176 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 relates to the technical field of electric power operation scene modeling, and discloses a WebRTC technology-based audio and video interaction method and system in a hot-line operation scene, and the method comprises the steps: collecting real-time data of an operator and an operation environment, collecting environment point cloud data through a laser radar module, and generating a three-dimensional space model of the operation scene; superposing key region labeling information of the three-dimensional space model in the WebRTC video stream to realize visual enhancement of the working environment; the safety distance between the operator and the electrified body and the grounding body is calculated in real time, and when the safety distance is lower than a preset threshold value, active early warning is triggered through audio and video streams. Compared with the prior art, efficient dynamic modeling, low-delay video interaction and real-time safety monitoring are integrated, and a reliable guarantee is provided for intellectualization and safety of hot-line work.

A method includes receiving, by a first device, a request from a second device to participate in a session, the request being a message compliant with a WebRTC framework and including an identifier of a process hosted by the second device; verifying, by the first device, a type of the process hosted by the second device based on the identifier; initiating, by the first device, one or more actions on the first device in response to verification of the type of the process, the actions being other than those to communicate data between the first device and second device; and establishing, by the first device, the session with the second device after initialization of the actions on the first device.

The invention relates to the technical field of video acquisition coding and transmission, and provides a WebRTC-based method for reducing transmission bandwidth and improving transmission stability, which comprises the following steps: S1, a video acquisition module acquires a video frame from a camera and sends the video frame to a video coding module; s2, performing AI scene detection on the video frame to obtain a picture scene of the video frame, and performing region-of-interest ROI detection on the video frame to obtain a region-of-interest ROI range of the video frame; s3, optimizing the original funnel algorithm, setting different code rates for different video frames according to different video frame scenes, and setting different code rates for different areas of the video frames according to different ROI (Region of Interest) ranges of the video frames; s4, encoding the video frame, and sending the encoded video frame to a receiving end; s5, the receiving end decodes the received compressed video data into original video frames by using a video decoder; and S6, transmitting the decoded video frame to a rendering module for video rendering and display. The transmission bandwidth is reduced, and the transmission stability is improved.

The invention discloses a WebRTC audio and video stream parameter dynamic adjustment method and device and computer equipment. The method comprises the following steps: acquiring a WebRTC audio and video stream and a current network state; performing comprehensive score calculation on the WebRTC audio and video streams to obtain a calculation result; when the calculation result is smaller than a set threshold value, inputting the current network state into an adaptive model to obtain an adjustment strategy; dynamically adjusting the size of a buffer area of audio and video data according to the adjustment strategy; according to WebRTC audio and video streams monitored in real time and user preferences, the GOP size, the QP value and code rate control are adjusted. By implementing the method provided by the invention, the parameters can be dynamically adjusted to adapt to a dynamically changing network environment, and the transmission efficiency and quality are improved.

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 distributed GPU scheduling, data encryption management and dynamic charging system oriented to privatized AI deployment, and belongs to the technical field of distributed computing and artificial intelligence. The system comprises a decentralized scheduling module, which distributes tasks to distributed GPU nodes through a dynamic weight scoring model (weight factors comprise GPU utilization rate, network delay and task priority); the central control load balancing unit is used for monitoring node states in real time and dynamically adjusting a scheduling strategy; the data encryption module is used for establishing a point-to-point encryption channel between a user terminal and a GPU node based on a WebRTC technology, and data is encrypted by a national cryptographic SM4 algorithm and is decrypted only at a target node; and the dynamic billing unit is used for realizing 0.01-element micro billing according to the GPU computing power consumption and automatically distributing earnings through an intelligent contract. Experiments show that the task completion time of the system in a 100-node cluster is shortened by 30%, the resource utilization rate is improved by 45%, the data leakage risk is reduced by 98%, and the system is suitable for enterprise sensitive data protection and personal user elastic computing power requirements.

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 vehicle detection path planning and interaction method and system based on the universe and application of the vehicle detection path planning and interaction method and system in Mars rover detection. The method comprises the following steps: constructing collaborative interaction: realizing voice, video and data transmission based on a WebRTC protocol, classifying gesture actions of a user by adopting a support vector machine algorithm, and mapping a classification result into a vehicle movement, data acquisition or path modification operation instruction in a meta-universe environment; a machine learning method is used for extracting geological data features, a terrain detection target is combined, a vehicle terrain detection path plan is generated and projected to the meta-universe environment in real time, and a user shares and visualizes data in the meta-universe environment through a collaborative interaction module and checks and modifies the path plan; and after the user confirms that the path planning is correct, the final optimized path planning and related operation instructions are fed back to the vehicle to guide the vehicle to execute the terrain detection task, so that the efficiency and scientificity of terrain detection are improved.

The invention discloses a Grafana-based video monitoring method, system and device, and belongs to the technical field of industrial Internet of Things, and the system comprises an industrial personal computer, go2rtc software and Grafana software. The industrial personal computer is used for providing a running environment for the go2rtc software and the Grafana software; the go2rtc software converts a video stream into a WebRTC video stream through protocols such as RTSP, MJPEG, JPEG and RTMP; and the Grafana software is responsible for analyzing the WebRTC video stream and publishing the WebRTC video stream to a Web server, and a user remotely accesses a visual interface through a Web browser. The problems that video monitoring of an existing monitoring system is difficult to integrate with other service systems, video picture delay is high, cross-platform access is limited, maintenance cost is high, and deployment platforms are limited are solved, and digital transformation of enterprises is accelerated.

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 provides a video tag generation method based on geographic information fusion. The method comprises the following steps: S1, building a GIS map three-dimensional scene; s2, controlling and accessing the network camera; s3, calling the webtc-streamer. Js to play and display the video in real time; s4, calibrating a GIS map and a camera picture, and guiding the camera to rotate; s5, any point O is selected in the map, the camera is guided to rotate to obtain a horizontal direction value, a pitching value and a focal length value, and an angle APB is obtained; s6, acquiring a camera field angle of the three-dimensional map; s7, acquiring coordinates of a camera visual angle point when the camera looks towards the O point; s8, adjusting the view field angle of the camera in the map to a view angle point position; s9, transparentizing the background of the map, and reserving the original entity label in the GIS map; and S10, superposing the transparent GIS map scene to a video picture of the camera. The problems of high cost and large limitation of a video tag generation method are solved.

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.