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

An apparatus may be configured to: receive, from a first user equipment, at least one real-time transport protocol header extension configured to handle at least one protocol data unit, wherein the at least one real-time transport protocol header extension comprises, at least, one or more application controlled bits; determine whether the one or more application controlled bits are valid; and set values for one or more network controlled bits of the at least one real-time transport protocol header extension based, at least partially, on a determination of whether the one or more application controlled bits are valid, wherein the one or more application controlled bits comprise bits that do not overlap with the one or more network controlled bits.

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.

The invention relates to a robot control system and method, a storage medium and electronic equipment. The system runs in a heterogeneous system, the heterogeneous system comprises a non-real-time system and one or more hard real-time systems, a first control system runs on the non-real-time system, and a real-time control framework runs on the hard real-time system; the real-time control framework comprises a control communication module, a motion algorithm module and a distributed communication module; the hard real-time system sends collected feedback data of the robot to a first control system on a non-real-time system by calling a sending function in a real-time transmission protocol of a distributed communication module; a control instruction of the first control system is received by calling a receiving function in a real-time transmission protocol of the distributed communication module; and the control instruction is sent to the robot through the control communication module. The technical problem that a soft real-time system is poor in real-time performance and unstable when controlling a robot is solved.

The invention provides a low-delay real-time voice transcription method and system, and the method comprises the following steps: A, carrying out the coding compression of original audio data collected by a client, transmitting the obtained audio coding data to a server through a real-time transmission protocol, introducing an error correction and packet loss retransmission mechanism, and improving the transmission reliability; b, detecting voice in the audio data by a voice activity detection algorithm at the server side, dividing the audio data judged as the voice into a plurality of independent parts, and adding a time sequence identifier for each part; and C, performing real-time transcription on the segmented audio data according to a time sequence to generate corresponding text content.

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.

A method for handling network access during by a User Equipment (UE) in a wireless communication system, may include: identifying an initiation of a handover from a 3GPP network to a non-3GPP network; determining whether the handover is initiated during an ongoing call at the UE, or an always-on packet data unit (PDU) session at the UE; suspending at least one suspended real-time transport protocol (RTP) packet on the 3GPP network to prevent transmission of at least one service request to the 3GPP network during the handover based on the handover being initiated during the ongoing call at the UE, and configuring a status of the always-on PDU session as inactive for the 3GPP network to prevent the transmission of the at least one service request to the 3GPP network during the handover based on the handover being initiated during the always-on PDU session at the UE.

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 belongs to the technical field of data transmission, and provides an internet-of-things data transmission protocol adaptation method and system based on big data, and the method comprises the steps: obtaining data transmission protocol adaptation features according to data collection features, data essential feature coefficients and protocol parameters of internet-of-things equipment; and according to the data transmission protocol adaptation characteristics and the corresponding transmission protocol adaptation method, constructing an Internet of Things data transmission adaptation model, using the model to carry out adaptive transmission protocol adaptation on subsequent collected data and output a real-time transmission protocol adaptation method, and according to the real-time transmission protocol adaptation method, controlling a protocol converter device. And thus, the data transmission of the internet of things is subjected to adaptive management. The conversion efficiency of the data adaptation transmission protocol is enhanced through multi-dimensional feature processing of the data of the internet of things and the improved activation function big data model, the transmission speed and quality of the data of the internet of things are improved, and the manual adaptation loss of the data transmission protocol is reduced.

The invention discloses an ASIC chip for high-performance AOIP network audio stream transmission, and the interior of the chip comprises a function simulation circuit subsystem, a sensor controller, a clock, and a reset control module. The high-speed peripheral interface subsystem is arranged on the first bus, the hardware acceleration subsystem is arranged on the second bus, the low-speed peripheral interface subsystem is arranged on the third bus, the second bus is connected with the first bus and the third bus, and the function simulation circuit subsystem is connected with the third bus through the sensor controller; the hardware acceleration subsystem comprises a plurality of processor cores, two gigabit or 100M bit medium independent interfaces and a real-time transport protocol RTP transceiving engine; the two gigabit or 100M bit medium independent interfaces are compatible with the simplified medium independent interface, one of the two gigabit or 100M bit medium independent interfaces is used as a main network interface, and the other one is used as a standby network interface. The reliability problem of AOIP network audio stream transmission is solved.

Example methods, devices, and computer-readable media are described. An example method includes sending or receiving, by a first computing device to or from a second computing device, information indicative of the use of a first meaning of a plurality of meanings of a Real-Time Transport Protocol (RTP) header extension syntax of an RTP header extension of an RTP packet. The method includes processing, by the first computing device, the RTP packet comprising extended reality (XR) application data based on the first meaning.

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.

A method and apparatus for signaling multiple audio mixing gains using a single Real-Time Transport Protocol (RTP) header extension in a teleconference is provided. The method includes: receiving an input audio stream from a 360-degree video / audio stream, the input audio stream including multiple mixing gains from the input audio stream and an overlay audio stream; declaring a single RTP header extension for all mixing gains (including one or more of an element identifier, a length of the extension element, and a magnitude of the mixing gain); and signaling the mixing gains using the single RTP header extension. The single RTP header extension can use a one-byte header extension format or a two-byte header extension format, and the one-byte header extension format or the two-byte header extension format is declared using the Session Description Protocol (SDP).

A device configured to transmit data may generate one or more source protocol data units (PDUs) in a PDU set, and generate one or more corresponding repair PDUs in the PDU set, wherein the source PDUs and the repair PDUs share the same PDU set sequence number (PSSN), and wherein the corresponding repair PDUs are used for forward error correction (FEC), and transmit the PDU set. A real-time transport protocol header extension (HE) may include information that indicates whether a PDU in the PDU set is a source PDU or a repair PDU.

A method and device for signaling multiple audio mixing gains in a teleconference using a single Real-time Transport Protocol (RTP) header extension is provided. The method includes receiving an input audio stream from a 360-degree video / audio stream that includes multiple mixing gains from the input audio stream and overlay audio stream, declaring a single RTP header extension (including one or more of an element identifier, a length of an extension element, and a magnitude of the mixing gains) for all the mixing gains, and signaling the mixing gains using the single RTP header extension. The single RTP header extension may use a one-byte or two-byte header extension format which is declared using a Session Description Protocol (SDP).

The invention particularly relates to a GB28181 streaming media optimization method based on a QUIC protocol. According to the GB28181 streaming media optimization method based on the QUIC protocol, a QUIC protocol stack is introduced into a GB28181 media transmission layer, a traditional TCP / UDP protocol of the media transmission layer is replaced, and interaction with a GB28181 signaling layer is achieved; a real-time transport protocol RTP data packet is packaged into a QUIC load, and flow pushing and flow receiving are carried out through a QUIC channel; qUIC transmission parameters are dynamically adjusted based on the network state, and the media stream transmission performance is optimized. According to the GB28181 streaming media optimization method based on the QUIC protocol, the problems of high delay, insufficient packet loss resistance and the like of traditional GB28181 in media transmission are solved, deep fusion of a QUIC protocol stack and GB28181 signaling interaction is realized, the efficiency, reliability and safety of media stream transmission are remarkably improved, and the GB28181 streaming media optimization method based on the QUIC protocol is suitable for video monitoring scenes in high-concurrency and weak network environments.

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.

A user equipment (UE) is configured to: determine a real-time transport protocol (RTP) packet size; determining to aggregate one or more voice packets into the RTP packet based on the RTP packet size; determining the number of the one or more voice packets; and transmitting the RTP packet including the number of the one or more voice packets.

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 present application relates to the field of data transmission technologies, and particularly relates to a method and apparatus for calculating delay. Applied to a first terminal device, a Real-time Transport Protocol (RTP) is run between the first terminal device and a second terminal device. The method includes: determining a current reference delay, where the current reference delay is a loopback delay calculated based on an RTP Control Protocol message and between the first terminal device and the second terminal device; calculating a one-way transmission delay for transmitting the RTP message based on the reception timestamp of the RTP message sent by the second terminal device received and the transmission timestamp carried in the RTP message indicating the second terminal device sending the RTP message; determining a one-way transmission delay deviation between two RTP messages based on the one-way transmission delay of the RTP message and the one-way transmission delay of the RTP message calculated in the previous time; and adjusting the current reference delay between the first terminal device and the second terminal device based on the current reference delay and the one-way transmission delay deviation.

An embodiment of the present application discloses a call processing method, device and system, which includes: a media server receives a call request, negotiates video media resources with a first terminal, and based on the negotiation result of the video media resource negotiation with the first terminal, sends a real-time transport protocol RTP stream to the first terminal, the RTP stream including a first RTP message of the first video media and a second RTP message of the second video media, the first RTP message including a synchronization source SSRC identifier of the RTP stream and an SSRC identifier of the first video media, and the second RTP message including an SSRC identifier of the RTP stream and an SSRC identifier of the second video media. Without splicing the first video media and the second video media in advance, the media server can instruct the first terminal to play the first video media and the second video media on the screen simultaneously through the RTP stream, which is conducive to saving computing resources required for video splicing.