43 results about "Real-time Transport Protocol" patented technology

Low- to ultralow-latency content delivery via real-time transport protocol (RTP) is provided. In an example, transport packets, which carry a packetized elementary stream (PES), are selectively marked based on frame size. More particularly, if a number or size of packets needed to transport a picture or PES packet exceeds a threshold, they are marked for preferential processing, such as low latency, low loss, and scalable throughput (L4S) processing. If the number or size of packets is below the threshold, they are marked for default or non-preferential processing. The marking may be applied to entire pictures, tiles, and / or slices. Related apparatuses, devices, techniques, and articles are also described.

Methods, systems, and devices for wireless communications are described. Techniques described provide for real-time transport protocol (RTP) header compression for transport over a non-terrestrial network (NTN). In some examples, a first network entity may receive, from a second network entity supported by the NTN, a first RTP packet comprising an audio payload associated with the second network entity, where the first RTP packet excludes one or more RTP header fields associated with an RTP packet generation protocol. The network entity may generate at least one RTP header field of the RTP header fields excluded from the first RTP packet. The first network entity may output, to a third network entity or to an upper layer RTP stack of the first network entity, a second RTP packet including the audio payload associated with the second network entity and the at least one RTP header field generated by the first network entity.

Various aspects of the present disclosure relate to user interaction data transportation using real-time transport protocol header extension. One or more physical device interactions by a user are sampled to obtain interaction data inputs, such as for a user playing cloud gaming or using extended reality on their user equipment. These interaction data inputs are generated frequently, such as approximately 250 to 1000 times per second. One or more interaction short messages (e.g., less than 100 bytes), which are time-sensitive, are generated from the user interaction data and included in one or more transport header extensions of a real-time transport protocol data unit. The real-time transport protocol data unit is transmitted to a remote device, such as a network entity.

The invention provides a voice communication method, device and system, and electronic equipment, and relates to the technical field of communication. The method comprises: a first terminal device receiving a plurality of real-time transport protocol (RTP) packets from a second terminal device, the plurality of RTP packets being RTP packets of a voice call between the first terminal device and the second terminal device, each of the plurality of RTP packets comprising detection information, the detection information comprising at least one of load type information and verification information; and when it is determined that the number of abnormal RTP packets in the plurality of RTP packets exceeds a preset threshold according to the detection information in the plurality of RTP packets, sending a re-negotiation request of a load code corresponding to the voice call.

A method for a sending apparatus comprises generating protocol data units (PDUs) for one or more media streams to be transmitted to a receiver in a real-time transport protocol (RTP) session; identifying PDU types that are not to be marked with PDU set information; defining an importance value for the identified PDU types; generating a mapping between the importance values and respective identified PDU types; signaling the generated mapping to the network to be used by the network to determine an importance value to be assigned to a PDU not being marked with the PDU set information; and transmitting all the generated PDUs to the network. Embodiments concern also a method for receiving, and apparatuses for implementing the methods.

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 invention relates to the technical field of image communication, and discloses a low-delay transmission method and system for automatic image quality enhancement in a weak network environment, and the method comprises the steps: receiving and analyzing a real-time transport protocol (RTP) data packet from a transmitting end, and extracting video coding data; monitoring quality parameters of a network transmission channel between the sending end and the receiving end in real time, wherein the quality parameters comprise a packet loss rate, network jitter and transmission delay; adaptively selecting an image quality enhancement algorithm according to the network transmission quality parameter, and carrying out decoding and image quality enhancement processing on the video coding data; and performing hardware coding on the video frame subjected to image quality enhancement processing by using a graphics processor, and outputting the video frame to a rendering display module of a receiving end. Through automatic detection of network conditions and automatic analysis of received video coding data, image quality enhancement coding processing is carried out on video images through noise reduction, super division, frame insertion, sharpening convolution and other algorithms, and then output display is carried out, so that low-delay transmission of weak network automatic image quality enhancement is realized.

The invention discloses a data rendering method, a media processing network element, terminal equipment and a storage medium. The data rendering method is applied to a media processing network element, and the method comprises: establishing a transmission channel between the media processing network element and a terminal device, and allocating a corresponding identifier to the transmission channel, the transmission channel at least comprising a first real-time transport protocol (RTP) channel, and the identifier corresponding to the first RTP channel being used for indicating the first RTP channel to transmit data to be rendered; obtaining to-be-rendered data sent by the terminal device through the first RTP channel; and rendering the to-be-rendered data to obtain rendered data.

Facilitating elastic distributed computing for resource intensive tasks including split rendering in advanced networks is provided herein. Operations of a system include receiving information indicative of a request for split rendering for an application executing at a user equipment. The information is received via a transport protocol header extension. The operations can also include, based on the information indicative of the request, determining that an available computational resource capacity at the user equipment is insufficient to process the application executing at the user equipment. Further, the operations can include, based on the determining, enabling a first rendering of a first portion of the application at a first device and a second rendering of a second portion of the application at a second device different from the first device.

The invention discloses a data transmission method and device, communication equipment and a readable storage medium, and belongs to the technical field of communication, and the data transmission method comprises the steps that a first node sends a first data packet to a second node; wherein the first data packet comprises static field information of a header, and the static field information of the header comprises at least one of the following items: static field information of a real-time transport protocol (RTP) header; static field information of a user datagram protocol UDP header; the invention relates to static field information of an Internet Protocol (IP) header.

The application discloses a video bit stream packaging method and device, electronic equipment and program product, and belongs to the technical field of audio and video. The video bit stream packaging method comprises the following steps: cutting the coded video bit stream to obtain a plurality of meta-codes; and packaging the meta-codes into real-time transport protocol (RTP) data packets according to the load type of each meta-code. Thus, the RTP packaging can be performed according to the load type of the meta-codes, so that the RTP data packets can be reasonably obtained, and the packet delay can be reduced.

The application provides an 8K high code rate panoramic live broadcast grading distribution method for a sports event private network, and belongs to the technical field of live broadcast distribution.The original 8K panoramic video is compressed and encoded by a high-efficiency video coding standard to generate a high code rate video stream in a live source node, and the high code rate video stream is pushed to a first forwarding node through a real-time transmission protocol.The maximum transmission unit of the private network full link is uniformly configured as 9000 bytes in the first forwarding node, and a data plane development kit user state protocol stack is enabled.A second distribution node concurrently pushes the high code rate video stream to terminal equipment through a lock-free ring buffer.A fault detection module combines active heartbeat detection with passive response time monitoring to monitor the node state in real time, and migrates the affected terminal to a backup node within a second, thereby solving the technical problems of high transmission delay and poor concurrent stability of the high code rate video stream during the grading distribution of the 8K high code rate panoramic video of the sports event private network.

The application discloses a satellite communication-based live broadcast interaction method and device, equipment and a storage medium, relates to the field of video transmission, and is applied to a target server and comprises the following steps: acquiring a target live broadcast video stream sent by a first type of client corresponding to a target live broadcast room through a first type of port number, and acquiring a target multicast stream corresponding to the target live broadcast video stream based on a first target transmission protocol; the first target transmission protocol is a real-time transmission protocol based on a user datagram protocol; acquiring a target interaction message stream sent by a second type of client through a second type of port number, and processing the target interaction message stream to acquire a corresponding target message stream; the target message stream is a message stream transmitted based on a second target transmission protocol; and the target message stream is displayed on each target client corresponding to the target live broadcast room based on the message types of the interaction messages in the target message stream. The application improves the live broadcast watching experience of users.

The invention discloses a transmission method and device of a voice real-time transmission protocol RTP data packet. The method comprises the following steps: receiving a target voice RTP data packet; determining an abnormal transmission condition of the target voice RTP data packet according to the transmission state information of the target voice RTP data packet; wherein the abnormal transmission condition comprises jumping or disorder; according to the abnormal transmission condition of the target voice RTP data packet, performing reordering processing on the target voice RTP data packet through a reordering queue in a running state or a reordering queue in an idle state; and transmitting the target voice RTP data packet after the reordering processing.

The invention discloses a live broadcast interaction method and device based on satellite communication, equipment and a storage medium, relates to the field of video transmission, is applied to a target server, and comprises the following steps: obtaining a target live broadcast video stream sent by a first class client corresponding to a target live broadcast room through a first class port number, obtaining a target multicast stream corresponding to the target live video stream based on the first target transmission protocol; the first target transmission protocol is a real-time transmission protocol based on a user datagram protocol; obtaining a target interactive message stream sent by the second type of client through the second type of port number, and processing the target interactive message stream to obtain a corresponding target message stream; the target message stream is a message stream transmitted based on a second target transmission protocol; and displaying the target message stream in each target client corresponding to the target live broadcast room based on the message type corresponding to each interactive message in the target message stream. The live broadcast watching experience of the user is improved.

This document provides a Real-Time Transport Protocol (RTP) sender for wireless communication, the RTP sender comprising: at least one memory; and at least one processor coupled to and arranged such that the RTP sender: receives a Protocol Data Unit (PDU) set size threshold, the PDU set size threshold determining the PDU set size flow characteristics of a service data stream; applies the PDU set size threshold to the PDU sets of the service data stream, the application including identifying a PDU set size change between a first PDU set and a second PDU set in the PDU sets of the service data stream, wherein the size of the second PDU set satisfies the PDU set size threshold; and transmits the first PDU set, the second PDU set, and an indication of the dynamic PDU set size change.

Embodiments of the present application disclose a data processing method and device, and a computer readable storage medium. A unified platform sends corresponding voice prompt information to a mobile terminal in response to a data processing request sent by the mobile terminal based on a session initiation protocol (SIP). The data processing request carries a target IMS account and an identification code of the mobile terminal. The target media terminal identification code sent by the mobile terminal based on the voice prompt information is received. If the identification code of the target mobile terminal obtained based on the target IMS account and the target media terminal identification code is equal to the identification code of the mobile terminal, a data viewing request is sent to the target media terminal according to the target IMS account and the target media terminal identification code. Media data collected by the target media terminal in response to the data viewing request is received, and the media data is sent to the mobile terminal based on a real-time transport protocol (RTP). The media data can be viewed through the data processing request, which has strong universality and improves security.

The invention discloses an audit data processing method, system and device based on multi-terminal transmission and a medium, and the method comprises the steps: obtaining metadata features of audit data uploaded by a data terminal, and matching a corresponding data warehouse for each piece of audit data according to the metadata features; acquiring a real-time transmission protocol corresponding to each data warehouse, and sending the audit data to the corresponding data warehouse according to the real-time transmission protocol to obtain a plurality of audit databases; selecting a target auditing database from a plurality of auditing databases according to an auditing request uploaded by an auditing terminal, and converting each piece of auditing data in the target auditing database into standard auditing data; obtaining a data quality evaluation result corresponding to each standard audit data through a data quality evaluation model; and determining a plurality of target auditing data from the plurality of standard auditing data according to the data quality evaluation result, and sending the target auditing data to the auditing terminal so as to improve the auditing efficiency.

The application provides a call method, device, equipment, medium and product. A call request is received from a calling terminal, and the call request is used for calling a called terminal. If an address field is carried in the call request, and the address in the address field does not conform to a preset address rule, a session initiation protocol (SIP) response information is sent to the calling terminal. The preset address rule is used to indicate whether a real public internet protocol (IP) address of the calling terminal exists in the address field. If the calling terminal and the called terminal are not successfully connected, and the first real-time transport protocol (RTP) silence data sent by the calling terminal according to the SIP response information is received within a preset time, the first public IP address of the calling terminal is determined according to the first RTP silence data. The real public IP address of the calling terminal can be obtained without depending on the network environment of the calling terminal, and the security of the call is improved.

Example apparatus and techniques for delay measurement are described. An example method includes determining whether a source port of a real-time transport protocol (RTP) packet of a multimedia application session is a same source port as a source port of a real-time transport control protocol (RTCP) packet of the multimedia application session. The method includes determining whether a destination port of the RTP packet is the same destination port as a destination port of the RTCP packet. The method includes determining whether to use the RTP header extension to perform the delay determination or to use the RTCP packet based on whether the source port of the RTP packet is the same source port as the source port of the RTCP packet and whether the destination port of the RTP packet is the same destination port as the destination port of the RTCP packet.

The invention provides a message processing method, equipment and device and a storage medium, and the method comprises the steps: receiving a second voice data packet after continuously receiving a plurality of first voice data packets; wherein the first voice data packet is a data packet in which a first real-time transport protocol (RTP) message is subjected to cyclic redundancy check (CRC) failure, the second voice data packet is a resynchronization data packet, and the resynchronization data packet is used for synchronizing compression information between the compression end and the decompression end; decompressing the second voice data packet to obtain a first message and an RTP serial number and a timestamp of the first message; based on the RTP serial number and the timestamp of the first message, determining the message content of the RTP header of each first voice data packet; and based on the message content of the RTP header of each first voice data packet, determining a second message corresponding to the first voice data packet, the first message and the second message being used for voice synthesis. Through the method, the packet loss rate of the voice data packet can be reduced.

The present disclosure provides a system (106) and method for preventing offline using call forwarding. A system (106) monitors one or more real-time transport protocol (RTP) data packets transmitted between a first user equipment (UE) (102-1) and a second UE (102-2). The system (106) determines whether the first UE (102-1) is unable to transmit one or more RTP data packets based on inoperation of a primary subscriber identity module (SIM) associated with the first UE (102-1). The system (106) simultaneously facilitates transmission of one or more RTP data packets over a secondary SIM associated with the first UE (102-1) in response to determining that the primary SIM is inoperative. The system (106) enables communication between the first UE (102-1) and the second UE (102-2) based on transmitting the one or more RTP data packets through the secondary SIM.

Embodiments may relate to a user equipment (UE) that is configured to determine a real-time transport protocol (RTP) media flow that includes visual data related to a plurality of images concurrently taken of a location and a supplemental information enhancement (SEI) message that is to be used to display at least a portion of the visual data. The UE is further configured to visually display, based on the visual data and the SEI message, the portion of the visual data to a user of the user device. Other embodiments may be described or claimed.

The application provides a voice single-pass detection method, device, equipment, medium and product, and belongs to the technical field of artificial intelligence. The method comprises the following steps: acquiring voice session data packets; splitting the voice session data packets according to data flow directions to obtain multiple data flows; determining the missing conditions of real-time transport protocol (RTP) data packets and the missing conditions of real-time transport control protocol (RTCP) data packets of each data flow; determining the data type of the voice session data packets based on the missing conditions of the RTP data packets and the missing conditions of the RTCP data packets of each data flow; and performing single-pass detection on the voice session data packets based on the data type to obtain a single-pass detection result. The voice single-pass detection method provided by the application can reduce the influence of the missing of part of field data on the detection of the entire voice session data packet, guarantee the voice single-pass detection coverage, and thus improve the accuracy of voice single-pass detection.

Methods, systems, and devices for wireless communications are described. Techniques described herein provide for real-time transport protocol (RTP) header compression for transport over a non-terrestrial network (NTN). In some examples, first network entity may receive, from a second network entity supported by the NTN, a first RTP packet comprising an audio payload associated with the second network entity, where the first RTP packet excludes one or more RTP header fields associated with an RTP packet generation protocol. The network entity may generate at least one RTP header field of the RTP header fields excluded from the first RTP packet. The first network entity may output, to a third network entity or to an upper layer RTP stack of the first network entity, a second RTP packet including the audio payload associated with the second network entity and the at least one RTP header field generated by the first network entity.

The application discloses an ASIC chip for high-performance AOIP network audio stream transmission, which internally comprises a functional analog circuit subsystem, a sensor controller, a clock and reset control module, a high-speed peripheral interface subsystem arranged on a first bus, a hardware acceleration subsystem arranged on a second bus, and a low-speed peripheral interface subsystem arranged on a third bus, the second bus being connected with the first bus and the third bus respectively, and the functional analog circuit subsystem being connected with the third bus through the sensor controller; the hardware acceleration subsystem comprises a plurality of processor cores, two gigabit or megabit media independent interfaces and a real-time transmission protocol (RTP) transceiving engine; the two gigabit or megabit media independent interfaces are compatible with simplified media independent interfaces, one of the two gigabit or megabit media independent interfaces is used as a main network interface, and the other is used as a backup network interface. The reliability of AOIP network audio stream transmission is solved.

The embodiment of the invention discloses a voice communication method and related equipment, which are used for protecting voiceprint privacy of a caller and reducing network transmission load while realizing cross-language real-time communication. The method provided by the embodiment of the invention comprises the following steps: determining a current call with a second communication terminal, and obtaining a first voice signal in the current call; converting the first voice signal into corresponding first text data; packaging the first text data into a first real-time transmission protocol data packet; wherein the load of the first real-time transmission protocol data packet comprises first text data; and sending the first real-time transmission protocol data packet to the second communication terminal, so that the second communication terminal extracts the first text data from the load, converts the first text data into a first playing signal and plays the first playing signal.

The present application relates to the technical field of communications, and discloses a data transmission method and apparatus, a communication device, and a readable storage medium. The data transmission method in embodiments of the present application comprises: a first node sends a first data packet to a second node, wherein the first data packet comprises static field information of a header, and the static field information of the header comprises at least one of the following: static field information of a real-time transport protocol (RTP) header; static field information of a user datagram protocol (UDP) header; and static field information of an Internet protocol (IP) header.