19results about How to "Improve call experience" patented technology

The invention relates to the technical field of voiceprint recognition, in particular to an anti-fraud communication method based on voice change recognition. The method comprises the following steps: acquiring multi-dimensional voice feature data of a user, wherein the multi-dimensional voice feature data comprises timbre, frequency features, language rhythm, emotion features and pause modes; performing voice change possibility prediction according to the multi-dimensional voice feature data to obtain voiceprint feature prediction data; performing deep learning model training according to the voiceprint feature prediction data, and mapping voiceprint states and features under different call conditions to obtain voiceprint feature map data; and according to the voiceprint feature map data, carrying out simulation analysis on voice change recognition behaviors in different call states so as to obtain voice change recognition simulation data. According to the method, the call environment noise data and the network quality data are obtained in real time, the call quality model is constructed, cross-scene optimization and real-time adjustment are carried out, and the influence of the environment on recognition is effectively reduced.

The invention discloses a call method and device, and belongs to the technical field of interaction. The call method comprises the following steps: determining a first role type of a call object, and obtaining first role information of the call object; according to the first role type, screening out a first language model corresponding to the first role type from the first model set; determining a first prompt word based on the first role information; inputting the first cue word into a first language model to obtain a first response result output by the first language model; and making a call with the call object based on the first response result.

The embodiment of the present application discloses a method and system for adjusting the audio bit rate of audio and video transmission. The technical solution provided by the embodiment of the present application obtains the congestion information of the current audio and video transmission, determines based on the congestion information that the current audio encoding bit rate is in a bit rate gear adjustment state, obtains the target bit rate of the audio and video transmission statistics of the current acquisition period, and determines the bit rate gear of the current audio encoding bit rate. Based on the target bit rate and the bit rate gear of the current audio encoding bit rate, the bit rate gear of the current audio encoding bit rate is adjusted, and the audio encoding of the current audio and video transmission is performed based on the audio bit rate value corresponding to the adjusted bit rate gear. The above-mentioned technical means are adopted to adjust the audio bit rate by combining the network congestion situation and the target bit rate of the audio and video transmission, and the audio bit rate is adaptively adjusted according to the real-time network situation, so as to adaptively provide a reasonable audio bit rate service, optimize the audio and video call effect, and improve the audio and video call experience.

The utility model discloses an earphone which comprises a sound production part, an abutting part and an ear hook, the ear hook is connected with the sound production part and the abutting part, in a wearing state, the sound production part and the abutting part form a clamping state on the two sides of a helix, and the sound production part is located in an auricular conchae cavity; the sound production part comprises a shell, a microphone and a sound production assembly, the microphone and the sound production assembly are arranged in the shell, a sound inlet hole and a sound outlet hole are formed in the shell, the microphone collects external sound through the sound inlet hole, the sound produced by the sound production assembly is transmitted outwards through the sound outlet hole, the sound inlet hole is provided with a sound inlet end located on the outer wall face of the shell, and the sound outlet hole is provided with a sound outlet end located on the outer wall face of the shell. The sound outlet hole is provided with a first sound outlet end located on the outer wall face of the shell, a first shortest straight line segment is arranged between the hole edge of the sound inlet end and the hole edge of the first sound outlet end, and the length of the first shortest straight line segment is larger than or equal to 9 mm. Through the above mode, the possibility of echo when a user uses the earphone to carry out a call can be effectively reduced.

The invention relates to the technical field of Bluetooth headset equipment, in particular to a vehicle-mounted Bluetooth headset communication system integrated with a Hall sensor based on the application of the Hall sensor. The equipment comprises a detachable Bluetooth earphone and a charging base, wherein the Bluetooth earphone is electrically connected with the charging base and an automobile electrical system. The control device arranged in the Bluetooth earphone and the charging base communicates with the mobile phone through Bluetooth, so that when the mobile phone receives an incoming call, a user can answer the call only by taking out the Bluetooth earphone from the charging base, the convenience in use and the safety in driving are enhanced, and meanwhile, the privacy of the call is increased.

The invention relates to the technical field of communication, in particular to a communication method, electronic equipment and a computer readable medium. The method comprises the following steps: the user equipment is in a communication connection state and does not detect a data communication service; the user equipment starts a first timer; when the user equipment determines that the timing duration of the first timer is greater than the first duration, the user equipment enters an idle state, and a first time difference between the first duration and the second duration of the second timer of the network equipment connected with the user equipment is greater than a time difference threshold, and the network equipment sends a connection release message to the user equipment when the timing duration of the second timer is greater than the second duration. Thus, the problem that the user equipment cannot receive the paging message and misses an incoming call due to the fact that the user equipment state inquired by the network side is not synchronous with the actual state of the user equipment can be avoided, and the call experience of a user can be improved.

This application provides a mobile call method and an electronic device, which relate to the field of communication technologies and can reduce a possibility that a mobile call is disconnected by mistake and improve call experience of a user in a process in which two electronic devices perform the mobile call by using a preset call function. The method is applied to a first electronic device, and the first electronic device establishes a first wireless communication connection with a second electronic device. The method includes: establishing, by the first electronic device, a mobile call connection with a third electronic device and performing a first mobile call with the third electronic device by using the mobile call connection; when a preset call function is enabled, supporting, by the first electronic device by using the mobile call connection and the first wireless communication connection, the second electronic device to perform a second mobile call with the third electronic device; and in response to a first event, stopping supporting, by the first electronic device, the second electronic device to perform the second mobile call with the third electronic device, and performing a third mobile call with the third electronic device by using the mobile call connection.

The invention provides an acoustic echo cancellation method for two-way communication. By adopting the echo cancellation method provided by the invention, the neural network model is utilized to carry out end-to-end learning and modeling on the input signal and the residual echo signal, so that more accurate and effective echo cancellation is realized. After a near-end signal and a far-end signal are subjected to time delay alignment and linear echo cancellation, part of noise and echo still may remain in the near-end signal. Through the neural network, the residual components can be better eliminated, and the residual problem which affects the hearing feeling is avoided. The method combines the advantages of a traditional method and the capability of a neural network technology, can better adapt to a complex call environment, reduces the calculation cost of a system, and provides higher-quality call experience.

An embodiment of the present application provides a call forwarding method, a call center and a call forwarding device, the method comprising: generating a first information list and associating a user's identifier, when the call center, as the calling party, calls the user and completes the call, generating a first history record according to the content of the call; updating the first information list according to the first history record; when the call center, as the called party, receives a call from the user, obtaining the user's first information list according to the user's identifier; generating a voice service according to the first information list, the voice service comprising prompting the user to handle related services; and transferring the call to at least one of a plurality of manual seats of the call center according to the user's feedback information on the voice service, thereby improving the user's calling experience.

The invention provides a call data processing method and device, equipment, a medium and a product, and the method comprises the steps: determining a data processing task type corresponding to a call theme of call voice data according to the call voice data of two call parties in a call process, the data processing task type comprises at least one of a content generation task and an information retrieval task; performing corresponding data processing on the call voice data according to the data processing task type to obtain a data processing result; and outputting a data processing result to the terminal devices of the two parties of the call. The content of the data processing result is richer through diversified task types, so that the information processing capability in the call process is improved, and richer interaction modes are provided for the call process.

The invention provides a sensitivity-based user label evaluation method and device and a server, and relates to the technical field of information processing, and the method comprises the steps: obtaining a mobile phone number of a user side and a user call record corresponding to the mobile phone number; carrying out keyword analysis processing on the user call record, determining dialing industry information, carrying out sensitivity analysis processing on each piece of dialing industry information so as to determine the industry sensitivity of the industry to which each dialing enterprise belongs, and obtaining the number of calls between the dialing enterprises in the industry and the user side within a preset time interval; and performing label grade evaluation processing on the user side by using the call times and the industry sensitivity so as to determine a target label of the user side. According to the invention, label evaluation can be carried out on the user according to the industry sensitivity, the accuracy of user evaluation is improved, and the user call experience is effectively improved.

The present application discloses a call processing method and a call processing device, belonging to the technical field of data calls. The call processing method is used for a terminal, and the method comprises: receiving a voice data packet sent by a network server; in the case of an abnormality in the voice data packet, sending a first signaling to the network server, wherein the first signaling is used to enable the network server to switch to at least one of a media server for generating the voice data packet and a domain for transmitting the voice data packet.

This application belongs to the field of financial technology and is applied to the field of insurance pricing guidance in financial business. It relates to a method and related equipment for recommending call leads in auto insurance business based on NLP. The method includes: recognizing recorded human-computer dialogue; performing text-to-text processing to dynamically acquire dialogue text data in real time; using NLP (Natural Language Processing) to process the dialogue text data; obtaining customer intent clustering results; recognizing the dialogue intent of the calling customer; and selecting the corresponding call lead based on the dialogue intent and assigning it to the next agent after the current human-computer dialogue, thus completing the call lead recommendation for the agent. This application uses an ASR+Kafka+dynamic data stream acquisition framework+NLP approach, enabling human customer service representatives to quickly locate the dialogue intent of customers previously served by the AI ​​voice service assistant when faced with lengthy AI intelligent outbound call data records. This improves the efficiency of human agents and ensures a good call experience for customers.

The utility model belongs to the technical field of telephone communication, in particular to an anti-interference SIP telephone communication structure, which comprises a core layer, an edge layer and a user layer, and is characterized in that the core layer comprises an SIP server and a media server and is used for processing SIP signaling and media streams; the edge layer comprises a firewall / NAT (Network Address Translation) traversal device and a load balancer, and is used for protecting system security and realizing load balancing; the user layer comprises an SIP telephone terminal and client software and is used for initiating and receiving telephone calls and managing information such as user accounts. According to the utility model, through multiple measures such as encryption and authentication mechanisms, call filtering and blocking functions, network quality detection and optimization, load balancing and fault-tolerant mechanisms and the like, the security and call quality of the system are effectively improved, unnecessary interference is reduced through call filtering and blocking, and the service life of the system is prolonged. And the availability and stability of the system are improved by using load balancing and fault tolerance, meanwhile, voice quality monitoring and coding transmission are optimized, and high-quality call experience is provided for users.

This application discloses a control method, a display device, and a storage medium. The control method is applied to the display device, which includes at least a first display area and a second display area. The first display area and the second display area are rotatable relative to a rotatable component. The control method includes: displaying a first call interface in the first display area and displaying a second call interface in the second display area; responding to a first operation command on a first position of the first display area, controlling the first call interface to switch from a first state to a second state, and controlling the second call interface to switch from a third state to a fourth state, wherein the first state and the fourth state indicate that the corresponding call interface is in a normal call state, and the second state and the third state indicate that the corresponding call interface is in a call hold state.

The invention relates to the technical field of interphones, in particular to a dustproof interphone which comprises an interphone body, a hook, an antenna, a control area and an intercom button. The dust blocking structure is optimally arranged and located on the front side of the horn opening and the microphone opening, dust, sand grains and other impurities can be effectively prevented from entering the microphone and the horn, sensitive electronic components are protected, the service life of equipment is prolonged, the size of the opening of the dust blocking structure can be adjusted, and the practicability is high. The invasion of dust and impurities can be reduced during sound transmission; the supporting structure is optimally arranged and can be pressed into the rear part of the interphone body after being stored and folded, so that the area exposed outside is reduced, the interphone body is convenient to take and use, and when the interphone body needs to be placed, the supporting structure can be pulled out from the rear side of the interphone body, and the bottom of the supporting structure is supported at the placing position; and the position of the supporting part is flexibly changed, so that the interphone body is placed at different angles, the interphone body can be conveniently fixed when being used on a desktop by a user, and better call experience is provided.

The invention belongs to the field of video calls, and relates to a video call intelligent switching method and system based on an IMS system and a readable medium, and the method comprises the steps: detecting the hardware state of each media processing server in real time in a video call process; judging whether the hardware state exceeds a threshold value or whether the video call quality is reduced; if the hardware state exceeds a threshold value or the video call quality is reduced, a healthy media processing node is selected from a load balancing cluster through an intelligent switching strategy; ending a reply on the original media processing node, and re-establishing a new media stream on the new media processing node; and the session information on the original media processing node is migrated to the new media processing node, and the session state of the original node is recovered on the new media processing node, so that the intelligent switching of the video call is realized. The state of the media processing server is monitored in real time, and the healthy service node is intelligently selected to continue the video call when the problem is detected, so that the continuity and stability of the video call are ensured.

The present invention is applicable to the field of VR call technology, and provides a VR call method and system based on blockchain technology. The present invention obtains basic background video; obtains environmental shooting data, determines multiple target environmental entities; performs real-time call shooting, obtains call shooting data; obtains the current queue number, calculates the standard processing number; when the current queue number is less than the standard processing number, performs fine rendering edge processing and transmission; when the current queue number is not less than the standard processing number, performs basic rendering edge processing and transmission. It can combine local processing with edge computing, and monitor the data queue and processing performance of local call equipment and network edge equipment. In different states, fine rendering or basic rendering edge processing and transmission are performed respectively, so as to achieve low-latency, high-quality VR calls, effectively avoid call freezes, blurred images and other phenomena, and improve the user's call experience.