645results 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.

The invention relates to the field of communication, in particular to a railway electronic speed controller straight-through telephone protocol converter based on analog-to-digital conversion, which comprises seven parts, namely a network address conversion module, an Ethernet switching module, an SIP (session initiation protocol) processing module, a voice coding and decoding module, a TDM (time division multiplexing) switching module, an analog signaling processing module and a Z interface module, an IP telephone, an IAD or an ONU are connected through an IP network, the Z interface module provides a telephone line interface externally and is used for being connected with a digital dispatching switch, all the internal modules are matched with one another, conversion between analog signals and digital signals is completed, and the function of transmitting and dispatching direct-through telephone services through the IP network is achieved. The method has the advantages that the railway dispatching system is matched with the data network and the PON network, so that stable and reliable electronic speed controller straight-through telephone service is realized, and cost reduction and benefit increase are really realized; business protection is achieved through various protection means of a railway dispatching system, and business safety is guaranteed.

An apparatus may comprise a memory communicatively coupled to a processor. The processor may be configured to receive a request to perform multiple data processing operations and determine multiple sub-dialogues corresponding to the request. The sub-dialogues are part of a plurality of IVR operations. The processor may be further configured to route the request to a microservice of the microservices based at least in part upon the sub-dialogues. The microservice may be configured to fulfill the request. The processor may generate a response to the request associated with the microservice, update publishing commands in accordance with one or more rules and policies in response to generating the response, and publish the first microservice. The publishing commands associated with the microservice may be updated in isolation from the rest of the microservices.

The invention discloses a voice optimization method and device for internet protocol voice communication, equipment, a storage medium and a program product, and relates to the technical field of data processing, and the voice optimization method for internet protocol voice communication comprises the steps: obtaining network features and voice data of a voice communication network; performing network jitter prediction based on network features, and dynamically adjusting an adaptive jitter buffer based on a prediction result; performing data enhancement processing based on the voice data to obtain enhanced voice data; caching enhanced voice data based on the dynamically adjusted adaptive jitter buffer area; and encoding and decoding the enhanced voice data in the adaptive jitter buffer area to obtain optimized voice data. Through processing of network characteristics and voice data, voice optimization of Internet protocol voice communication can be carried out more comprehensively. By adjusting the adaptive jitter buffer area, high adaptation to a complex network environment is realized, and the quality of voice communication is effectively guaranteed.

The invention discloses a voice packet loss processing method and device based on deep learning and variational mode decomposition, equipment and a storage medium, and relates to the technical field of communication, and the method comprises the steps: extracting the multi-dimensional features of a received voice signal; performing packet loss detection on the multi-dimensional features by using a space-time convolution automatic encoder network to generate estimation features of lost frames, and reconstructing the voice signal based on the estimation features to obtain the voice signal after interpolation reconstruction; performing variational mode decomposition on the speech signal after interpolation reconstruction to obtain an effective component; according to the method, the effective component is reconstructed to obtain a filtered voice signal, post-processing optimization is carried out on the filtered voice signal to obtain an optimized voice signal, and by fusing the feature learning capability of deep learning and the adaptive signal processing advantage of variational mode decomposition, bandwidth consumption is reduced, voice communication quality in a packet loss scene is improved, and voice communication quality is improved. And the requirements of real-time, high-efficiency and high-quality VoIP communication are met.

The invention provides a communication method and device of VOIP equipment, equipment and a storage medium. The method comprises the steps of collecting and encoding voice data, detecting encoded data features and a network environment state, generating a feature vector set and a channel stability index, and sending the feature vector set and the channel stability index to an access point; receiving network resources and scheduling information allocated by the access point, and processing the encoded voice data packet to form a VOIP data packet to be transmitted; sending the VOIP data packet to other VOIP equipment through a wireless channel, coordinating transmission resources with other equipment, sharing the wireless channel, and adjusting transmission parameters according to the interference condition; and monitoring the transmission performance of the sent data packet, performing call quality evaluation, and recording the transmission performance and a quality evaluation result. According to the invention, through dynamic adjustment of a transmission strategy, cooperative transmission and comprehensive quality evaluation, the performance and user experience of VOIP communication in a high-density wireless network environment are significantly improved.

An integrated audio microphone system simultaneously provides in-room amplification and remote communication during a collaboration session with one or more local participants in a shared space and one or more remote participants. The integrated audio microphone system includes at least one microphone array and speaker system located in the shared space, at least one microphone system for amplification located in the shared space, an audio processor connected to the microphone system and the microphone array and speaker system, and a computer connected to the audio processor. The audio processor is configured to generate a plurality of virtual microphone bubbles and to amplify received audio signals from the microphone system for in-room audio amplification. The amplified audio signals are output to speakers of the microphone array and speaker system. Audio signals from the microphone system and the arrays of microphones are output to the computer to send the audio signals to one or more remote participants.

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.

Characteristics of a voice call in a telecommunications network are identified. A voice call that is in process in the telecommunications network is accessed and, for a specified segment of the voice call, a sample of the voice call is analyzed. Based on the analysis, a prompt is generated for input to a machine learning model. The prompt is usable to prompt the machine learning model to determine a likelihood that the voice call meets one or more characteristics. The prompt includes an example of a different voice call that meets the one or more characteristics.

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.

A method including: receiving a user report dataset indicating fraudulent activity corresponding to a phone number; responsive to receiving the user report dataset, identifying a record in a database corresponding to the phone number; and tagging the record to identify the phone number as being associated with fraudulent activity.

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.

A method performed by a first device, which includes performing an audio call with a second device by transmitting a microphone signal as an uplink signal and receiving a downlink signal for driving a first speaker and while performing the audio call, performing a joint media playback session in which both devices independently stream a piece of media content for synchronous playback such that both devices receive an audio signal of the piece of media content for driving respective speakers at the same time, determining that a voice activity detection (VAD) signal indicates that the downlink signal includes speech, in response to determining that the VAD signal indicates that the downlink signal includes speech, processing the audio signal of the piece of media content by applying a scalar gain, and driving the first speaker with a mix of the downlink signal and the audio 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.

To unlock auto-locks at entrances, or the like.SOLUTION: The building is provided with an indoor intercom master unit 7, a first button pressing device 5 for pressing a talk button, a second button pressing device 6 for pressing an unlock button, and an indoor communication terminal 2 for sending operation commands to the first and second button pressing devices 5, 6. A management server 1 is capable of communicating with the indoor communication terminal 2. The management server 1 causes a user terminal 3 to display a first-stage display for unlocking the auto-lock, and when an operation is performed on the user terminal 3 in response to the first-stage display, causes the user terminal 3 to display a second-stage display related to operation of the intercom. After the user operates the intercom in accordance with the second-stage display, the management server 1 receives an unlocking command from the user terminal 3, and upon receiving the unlocking command, sends an instruction to the indoor communication terminal 2 to drive the first button pressing device 5 and the second button pressing device 6.SELECTED DRAWING: Figure 1

The invention relates to the technical field of communication, and particularly discloses an IPPBX voice exchange management system and method based on SIP, and the method comprises the steps: carrying out the identity verification of a user through a voice management end, and distributing an internal contact number; generating a call path based on a calling and called number matching routing strategy, and optimizing an SIP path by using a machine learning model; dynamically distributing media resources according to a call path and a network condition, starting RTP stream to transmit voice data, and dynamically adjusting voice compression parameters; and continuously monitoring the session state in the call process, and triggering a corresponding processing mechanism according to the voice session state. The invention provides a safe, efficient and intelligent IPPBX voice exchange management method.

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.

Provided are an audio signal processing method, a training method, an apparatus and a storage medium, relating to the field of data processing, in particular to, the field of voice. The audio signal processing method includes: eliminating at least pail of a linear echo signal from a mixed voice signal, to obtain an intermediate processing signal, where the mixed voice signal is obtained by mixing a target voice signal with an echo signal, and the echo signal is generated in an environment where the target voice signal is located and includes the linear echo signal and a nonlinear echo signal; and removing the nonlinear echo signal and a residual part of the linear echo signal from the intermediate processing signal, by using a target full convolution neural network model, to obtain an approximate target voice signal, the target full convolution neural network model including at least two convolution layers.

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.

A jitter compensation during reception over an IP-based network of audio content (1, 2, 3, 4) in an audio packet (110, 120; 210) having a header (H) and a payload (PL) is described, in which one time information of a transmitter time of a transmitter is included in the audio packet (110, 120; 210), indicating a transmission time (ts) of said audio packet (110, 120; 210). The following is provided: a receiver initializes a receiver clock by using the transmitter time according to the time information in an initial packet, the receiver determines a minimum relative packet transmission duration (delta) after initialization of the receiver clock, adjusts the receiver time according to the minimum relative packet transmission duration (delta), during reception of a first audio packet (111) having audio content, determines an actual reception time of this audio packet, and determines a buffer (DJB) according to the actual reception time.