1192results 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.

A system and method for chatbot and search engine integration comprising traversing a conversational flow of a chatbot, generating a data structure based on the conversational flow, embedding the data structure in a webpage, and directing a search engine to the webpage for the search engine to assign a webpage rank to the data structure. In some embodiments, this integration may allow search engine users to be redirected to a non-public website hosting the chatbot when that chatbot would not otherwise be identified by the search engine.

Systems and methods for generating speaker video and audio in multiple languages for videoconferencing are provided. For example, a computing device can access a speaker speech audio signal that includes a speaker speech in a first language, a video of the speaker and a translated speech audio signal of the speaker speech in a second language. The computing device generates, based on the translated speech audio signal, a converted translated speech audio signal that includes a speech in the second language having voice characteristics in the speaker speech. The computing device further generates a lip-synched speaker video based on the video of the speaker and the converted translated speech audio signal. Lip movements in the lip-synched speaker video correspond to the converted translated speech audio signal. The converted translated speech audio signal and the lip-synched speaker video are transmitted to a video conference provider configured to host the video conference.

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.

Systems and techniques are provided for wireless communications. A process can include determining that a quantity of a plurality of Mobile Terminated (MT) calls received within a first time period is greater than a first configured threshold. A process can include determining a subset of non-verified MT calls from the plurality of MT calls based on a call verification score for each respective MT call of the plurality of MT calls received within the first time period. A process can include determining that a quantity of non-verified MT calls included in the subset is greater than a second configured threshold. A process can include performing modem-level blocking based on determining a corresponding calling party identity associated with a non-verified MT call included in the subset.

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.

A method includes receiving an audible response to a query, and prior to playing back the audible response, providing, for output from an acoustic speaker, an alignment output audio stream that encodes an audio watermark. The method also includes receiving an alignment input audio stream captured by a microphone array and encoding an acoustic echo of the audio watermark, processing the alignment input audio stream to detect the acoustic echo, and determining a time alignment value between the alignment output audio stream and the alignment input audio stream. The method also includes playing back a response output audio stream that encodes the audible response and receiving an input audio stream. The input audio stream includes acoustic echo corresponding to the audible response played back. The method also includes processing the input audio stream to generate a respective target audio signal that cancels the acoustic echo.

A three-dimensional sound generation system includes one or more processors of a computing device, including capability to receive sound tracks, each of the sound tracks comprising one or more sound sources, each of the one or more sound sources corresponding to one or more respective sound categories, receive or determine a first configuration in a three-dimensional space, the first configuration comprising a listener position and a computing device location relative to the listener position, determine a second configuration comprising a change to at least one of the listener location or the computing device location relative to the listener position, generate, using the one or more sound tracks and the second configuration, one or more channels of sound signals, and provide the one or more channels of sound signals to drive one or more sound generation devices to generate a three-dimensional sound field.

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 an exception handling method based on an IPPBX call queue, a medium, equipment and IPPBX.The method comprises the steps that when a call request enters the call queue, the states of all seats in the call queue are obtained, and available seats in the queue are screened out; recording dialing starting time and reading a dialing round of the current call in the queue; starting to dial available seats in the queue, and if the available seats in the queue are all in a non-response state, quitting dialing waiting; if the current time exceeds the preset maximum timeout time, resetting the dialing round and recording the dialing starting time again; otherwise, the dialing round + 1; judging whether the dialing round is greater than or equal to a preset dialing time threshold value or not; if yes, the current call enters a dormant state, and system resources occupied by the current call are released. By using the technical scheme, the normal operation of the IPPBX is ensured by solving the problem that the system resources cannot be released when the IPPBX call queue is in the abnormal scene.

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.

This application discloses a call method and system, and a related apparatus. When a specific condition is met (for example, a callee fails to be paged), a network device forwards, to a VoIP server, a first call request to a target number, where the first call request is a call request initiated by a calling terminal to the target number, and the target number has been subscribed to a call forwarding service. Then the VoIP server parses a forwarding message from the network device to obtain a called number, searches, based on the called number, for the called terminal associated with the called number, and initiates a VoIP call to the called terminal, so that the callee in an out-of-service state (for example, there is no signal of an operator network or the callee is in an airplane mode) can still receive the call request from the caller.

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.

The invention relates to a system and method for the management of decentralized digital identities, used in a distributed environment for making virtual transactions, document transfer and encrypted communication in a secure and integral way. The system works on the basis of blockchain technology and includes: a token-type hardware device (DH) capable of generating and storing encryption keys associated with decentralized identities (IDs), a mapping module (MAP) for correspondence between email addresses and decentralized digital identities (IDs), a communication module (MC) for blockchain-encrypted email and chat, a compliance module (MConf) for EU identity recognition, a decentralized identity module (MID) for identity management through blockchain and an authentication module (AUT) that combines two-step authentication with one-time authentication. The method involves enrolling users, generating decentralized identities, authenticating through blockchain, and making virtual transactions using decentralized identities in a distributed system.

Described is a method comprising receiving a rerouted phone call identified as a scam call, processing received caller speech from the rerouted phone call to determine a response, interacting with a caller using the determined response, and processing at least a part of the call conversation to determine one or more scam parameters. The response is determined in order to extend a duration of a conversation. The scam parameters may include one or more of the following: a scam target, a scam structure, a scam technique, a financial instrument, scammer phone number, scammer voice prints, classification of background noise during scam and / or a scam classification. The processing may further comprise identifying features in the received call speech associated with ending and / or extending a call.

Embodiments described herein provide for automatically authenticating telephone calls to an enterprise call center. The system disclosed herein builds on the trust of a data channel for the telephony channel. Certain types of authentication information can be received through the telephony channel, as well. But the mobile application associated with the call center system may provide additional or alternative forms of data through the data channel. The system may send requests to a mobile application of a device to provide information that can reliably be assumed to be coming from that particular device, such as a state of the device and / or a user's response to push notifications. In some cases, the authentication processes may be based on quantity and quality of matches between certain metadata or attributes expected to be received from a given device as compared to the metadata or attributes received.

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.

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.

The invention discloses a hard phone scheduling method and system, a storage medium and computer equipment. The method comprises the following steps: receiving state information of a hard phone based on an internet protocol network, and pushing the state information to a soft operation platform; based on an internet protocol network, forwarding an operation signaling sent by the soft operation platform based on the state information to the hard phone so as to schedule the hard phone; wherein the soft operation platform is connected with the internet protocol network, and the hard phone is connected to the internet protocol network based on the external exchange gateway. Through the method, the entity operation of the hard phone can be seamlessly integrated on the soft operation platform for software implementation, the state monitoring and scheduling operation of the hard phone can be directly realized through the soft operation platform, the scheduling operation efficiency and convenience can be effectively improved, and the scheduling accuracy and real-time performance are guaranteed.

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.

Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for a virtual intercom. In some implementations, a request from a visitor device to communicate with an occupant of a particular unit of the facility is received. A location of the visitor device is received. Based on the received location, it is determined that the visitor device is within a predetermined bounded area. An occupant profile is identified based on the received request. Information associated with the occupant profile is obtained. Based on the obtained information, it is determined the occupant has registered a profile. Based on the obtained information, a communication path to an occupant device is identified. The request from the visitor device is sent, using the communication path, to an occupant device belonging to the occupant. A response from the occupant device is received. Based on the response, an action is performed.

The invention discloses a converged communication system. The converged communication system comprises a core module and a function module which is in communication connection with the core module and comprises a coding and decoding module, a terminal protocol module and a dialing routing module, and the coding and decoding module is used for supporting conversion between different audio and video coding and decoding formats and a preset format; the terminal protocol module is used for supporting docking between different communication protocols and a preset protocol; the dialing routing module is used for determining a corresponding target terminal and a routing decision based on a call request of a source terminal; and the core module is used for scheduling the coding and decoding module, the dialing routing module and the terminal protocol module so as to carry out calling and state control on the source terminal and the target terminal. By means of the system, on the premise that existing terminal equipment functions and interaction logic are not damaged, seamless butt joint between terminals of different systems can be achieved, communication efficiency is improved, stability and flexibility of the converged communication system are ensured, and the requirement for multimedia communication can be met.

Systems, methods, and computer program products for echo cancellation with prediction filter adaptation and detection of wideband offset between a reference signal (available to the echo canceller) and an output signal (unavailable to the echo canceller), where the output signal has been generated by applying at least one level shift to the reference signal, e.g. such that the level shift is unknown to the echo canceller.

To allow an inpatient to call a nurse when his or her condition suddenly get worsened or when he or she cannot move his or her body.SOLUTION: Provided is a nurse call system for inpatients. The system comprises: means for detecting sounds made by the patient; means for detecting facial expressions of the patient; means for determining whether or not the patient is abnormal; means for automatically contacting, upon detecting an abnormality of the patient, a nurse center, sending a voice or text information on the abnormality; and means for using a small voice or facial expression during sleeping, etc. when determining whether or not the patient is abnormal.SELECTED DRAWING: Figure 1

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.