367results about How to "Reduce packet loss" patented technology

Telemedicine systems and methods are described. In a telemedicine system operable to communicate with a remote operations center, communications can be transmitted/received using a transceiver having an antenna. The antenna can include first and second di-pole antenna elements, the first di-pole antenna element being vertically polarized and the second di-pole antenna element being horizontally polarized. A controller of the system can establish, using the transceiver, a telemedicine session with the operations center using a Transport Morphing Protocol (TMP), the TMP being an acknowledgement-based user datagram protocol. The controller can also mask one or more transient network degradations to increase resiliency of the telemedicine session. The telemedicine system can include a 2D and 3D carotid Doppler and transcranial Doppler and/or other diagnostic devices, and provides for real-time connectivity and communication between medical personnel in an emergency vehicle and a receiving hospital for immediate diagnosis and treatment to a patient in need.

A router, if judging that a routing function is impossible to execute, sends out a connection instruction message including the address of a terminal on a local network terminating the relaying flow, to another router to solicit switching. The router having received the connection instruction message starts a process for connecting with an external network while executing a process for link connection with a terminal, and then sends out a connection completion notice message when the connection with both the external network and the terminal is established. This process enables the router to switch a path rapidly.

Embodiments of the present invention provide an automatic protection switching method, device and system. The method includes: determining a part of services as to-be-switched service(s) according to change of bandwidth when monitoring that the bandwidth of the first transmission path changes; and switching the to-be-switched service(s) between links on the second transmission path and the first transmission path. Another method includes: receiving a partial switching message from a peer network edge node through the first transmission path or the second transmission path; and determining the to-be-switched service(s) according to indication information about the to-be-switched service(s) or bandwidth change information in the partial switching message, and switching the to-be-switched service(s) between links on the first transmission path and the second transmission path.

The invention discloses a method and system for ensuring streaming media service quality. The method comprises the steps of monitoring the performance of a current network and triggering switch when monitoring that the performance of the current network is insufficient to meet the demands on the service quality of steaming media service; determining a target network, notifying a mobile node aboutthe relevant information for switching the target network, and sending service quality parameter information of the steaming media service of the mobile node to a target service quality proxy functionmodule in a target network node from an original service quality proxy function module in the original access network node; using the target service quality proxy function module in the target network node for establishing service quality mapping; and adopting different switch strategies according to different types of the mobile nodes: adopting a synchronous algorithm for carrying out data transmission when the mobile node is dual interfaces and adopting the switch strategy based on cache when the mobile node is a single interface. The method and the system can avoid the problem of low datatransmission efficiency due to the use of dual network interface cards in the prior art, lead a plurality of access modes to work cooperatively and realize seamless switch.

An embodiment of the invention discloses a method for regulating the transmitting speed of data package, a system and a device thereof. The method for regulating the transmitting speed of the data package comprises the following steps: calculating the shaking statistic of the data in the transmission link according to the received backward reporting BR package; determining whether the shaking statistic exceeds the prearranged first speed reduction threshold; and if the shaking statistic exceeds the first speed reduction threshold, executing speed reduction procession to the transmission speed of the data package. Through the embodiment of the invention, the base station controller self-adaptively regulates the data package transmission speed of the base station controller according to the shaking statistic calculated by the base station controller thereby effectively controlling the discharging flow rate of the base station controller, increasing the stability of the system under the state of heavy load, reducing the package loss and time delay of the transmission link and increasing the usage factor of the band width.

The invention relates to a special wireless chain redistribute method with refreshable activate time, used in wideband code-division complex system, wherein said method comprises that: BSC and base station process redistribution of synchronous wireless chain at Iub interface; BSC and terminal process redistribution of wireless interface; BSC and RRC layer confirms the successful transmission of redistribution message, wait T2 time, and convert to new distribution, send trigger message to the base station; base station receives trigger message, converts to new distribution, and sends descending DPCCH signal to terminal via new physical channel; when the terminal redistributes successfully, physical layer receives DPCCH signal at new channel, synchronizes descending data and informs the RRC layer, terminal converts to new wireless chain and sends wireless interface redistribution complete message to BSC. The inventive method can shorten time delay and reduce loss.

The invention discloses a method for optimizing the video quality of service (QoS) of a video wireless transmission monitoring system. In the wireless communication process of a video signal acquisition processing transmission terminal and a video signal receiving play server, the video signal receiving play server counts video stream data packets sent by the video signal acquisition processing transmission terminal in real time, analyzes and judges the data packets, and sends feedback information to the video signal acquisition processing transmission terminal when a video parameter is needed to be adjusted, and the video signal acquisition processing transmission terminal receives the feedback information and adjusts the video parameter. By the method, video resolution and code stream are dynamically adjusted, and the transmission characteristic is controlled in real time, so that the packet loss rate is reduced, the definition of a transmitted frame is increased, the utilization rate of a wireless network is improved, and video transmission quality and stability are greatly improved; therefore, the video wireless transmission monitoring system can adjust video transmission code rate and resolution at different times according to the real-time condition of the network.

The invention discloses a resource scheduling method and system, a base station and terminals therefor. The resource scheduling method comprises the following steps: the base station receives a bandwidth request message from a plurality of terminals of the request real-time service, wherein the bandwidth request message carries queuing time delay information of real-time service data in the data queue of the corresponding terminal; and the base station determines the priority of the terminals according to the queuing time delay information of the terminals and carries out the resource scheduling on the terminals according to the priority of the terminals. By introducing the data queuing time delay factor in the process of uplink real-time service resource distribution, compared with the prior art, the base station can distribute resources to the terminals with large data queuing time delay in time to reduce the phenomenon of dropped-out data, thereby fairly and effectively distributing resources to the terminals and enhancing the user experience.

The invention discloses a method for determining a timing reference cell during carrier aggregation. By the method, the reliability and accuracy of timing reference can be guaranteed, and the packet loss during data transmission can be reduced. The method provided by the embodiment of the invention comprises the following steps of: judging whether the current timing reference cell of a secondary time aligning group (sTAG) meets effective conditions by a mobile terminal, if so, continuously taking the current timing reference cell as a timing reference cell to be used, otherwise, sending a timing reference changing request for changing the timing reference cell to a base station by the mobile terminal; selecting a new random access cell for the sTAG of the mobile terminal according to the timing reference changing request by the base station; launching random access to the mobile terminal in the new random access cell; and determining that the new random access cell is the changed used timing reference cell by the mobile terminal after the random access is launched successfully.

The invention provides a method and device for realizing software bypass in a firewall. The method comprises a step of judging whether a current network traffic flow is larger than or equal to a preset flow threshold, a step of calling a pre-packaged software bypass operation interface in a message receiving end function or an important safety service processing function if the current network traffic flow is larger than or equal to a preset flow threshold such that the software bypass operation interface obtains a current CPU use rate and a current network port receiving rate through calling a pre-packaged calculation interface and the software bypass is carried out between preset paired software bypass network ports according to a preset rule when the current CPU use rate and the current network port receiving rate are larger than a current hardware equipment platform safety service processing packet loss threshold value. According to the method and the device, the problem of network jam or disconnection caused by equipment packet loss when current network flow is large or the sudden traffic of a peak time appears and the current network flow reaches the maximum performance upper limit of the safety service processing of a firewall in the prior art is solved.

The invention provides a method for controlling a communication bus for solving the problem of great collision occurrence probability caused by traditional technological methods. The method comprises the following steps: generating mutually different delay times on at least two line cards; delaying according to the mutually different delay times when detecting that the communication bus is idle by line cards with communication requests; and preempting the current idle communication bus by the line cards with the communication requests according to the respective delay times and sending data to the communication bus. The invention also provides a device for controlling the communication bus. The method and the device lower the collision probability of the bus to a maximum extent, reduces the data packet loss ratio caused by collision, improves the working stability of the bus and enables corresponding equipment to work more reliable and stable.

The utility model discloses a data loss solution, which comprises 1. Create a SN -PDP buffer; 2. Each SN -PDP buffer is connected with one or a couple of N-PDU buffers which can receive the SN -PDU piece as per the packet number of SN PDU piece. When the receiving timer of N-PDU buffer is timing out and failing to receive all SN -PDU pieces, the N-PDU can release the buffer occupied by the SN -PDU piece; 3. Each N-PDU buffer is connected with one or a couple of SN -PDU piece buffers; 4. Each SN -PDP buffer is connected with one or a couple of buffers which lose the memory. The packet number of the SN -PDU piece without the complete package shall be left in the memory buffer via SGSN. Also, as per the packet number, the unconfirmed upper pieces with the same packet numbers can be discarded. The utility model solves the upper data package loss of SGSN via non-confirming method.

The embodiment of the invention belongs to the communication field and discloses a service access control method, a wireless access system and an access control device which are used for improving system performance. The method of the embodiment comprises the following steps: acquiring the sent data quantity of each service within refresh time, and determining the bandwidth occupation parameter of each service within the refresh time according to the data quantity; determining the service meeting an adjustment condition according to the bandwidth occupation parameter of each service and each corresponding bandwidth occupation parameter threshold; and adjusting the access priority level of the service meeting the adjustment condition. The embodiment of the invention can adjust the access priority levels of different services according to bandwidth occupation ratios, thereby balancing the access priority level of each service, decreasing the packet loss ratios of the services with low priority levels, and improving the system performance.

The invention relates to a classified pack distributing method based on service quality. Wherein, it comprises: setting the allowance time delay met the demand of service quality QoS for the priority queue; calculating the distributing weight of each priority queue; said weight is directly rationed with the signal channel quality parameter, and is directly rationed with the waiting time of group data pack in the queue, and is inversed rationed with the allowance time delay of queue; then distributing the group data pack of queue with highest weight. The invention can improve the throughout of region and reduce the pack loss.

The present disclosure relates to a communication path control technology, and more specifically, to technology for selecting a network and scheduling packets for multi-homing based transmission. The objective of the present disclosure is to reduce a packet loss by determining the number of packets to be transmitted based on the degree of congestion on each network path for a multi-homing based transmission technology. A packet transmission apparatus selects a network path to be used to transmit a packet based on congestion information, which is calculated based on state information of each network path, and then allocates the number of packets to be transmitted and sequence numbers of the packets to the selected network path.

The invention discloses a method and a device for calculating packet loss rate as well as a method and a device for controlling network transmission, belonging to the field of network transmission quality control. The packet loss rate can be calculated and the network transmission quality can be improved under the circumstances of network bandwidth fluctuation. The method for calculating the packet loss rate comprises the following steps of: calculating the packet loss rates of multiple time periods within a preset time range, and calculating the change rate of the packet loss rate of each time period relative to the packet loss rate of the previous time period; judging whether the change rate is in a preset first value range, and acquiring the packet loss rates with the change rates in the preset first value range; and calculating the average value of the packet loss rates acquired within the preset time range. The methods and devices are mainly used for detecting and controlling network transmission quality.

The invention provides a method and a system of flow control based on exchanger cache allocation. The method comprises the following steps: S1, initializing a domain of a congestion window in a packet before a sending terminal sends a data packet; S2, counting and maintaining total data flows N of a port with the exchanger; S3, when the exchanger receives the data packet, calculating the value of the congestion window of the data flow according to size of a caching pool at an outlet port; S4, comparing and determining the value of the congestion window; S5, sending the data packet to a receiving terminal with the exchanger, and sending a feedback packet to the sending terminal from the receiving terminal, wherein the feedback packet comprises the value of congestion window; and S6, when the sending terminal receives the feedback packet, giving the value of the congestion window in the feedback packet to the value of the congestion window of the sending terminal data packet. According to the invention, exchanger cache is allocated, and the bandwidth for each flow is distributed, which realizes the characteristics of quick convergence and less packet loss, and solves various problems due to great delay of short flow and packet loss.

The invention discloses a rate reduction parameter optimization method based on a congestion degree factor. The rate reduction parameter optimization method is applied to a point-to-point multimedia real-time communication system based on an RTP/RTCP protocol. The method comprises the following steps of the receiving end adaptively selects a congestion threshold value according to the inter-frametime delay change condition, generates a state signal S by combining the queuing time delay change condition of the current frame and the congestion threshold value, and adjusts the sending rate according to the state signal. When the network is congested, the congestion degree is quantified, a rate reduction parameter is calculated according to the congestion degree, and the sending rate is controlled to be reduced as the step length; and finally,the final sending rate is calaculated by combining the estimated values of the rate control modules of the sending end and the receiving end to thesending rate. Compared with the prior art, the method can be suitable for various network environments, the incoming rate can be increased, the packet loss rate and the network delay can be reduced, and more stable and smooth multimedia real-time communication service can be brought to users.

The invention discloses a data forwarding method, device and system which are used in a two-layer virtual private network system. The two-layer virtual private network system comprises provider edges (PEs) and a customer edge (CE), wherein the main PE and the backup PE form a cross-node redundancy group (RG). The method comprises the following steps: automatically establishing a data channel for link protection with the backup PE; monitoring a current data transmission link; and when the current data transmission link fails, performing data forwarding through the backup PE and the data channel for link protection. According to the invention, the link delay as well as the packet loss can be reduced.

An improved three-way handshaking method in a Mesh network based on IEEE802.16 relates to the communication field. The improved method is provided aiming at the problem of slot collision during data transmission among nodes under the three-way handshaking mechanism in the universal IEEE802.16 protocol and specifically comprises the followings: a 1-hop neighbor node list, a whole neighbor list, anavailable transmission slot list of the node and a transmission slot list are established at each node; the node receives a GRANT message sent by the neighboring node and processes the information inthe GRANT message according to whether the node is the destination node of the GRANT message under the following two conditions: if the node is the destination node, the minislot in the GRANT messageis compared with an available data transmission slot list; if the node is not the destination node, the minislot in the GRANT message is compared with the transmission slot list, and if collision is found, three-way handshaking is carried out again. The method can effectively avoid the problem of slot collision under the original mechanism, thus reducing the packet loss caused by the collision andincreasing the network throughput.