the call connection of the telephone handset. The telephone handset is remotely connected to the telephone repeater via a voice line, and when the telephone repeater receives a predetermined call control signal via the transmission line from the telephone base unit control unit, it connects the telephone line and the voice line to control the call connection between the telephone base unit and the telephone handset.

The invention provides a translation box and a real-time conversation translation method based on the translation box, which are applied to a fixed-line telephone, and comprise a telephone host port used for being connected with a telephone host; the telephone receiver port is used for being connected with a telephone receiver; the audio acquisition module is used for acquiring an analog voice signal; the analog-to-digital conversion module is used for converting the analog voice signal into a digital voice signal; the digital-to-analog conversion module is used for converting the digital translation signal into an analog translation signal; the communication module is used for accessing the Internet and transmitting digital signals with a cloud server; the audio output module is used for sending an analog voice signal to the telephone host or the telephone receiver; and the main control module is used for setting a translation process and controlling the audio acquisition module, the analog-to-digital conversion module, the digital-to-analog conversion module, the communication module and the audio output module to execute the translation process. The translation box can realize a real-time translation function with zero transformation access, and cross-language real-time dialogue can be realized by installing the translation box on an original fixed-line telephone by a single party of a talker.

A virtual telephony platform may establish at least one communication connection with one or more multiple modem array devices that are deployed in one or more geographical regions. The platform may be provided with access to multiple Subscriber Identity Modules (SIMs). The platform may provision a unique device profile to a particular modem in a specific modem array device deployed in a particular geographical region through a communication connection. The platform may further provide the particular modem with a specific SIM profile of a particular SIM of the multiple SIMs through the corresponding communication connection, such that the particular modem is configured to use the specific SIM profile to access a base station of a wireless carrier in the particular region. The platform may further route telecommunication data between a client device and the wireless carrier network in the particular region through the particular modem.

This application provides a call process implementation method and an apparatus, applied to the field of communication technologies, to enrich visual and auditory experience of a user by adding an electronic gift service or a virtual item service in a calling phase or a ringing and waiting phase of the user, so as to improve user experience, and meet diversified requirements of network users. The method includes: A terminal receives a media data stream corresponding to a virtual item, where the terminal is a recipient of the virtual item; and the terminal plays media information corresponding to the virtual item in a call process, where the terminal is a terminal of a caller user of a call service or a terminal of a callee user of the call service.

To provide an intercom device that can effectively prevent theft when it is installed on a wall surface, and an installation structure for the intercom device.SOLUTION: A magnet holder 17 that holds a magnet 16 is attached to a fixed frame 13, and a magnetic sensor 11 capable of detecting the magnet 16 is built in a main body case 2 of a collective entrance device 1 at a position facing the magnet 16. Then, a control unit 7 that controls the main operation of the collective entrance device 1 constantly monitors the magnetism of the magnet 16 using the magnetic sensor 11, and when the positional relationship between the magnetic sensor 11 and the magnet 16 changes in such a case, for example, that the collective entrance device 1 is about to be removed from the fixed frame 13, activates a speaker 3 to emit a warning sound.SELECTED DRAWING: Figure 2

There is provided a method for real-time voice communication over a telecommunication network having a voice channel and a data channel, the method comprising receiving, at the first device, speech input from a first user. Generating voice input data, at the first device, based on the received speech input. Generating text data from the received voice input data. Generating voice output data from the generated text data, wherein the generated voice output data is generated based on a first user profile, wherein the first user profile is a profile associated with the first user. Processing, at the second device, the voice output data to generate speech output. Outputting, at the second device, the generated speech output, and monitoring the quality of the generated speech output. There is also provided a device and a system.