150 results about "WebRTC" patented technology

The invention relates to the technical field of content transmission methods, in particular to a content transmission method based on a multi-modal streaming media fusion technology system, which comprises the following steps: S1, multi-source data acquisition and time-space alignment: adopting an IEEE 1588PTP protocol, deploying a hardware clock synchronization module at a camera, a microphone and IMU equipment, improving the timestamp precision to a mu s level, and performing time-space alignment on the camera, the microphone and the IMU equipment; carrying out sampling rate normalization on the sensor data; s2, feature level cross-modal fusion: extracting each modal feature by using a lightweight Transform model, executing cross-modal attention calculation in an edge server, and generating a fusion feature tensor Ffusion belonging to RN * D; the system content transmission method based on the multi-modal streaming media fusion technology solves the problems that a traditional multi-modal transmission system usually adopts a separated transmission architecture, video streams are transmitted through an RTMP protocol, audio streams are encoded through Opus and are transmitted through WebRTC, sensor data are sent through an MQTT protocol in a JSON format, ABR only adjusts the video code rate, and the transmission efficiency is low. And joint optimization of multi-modal data is not coordinated.

The invention belongs to the technical field of high-power microwave effect tests in laboratories, and relates to a lossless reinjection data synchronization method and device for a multi-mode test body, and the method comprises the steps: S1, data collection and transmission: collecting electric field data in the multi-mode test body, and transmitting the electric field data out of a shielding cabin of the multi-mode test body; s2, an edge processing step: sending the electric field data transmitted to the outside of the shielding cabin into an edge GPU node, and carrying out voxelization processing and WebGL body rendering; s3, a data synchronization step: synchronizing the three-dimensional rendering data stream to a plurality of user side browsers in real time through WebRTC and HTTPS (Hyper Text Transfer Protocol over Secure Socket) protocols; s4, time synchronization is carried out, a unified high-precision time reference is provided through a Whit-Rabbit clock, and sub-nanosecond alignment of multi-channel data is achieved; and S5, a data storage step: generating a hash abstract for the synchronous data and writing the hash abstract into the block chain. By means of the method, the safety, the real-time performance and the reliability of data transmission can be effectively improved.

The invention discloses a WebRTC voice enhancement system and method based on a multi-modal large model, and belongs to the field of artificial intelligence and real-time communication cross technology, and the system comprises an audio and video collection module which is used for synchronously collecting original voice signals and corresponding video image data of a user side through a WebRTC protocol stack, high-precision alignment is realized through timestamp marking and a cache mechanism; the multi-modal feature extraction module is used for extracting audio features from the voice signals, extracting visual lip movement features from the video images and generating text semantic features through a voice recognition engine; a noise matching and updating module; a multi-modal semantic perception enhancement module; an audio reconstruction module; and a WebRTC integration module. According to the invention, high-precision noise suppression and millisecond-level delay are realized while the voice semantic integrity is guaranteed, and the requirements of industrial inspection, vehicle-mounted communication and other scenes on high fidelity, low delay and strong robustness are met.

Method and system for testing network latency, comprising a browser of a computer within a local network, downloading a first WebRTC client application. The first WebRTC client application making a connection over the internet with a control server external to the local network. The control server transmitting data causing a communication connection to be set up between the first WebRTC client application and a second WebRTC client application. Transmitting a plurality of data packets between the first WebRTC client application and the second WebRTC client application and back to an originating WebRTC client application, wherein one of the first WebRTC client application and the second WebRTC client application is the originating WebRTC client application. The originating WebRTC client application determining a round-trip time of the plurality of data packets.

The invention discloses a WebRTC-based vehicle-mounted video stream secure transmission management method, and relates to the technical field of real-time communication, and the method comprises the steps: 1, constructing a hierarchical security system, and building a basic security framework; the layered security system comprises an application layer, a transmission layer, a network layer and a physical layer; 2, according to the basic security framework, in the WebRTC interactive connection establishment process, zero-knowledge proof is introduced, access strategy dynamic updating is achieved through a block chain, an authentication access control strategy set is output, and the authentication access control strategy set comprises device access, a bidirectional identity authentication rule and a block chain-based fine-grained access control strategy; 3, on the basis of the authentication access control strategy set, data security is guaranteed through hybrid encryption, the integrity of the Hash tree is verified, and an improved encryption transmission scheme is output; wherein the Hash tree is verified, and anti-jitter and code rate self-adaption are carried out in combination with the LDPC and the DRL.

The invention provides a low-delay multi-channel video data processing method and system for remote driving, and is applied to the field of data processing. According to the invention, a low-delay multi-channel video data full-process processing system is constructed for remote driving of a target unmanned vehicle. And four types of core data including hardware, video, network and control are obtained. Aiming at multiple cameras, images are screened according to driving scene priorities, full-coverage dead-corner-free, core view-angle-first and vehicle-type adaptation constraints are applied, and preprocessed multiple video streams are generated. Encoding, slicing and decoding the preprocessed video stream to construct a low-delay processing mechanism; a customized WebRTC protocol is adopted, routing is optimized, direct connection between a vehicle end and a cockpit is achieved, meanwhile, the network state is monitored, and transmission stability and safety are guaranteed through multiple constraints when the network state is abnormal. Fusing multiple paths of decoded pictures, arranging and superposing multiple types of key information according to driving habits, and combining constraints to generate an offering cabin display result; and finally, cooperatively processing the control instruction and the video feedback to form an interactive monitoring visual interface.

Disclosed in embodiments of the present disclosure are an unmanned aerial vehicle-based live video streaming method and system, a device, and a storage medium. The method includes: acquiring original video stream data sent by an unmanned aerial vehicle; encoding / decoding the original video stream data to obtain output buffer data; converting the output buffer data into video frames supported by the WebRtc protocol; and sending the video frames to a preset server through a WebRtc transmission channel, so that the preset server performs live streaming display according to the video frames.

The invention discloses a safety monitoring system suitable for fireman training, relates to the technical field of fire-fighting training safety, and aims to solve the problems of difficulty in personnel positioning, communication distortion delay, low video monitoring efficiency and untimely safety intervention in existing fire-fighting training. The system comprises a UWB-based personnel positioning subsystem, a WebRTC-based real-time communication subsystem and a training guidance management subsystem. Wherein the UWB personnel positioning subsystem realizes accurate positioning and trajectory tracking of different training spaces, the WebRTC real-time communication subsystem realizes multi-mode real-time voice interaction and voice-text conversion, and the training guidance management subsystem realizes positioning display, communication scheduling, video linkage and alarm control through front and rear end cooperation. According to the system, the safety monitoring precision and the scheduling efficiency in fireman training can be improved, the training process backtracking and resetting are supported, and technical support is provided for the safety guarantee and efficiency improvement of fire training.

The invention discloses a Web honeypot automatic script identification and countering method based on a large language model and related equipment, and the method comprises the steps: obtaining a real IP address of an attack request through a WebRTC protocol; analyzing the attack request through a headless browser and a large language model, judging whether the attack request is an automatic script behavior or not, countering the attack request if the attack request is the automatic script behavior, and guiding the attack request to a honeypot if the attack request is not the automatic script behavior. The method can improve the accuracy of automatic script recognition, improves the honeypot stability, and can be widely applied to the technical field of network security.

The present application belongs to the technical field of remote control of robots, and particularly relates to a 5G cross-domain low-latency remote control system and method for an explosion-proof humanoid robot. The system comprises a master end teleoperation intelligent cockpit and a slave end remote explosion-proof humanoid robot, and the two are connected through a 5G network for information interaction. The system transmits video streams through H.264 encoding and WebRTC protocol, packs sensor data through JSON, and independently processes information for humanoid control and chassis control. The present application can realize accurate remote force feedback and control, with a humanoid control latency within 100 ms and a chassis control latency within 150 ms, effectively solving the stability and latency problems of remote cross-domain control and ensuring the personal safety of operators in dangerous operating environments.

The invention belongs to the technical field of robot remote control, and particularly relates to an anti-explosion humanoid robot 5G cross-domain low-delay remote control system and method. The system comprises a master-end teleoperation intelligent cabin and a slave-end remote anti-explosion humanoid robot which perform information interaction through a 5G network; the system transmits a video stream through H.264 coding and a WebRTC protocol, sensor data are packaged through JSON, and human shape control and chassis control information are independently processed. According to the invention, accurate remote force feedback and control can be realized, the time delay of human shape control is within 100ms, the time delay of chassis control is within 150ms, the problems of stability and time delay of remote cross-domain control are effectively solved, and the personal safety of operators in a dangerous working environment is guaranteed.

Technical field: The invention belongs to the field of online educational platforms and networks, specifically systems and methods for supporting students through online mentoring and tutoring. Abstract: The invention relates to an online mentoring and tutoring network for supporting students in achieving their learning goals. The network integrates experts from various fields who offer individually tailored guidance. A central, cloud-based system connects mentors and students, enabling scalable and secure data processing and storage. An algorithm using machine learning and AI analyzes student and mentor profiles to create optimal pairings. The platform supports real-time interactions through WebRTC technology, enabling video calls, chat, and screen sharing. Mentors create customized learning plans based on data analysis of learning progress and the student's individual needs. The platform utilizes big data and analytics tools to monitor learning progress and generate reports.After each session, students and mentors can provide ratings and feedback to continuously improve the support. Inventive achievement: The invention integrates modern technologies such as cloud computing, machine learning, AI, WebRTC, big data and sentiment analysis to create a personalized, efficient and flexible learning environment, offering a significant technological advancement over existing platforms.

A system and method for bi-directional quality testing of web real-time communications (WebRTC) sessions. In an embodiment, the system and method comprise an operator network, cloud contact center, cloud contact center agent application, and a synthetic software agent comprised of agent automation software, injected API shim code, virtual audio devices, audio processing applications, and media servers, capable of performing automated and to end communication testing. In order to provide end to end testing, especially with respect to voice quality, the synthetic agent software may control and monitor the audio channels (both send and receive) of the browser communication session.

Disclosed in the present application are an on-board interconnection method and apparatus, and an electronic device and a storage medium. The method is applied to a terminal side, and comprises: in response to a first user interconnection operation, establishing a basic interconnection with a head unit side; in response to a screen-mirroring instruction for a target application, determining corresponding interconnection scenario data on the basis of an application interconnection manner, and an application interface of the target application; and on the basis of WebRTC, establishing a real-time transport protocol channel with the head unit side, and sending the interconnection scenario data to the head unit side by means of the real-time transport protocol channel, and displaying the interconnection scenario data. By means of the technical solutions of the embodiments of the present application, a vehicle factory only needs to develop an on-board interconnection scheme to meet market and user requirements, thereby achieving high compatibility; and developers can realize control across all layers from upper-layer application to underlying implementation, such that upon encountering a problem, the solving efficiency is high, and problem localization is clear, thereby reducing the access difficulty of an on-board interconnection scenario and improving the development efficiency.

The application relates to the technical field of data communication, and discloses a device end data transmission method and system based on WebRTC technology, which establishes a communication network through WebRTC and an adaptive network protocol corresponding to the communication network, so that when the current network state is in a congested state, the data transmission rate is determined according to the network bandwidth and the round-trip delay, and when the current network state is in an idle state, the data transmission rate is determined according to the congestion window size and the packet loss rate. The network can accommodate the data amount based on the network bandwidth and the round-trip delay, and the data transmission rate is determined, the network bandwidth is fully utilized, the data preemption capability is improved, the congestion window size and the packet loss rate control the data transmission rate, the data error rate during data transmission is reduced, the reliability of data transmission is improved, the flexibility during WebRTC data transmission is improved from two aspects, and the data transmission effect is greatly improved.

Abstract of the application: The invention relates to a central digital platform in the field of educational technology that links various subject areas and promotes project-based learning approaches. The technical problem is that traditional education systems often teach isolated disciplines, which makes the practical application of knowledge in real-world contexts difficult. The invention's solution comprises a web-based platform that integrates interactive tools such as simulations, virtual laboratories, and data analysis software to enable the practical development of projects. The platform supports student teamwork and collaboration through WebRTC-based communication tools and provides teachers with a dashboard to monitor student progress. The platform enables the creation of interactive presentations and reports and ensures data security through access controls and encryption technologies.The main uses of the invention are in schools and educational institutions to promote interdisciplinary learning and the development of teamwork, critical thinking and creative problem-solving skills.

The application discloses an audio and video online communication system and method based on combination of 5G message and WebRTC, and the audio and video online alarm system comprises a 5G user terminal, an interrogator server, a signaling server, a logic processing server, a media server, a call plan database and a video storage database; the 5G user terminal is connected with the signaling server, the media server and the logic processing server respectively; the interrogator server is connected with the signaling server, the media server and the logic processing server respectively; the signaling server is connected with the logic processing server and the media server respectively; the logic processing server is connected with the call plan database; and the media server is connected with the video database. The application does not need to download and install an application and can communicate through a video.

The application relates to the technical field of Internet of Vehicles, and discloses a car-machine multi-screen cooperative display and interaction method, which comprises the following steps: maintaining a global session and a member view through a centralized session arranger, realizing session access and dynamic member management of multi-screen devices, and determining multi-screen related parameters; constructing a full-order execution rule based on a logical clock, the multi-screen related parameters and an operation identifier, and assigning a global serial number to each multi-screen operation; obtaining a focus of an operation request, and determining feedback parameters through a soft lock and a priority preemption strategy; periodically hashing and checking based on a shadow state and a domain Merkle tree, positioning state differences according to the feedback parameters, and performing incremental repair through an incremental patch; and adopting WebRTC layered coding and a weighted bandwidth allocation strategy, preferentially guaranteeing control signaling and a main driving screen experience, and dynamically adapting in a bandwidth-limited scene. The application can solve the problems of multi-screen interaction inconsistency and insufficient real-time performance, thereby improving the multi-screen cooperative experience in a vehicle.

The present invention relates to a method and a system for remotely controlling a robot by using multiple channels, the system comprising: a robot which captures an image by using a camera, and which transmits image / voice data in real time through a single image / audio channel; a relay server for receiving the image / voice data from the robot through the single image / audio channel, and transmitting the received image / voice data through a plurality of image / audio channels separated from the single image / audio channel; and a plurality of clients matched one-to-one with each of the plurality of image / audio channels so as to receive each piece of the image / audio data through the plurality of image / audio channels, wherein the single image / audio channel and the plurality of image / audio channels use the WebRTC protocol.

The present invention relates to a robot control system based on a microservice architecture and a method thereof. The robot control system and method can operate in a control environment based on a microservice architecture (MSA). The robot control system includes an application programming interface (API) gateway and an administrator terminal, the API gateway connects an authentication server for authenticating a robot, a management server for managing the robot, and an operation server for operating the robot to the robot in a hypertext transfer protocol (HTTP) communication manner, and the administrator terminal communicates with the robot through a web real-time communication (WebRTC).

The application relates to the technical field of the Internet of Things, in particular to a remote control method and device, equipment and a computer storage medium, the method comprises the following steps: acquiring user information; judging whether the user information has control authority; if the user information has control authority, receiving a user instruction; acquiring a target device based on the user instruction and a device list; establishing a WebRTC data channel based on a stun server and the target device; and sending a remote control signal to the target device based on the user instruction and the WebRTC data channel. The application has the effect of reducing data transmission delay.

A method for establishing a communication session between a public switched telephone network (PSTN) endpoint and a Web real time communications (WebRTC) endpoint includes receiving, by a PSTN microservice (MS), a session initiation protocol (SIP) based signaling message from a PSTN endpoint and acquiring, by the PSTN MS, endpoint context attribute information from the SIP based signaling message. The method further includes generating, by the PSTN MS, a WebRTC based signaling message that includes an address identifier of an associate endpoint (AE) that is associated with the context attribute information and transmitting, by the PSTN MS, the WebRTC based signaling message to the AE to initiate a call session between the PSTN endpoint and the AE.

The invention discloses an effective streaming media audio and video recording system, and aims to improve audio and video processing efficiency and flexibility. According to the system, audio and video stream characteristics and system resources are dynamically analyzed through the streaming media control module, an encoding and decoding or direct connection processing mode is adopted for intelligent decision, and the problems that the performance burden is increased and the direct packaging function of the coded stream is limited due to traditional decoding and re-encoding are effectively solved. The coding and decoding unit can process complex audio and video data and support advanced functions such as audio mixing synthesis; and the direct connection unit ensures high concurrent processing and efficient file segmentation storage. The system supports a plurality of streaming media protocols such as RTP / RTSP / RTMP / WebRTC and the like, and adapts to various transmission requirements. Through intelligent analysis and flexible processing, the system not only improves the audio and video recording efficiency and quality, but also enhances the information display and file processing capabilities, and provides a comprehensive and efficient streaming media audio and video recording solution for the fields of remote cooperation, online education, security monitoring and the like.

The invention discloses a front-end-driven multi-channel video stream dynamic filtering and rendering method and system, belongs to the technical field of computer communication, and aims to solve the technical problem of how to solve member state synchronization abnormity and picture residue caused by network fluctuation and detection delay. Comprising the following steps: establishing a persistent bidirectional connection between a client and a converged communication message service background through a WebSocket protocol; many-to-many real-time communication is realized based on a WebRTC and MCU scheme; in combination with a WebSocket automatic reconnection mechanism and a WebRTC packet loss recovery mechanism, the reliability and stability of connection between the client and the converged communication message service background in a high-concurrency environment are enhanced; a Vue.js framework is used for front-end development, and a componentized and responsive user interface is achieved; front-end calculation is carried out, and signaling-service double-track data difference detection is triggered when members change; and the screen fusion service is requested again, and screen fusion video stream rendering is carried out.

The invention discloses an e-commerce platform purchase and sale two-way voice and video interaction method and system. The system comprises a purchaser terminal, a seller terminal and a server, and the server matches a preset audio and video acquisition and processing parameter template corresponding to a commodity category according to a commodity identifier sent by the purchaser terminal and issues the audio and video acquisition and processing parameter template to the seller terminal; the seller terminal configures an acquisition device based on the parameter template to acquire a target commodity, a point-to-point transmission channel is established through a WebRTC protocol to transmit audio and video streams, and the purchaser terminal synchronously plays the audio and video streams and displays associated service information. According to the invention, audio and video transmission optimization for different commodity types is realized through a commodity adaptive parameter template matching mechanism, the technical problems of weak network delay, unstable image quality, poor cross-terminal compatibility and the like in an e-commerce scene are effectively solved, and the definition and interaction experience of remote commodity display are improved.

The application relates to a screen projection method, device, system, computer equipment and storage medium. The method comprises the following steps: a playing end receives a sending end signaling from a sending end; the sending end signaling is a WebRTC signaling transmitted by the sending end through a UPnP protocol; the playing end generates a playing end signaling based on the sending end signaling, and sends the playing end signaling to the sending end through the UPnP protocol, so that the sending end and the playing end establish a WebRTC communication connection; and the playing end receives and displays real-time audio stream or mirror data from the sending end based on the WebRTC communication connection. The method can use the UPnP protocol as a signaling exchange channel of the WebRTC, realizes support and expansion of the WebRTC screen projection on the basis of the current mainstream DLNA screen projection function, does not increase hardware cost, does not reduce the popularity, and realizes mirror screen projection and real-time audio and video stream screen projection functions.

The embodiment of the application provides a kind of operating system instance management system, method and medium, in its system, identity authentication unit, for responding to the target instance debugging request from user end, according to the identity authentication signature of user end, identity authentication is carried out, and identity authentication result is obtained;First connection establishment unit is used to when identity authentication result is passed, based on Websocket protocol and SSH protocol session, establish the two-way communication connection between user end and target node;The second connection establishment unit is used to according to two-way communication connection, establish the WebRTC channel between user end and target operating system instance;Debugging unit is used to respond to the target instance debugging command for target operating system instance from user end, and the target instance debugging command is transmitted to target operating system instance by WebRTC channel, to carry out debugging to target operating system instance, to improve the management efficiency of operating system instance.

A video forwarding system based on ROS2 and WebRTC comprises a data acquisition and publishing module used for acquiring image data and publishing the image data to a corresponding ROS2 topic; the multi-process streaming pushing module is used for creating an independent streaming pushing sub-process for each path of video stream needing to be forwarded outwards and sending the video stream to the streaming media forwarding module; the streaming media forwarding module is used for receiving the independent video stream from the multi-process streaming pushing module and forwarding the video stream to a user terminal according to a user subscription relationship; the signaling and process management module is used for receiving an external instruction from a user side and dynamically creating or destroying the corresponding streaming pushing sub-process according to the instruction; and the user terminal module is used for sending an instruction to the signaling and process management module and subscribing and receiving the video stream from the streaming media forwarding module through the WebRTC technology. The method has the advantages of extremely low end-to-end delay, extremely low remote equipment resource occupation, strong expandability and flexibility and the like.

The invention discloses a Webrtc mute detection method and system based on FreeSwitch, and relates to the field of communication, the method comprises the following steps: code extraction: extracting a core code for mute detection from a Webrtc code; extracting code transformation and dynamic library compiling to obtain a dynamic library file of modvad.so; compiling and configuring Freeswitch: configuring Freeswitch compiling and model selection according to a system environment; module loading: when the Freesswitch is started, automatically loading the modvad module according to the configuration; audio stream receiving: Freeswitch is used as a communication platform to receive the audio stream from a call party client; and mute detection calling: when the Freeswitch receives the audio stream, calling a packaging interface in the modad module, and processing the audio frame by the packaging interface by using an algorithm in the vadcore.c file to judge whether the audio frame is a mute frame. The mute segment can be identified more accurately, and the detection precision is improved.