1454 results about "Network congestion" patented technology

One embodiment of the present invention provides a system that avoids network congestion. During operation, the system can detect an onset of congestion in a first queue at a first node. Next, the first node can generate a first control-message, wherein the first control-message contains a congestion-point identifier which is associated with the first queue. The first node can then send the first control-message to a second node, which can cause the second node to delay sending a second message to the first node, wherein the second message is expected to be routed through the first queue at the first node. Next, the second node may propagate the control-message to a third node which may cause the third node to delay sending a third message to the second node, wherein the third message is expected to be routed through the first queue at the first node.

A new Fast Recovery Plus (FR+) mechanism, and associated method, for wireless and/or mobile network applications to control data flow and avoid network congestion in a TCP/IP packet-switched network.

The present invention relates to a method and network for controlling congestion in a packet-switched network, comprising traffic sources, traffic destinations and network nodes, wherein a packet queue length in a network node is determined and a congestion notification is transmitted back towards the source address of an incoming data packet received at the network node, if the detected packet queue length exceeds a predetermined threshold. Then, congestion control is performed at a predetermined intermediate network node in response to the receipt of the congestion notification. Thereby, burts of source traffic can be constrained and unnecessary packet losses can be avoided already at an intermediate access node and within the network. The congestion notification message generated due to an incipient congestion is immediately routed back according to its source address. As a result, control delay time is shortened, such that buffer size requirements and number of congestion notification messages are reduced.

The present invention provides a multi-service queuing method and apparatus that provides exhaustive arbitration, load balancing, and support for rapid port failover. Routers and switches according to the present invention can instantaneously direct the flow of traffic to another port should there be a failure on a link, efficiently handle multicast traffic and provide multiple service classes. The fabric interface interfaces the switch fabric with the ingress and egress functions provided at a network node and provides virtual input and output queuing with backpressure feedback, redundancy for high availability applications, and packet segmentation and reassembly into variable length cells. The user configures fixed and variable-length cells. Virtual input and output queues are coupled to a switch fabric. Statistics regarding the virtual input and output queues are collected and packet queuing for the virtual input and output queues is controlled using the collected statistic to provide congestion control for the virtual input and output queues and the switch fabric.

A new concept of soft quality-of-service (soft-QoS) is developed that bridges the gap between the efficient provision of network-level QoS and the requirements of multimedia applications. Soft-QoS is defined by a satisfaction index (a number that rates users' perceptual quality) and a softness profile (a function that captures the robustness of multimedia applications to network congestion). Another aspect of this invention is a bandwidth allocation scheme for multimedia applications with soft-QoS requirements is presented. The implementation of the bandwidth allocation scheme on a network element realizes a soft-QoS controller. The controller uses the connections' softness profiles to compute a bandwidth allocation that maximizes the minimum satisfaction index of active connections.

A method and system for Mobile Internet Protocol (IP) device redundancy. As mobile devices roam away from a home network and change a connective status, mobility binding records are sent to a multicast network address on the home network. The multicast network address multicasts the mobility binding records to other active Mobile IP home agent control nodes, standby home agent control nodes and standby home agents on the home network. The method and system allows standby home agent control nodes or standby home agents to be transparently switched for active home agent control nodes or active home agents that fail without downloading or uploading large numbers of mobility binding records after a failure. The method and system may also help reduce failed calls (e.g., data sessions including Voice over IP (VoIP), H.323, etc.), network congestion and improve user satisfaction in Mobile IP systems.

A system and process for controlling the coding bit rate of streaming media data is presented where a server streams data that exhibits one of a number of coding bit rates supported by the server. Initially, the server chooses the coding bit rate. However, after this startup period, the client provides coding bit rate requests. The server transmits the streaming media data at the most appropriate supported coding bit rate closest to the rate requested. The coding bit rates requested are those estimated to provide a high quality playback of the streaming data while still keeping a decoder buffer of the client filled to a desired level. A leaky bucket model is incorporated so that the changes in buffer duration due to natural variation in the instantaneous coding bit rate are not mistaken for changes in buffer duration due to network congestion.

A scalable video transmission scheme is provided in which client interaction and video content itself are taken into consideration during transmission. Methods and arrangements are provided to prioritize/classify different types of information according to their importance and to packetize or otherwise arrange the prioritized information in a manner such that lower priority information may be dropped during transmission. Thus, when network congestion occurs or there is not enough network bandwidth to transmit all of the prioritized information about an object, some (e.g., lower priority) information may be dropped at the server or at an intermediate network node to reduce the bit rate. Thus, when the server transmits multiple video objects over a channel of limited bandwidth capacity, the bit rate allocated to each object can be adjusted according to several factors, such as, e.g., information importance and client interaction.

Relay units (1002-1-1002-n) terminate a plurality of transport layer connections from terminals in a transport layer and relay each of these connections to other transport layer connections (relay connections). A transmission rate control unit (1003) determines the total transmission rate of the relay connections according to the number of connections that are being relayed and network congestion conditions such that a desired effective rate is obtained, and moreover, allots the total transmission rate as transmission rates to each of relay units (1002-1-1002-n) in accordance with the results of the analysis of information of the applications being carried on the transport layer connections during relay.

One embodiment of the present invention provides a system that facilitates congestion management in a Fibre Channel (FC) network. During operation, the system determines a threshold data rate on an outgoing link coupled to an FC switch. The system further determines the number of sources that send data to the outgoing link and an aggregate arrival rate of data for the outgoing link. Next, the system determines an injection data rate for a respective source based on the threshold data rate on the outgoing link, the number of sources transmitting data to the outgoing link, and the aggregate arrival data rate for the outgoing link. Subsequently, the system communicates the injection data rate to the source, thereby allowing the source to throttle its data injection in the FC network to prevent network congestion.

Methods and apparatus are provided for controlling congestion in a network such as a fiber channel network. Techniques are provided for detecting congestion at a shared resource such as a network node. When a controller sends data to a shared resource, the delay between sending the data and receiving a credit is measured. If the delay is significant, it is assumed that the path towards the destination associated with the data is congested and subsequent traffic from the port associated with the data is blocked.

A system and method for providing quality of service (QoS) service over a mobile IP network with dynamic domains, multiple or distributed QoS managers per domain and/or with network congestion feedback being used to establish an estimated total domain bandwidth which is used for regulating access to a domain.

An admission controller and its method of operation for controlling admission of data flows into an aggregate data flow are described. Admitted data flows are aggregated into an aggregate data flow for transmission by a router, for example, over a data network. The aggregate data flow typically follows a pre-established path through the network; for example a multi-protocol label switched (MPLS) path. The path has minimum and maximum bandwidth limits assigned to it. The admission controller controls admission of new data flows into the aggregate data flow by granting or denying new session requests for the new data flows. Congestion notifications received from the network and bandwidth limits of the path are considered in determining whether to grant or deny a new session request. In this way, the admission controller provides elastic sharing of network bandwidth among data flows without exacerbating network congestion and while remaining within the path's bandwidth limits. The data flows may include transaction oriented traffic.

The invention herein described consists of an algorithm, which performs an end-to-end estimation of the bandwidth available in an end-to-end connection established between a server and a client via a packet switching network such as the Internet Protocol Network (IP). This algorithm is used to properly regulate the input rate at the send side. Typical applications are delivering best effort traffic over TCP, or audio and video traffic over RTP/UDP. The invention is particularly effective over wireless Internet and can be used in a content delivery system to dynamically choose the best server between a set of servers to satisfy the request of a client, or to select the best route in a content deliver or global hosting system.

Methods and apparatus are provided for controlling congestion in a network such as a fibre channel network. Techniques are provided for characterizing traffic flow at a congested network node. The congested network node can generate various instructions such as quench messages to control traffic flow towards the congested network node. The quench messages can optionally include information about the characteristics of the congestion. The instructions are distributed to other nodes in the network. The other network nodes can interpret the instructions and control traffic flow towards the congested node.

A queue length measurement device that is comprised of a number of queues that are capable of holding data cells. A differential counter is coupled to each queue. The counter is incremented when a cell is written into the queue and decremented when a cell is read from the queue. An interval measurement device, coupled to the differential counter, generates a pulse to reset the counter at fixed intervals equivalent to n cells time (where n is the maximum number of cells the queue counter can measure). A multiplexer is coupled to the multiple differential counters. A transfer control circuit coupled to the interval measurement device selects the appropriate queue measurement to be output from the multiplexer to the other switch elements. A system and methodology provide for queue flow statistics and closed loop control of cell flow into the queue. A congestion control system is provided comprising a queue (having an input and an output, and capable of storing and outputting a plurality of data cells), an input processor (for coupling a plurality of data cells to the queue input), an output processor, an interval measurement device (generating a pulse at predetermined intervals), and differential queue length generation logic. The output processor, couples to the queue output, for receiving a plurality of data cells from the queue, which generates a queue change in size signal in response to determining for each of the predefined intervals of the difference between a present and a previous queue length.

The invention discloses a layered distributed cloud computing architecture and a service delivery method. The architecture is composed of a core cloud, an edge cloud and a terminal. The core cloud is a cloud computing service center composed of servers with powerful storage and computing capacity; the edge cloud is composed of server nodes distributed in different networks and regions and is connected with the core cloud by backbone network, and the terminal is an entity which uses a cloud computing service provided by the core cloud and edge cloud. The core cloud and the edge cloud work together to provide the cloud computing service for terminal users and uses relevance of data transmitted by the core edge and the terminal to process the data so as to reduce traffic. In the invention, network flow in most of cloud computing is confined to the local network between the edge and the terminal, thus lowering traffic of the backbone network, effectively avoiding the problems of network congestion, high latency and the like in the existing centralized cloud computing mode and improving user experience.

A system provides congestion control and includes multiple queues that temporarily store data and a drop engine. The system associates a value with each of the queues, where each of the values relates to an amount of memory associated with the queue. The drop engine compares the value associated with a particular one of the queues to one or more programmable thresholds and selectively performs explicit congestion notification or packet dropping on data in the particular queue based on a result of the comparison.

A flow based routing method and apparatus selects a path from a plurality of different paths for assignment to a flow. The path is selected based on a traffic performance measurements which identify relative congestion and performance of the different paths, so that traffic flows can be diverted away from network congestion points, thereby allowing network resources to be load balanced at a flow granularity. The present invention may be configured on physical or virtual links on an NE to enhance the forwarding of packets using the primary link and one or more alternate links to any given destination.

Methods and systems for detecting and predicting congestion patterns from network feedback are described. In the inventive embodiments, network feedback in the form of binary feedback messages is received. The network feedback can comprise any suitable feedback messages and, in the illustrated and described embodiments comprise packet loss events or ECN marks, to name just a few. The inventive embodiment uses the network feedback to predict when congestion is likely to occur in the future. Based on these predictions, the behavior of end users within the system can be modified so that congestion is reduced. Specifically, the data packets that are typically transmitted by the end users can be delayed until such a time when the predicted congestion is mitigated. Thus, the inventive systems and methods take a predictive, proactive approach to addressing network congestion issues, rather than a reactive approach.

A system for coding voice signal to optimize bandwidth occupation in a High Speed Packet Switching network while ensuring best voice transmission quality. The voice signal is first encoded using a conventional GSM like RPE/LTP coder providing first sub-frames of coded signal and tagging these first sub-frames as being non-discardable. In addition, a convenient difference between an RPE/LTP provided signal and a corresponding synthesized image is performed (see 36) and is also block encoded into second sub-frames which second sub-frames are tagged as being discardable sub-frames. Said second sub-frames when concatenated to corresponding first sub-frames provide so-called multirate frames. Then, when transmitting said multirate frames over the High Speed packet switching network, dropping discardable tagged data enables solution network congestion situations in any network node and at random with no significant disturbing effect over the voice communication operation.

Layered Video Multicast with Retransmissions (LVMR) is a system for distributing video content using layered coding over the Internet. The two key contributions of the system are (1) improving the quality of reception within each layer by retransmitting lost packets given an upper bound on recovery time and applying an adaptive playback point scheme to help achieve more successful retransmission, and (2) adapting to network congestion and heterogeneity using a hierarchical rate control mechanism. A hierarchical rate control system and method is disclosed for regulating the flow of video information in a network by establishing subnet agents for each subnet, and by establishing intermediate agents for each domain. The subnet agents gather information concerning add-layer experiments conducted by receivers in a subnet, and the information is provided to intermediate agents of the domains for regulating data traffic.

A packet network congestion control system using a biased packet discard policy includes a plurality of end points having codecs operating in a framework, e.g. ITU-T H.323 protocol to establish a communication session. The protocol enables the codecs to negotiate codec type and associated parameters. Once a connection and session are established, compressed voice and data packets start flowing between the two end points. A control entity supplies congestion control packets periodically. The control packets provide a "heartbeat" signal to the codec at the other end of the session. Each codec receiver uses the "heartbeat" signal as an indication of network congestion. As network congestion increases, routers within the network discard excess packets to prevent network failure. The network discards all packets classified as congestion control packets whenever a flow control mechanism detects congestion or a trend toward congestion. As packets are discarded, the end points renegotiate codec type and/or parameters to realize lower bit rates.

A wireless terminal comprising a radio transceiver for communicating with a base station, an infrared transceiver for communicating with another wireless terminal, a congestion control unit for detecting whether radio communication with the base station is in a collision-free state, and a communication control unit for performing radio communication with the base station by the radio transceiver after detecting unoccupied state in the radio channel by the congestion control unit and transmitting identification information to another wireless terminal by the infrared transceiver when the radio communication with the base station is completed.

A method and system are provided for automatically rerouting data from an overbalanced logical circuit in a data network. Status information is received for a logical circuit communicating data in the data network. The status information may include a forward explicit congestion notification (“FECN”) and a backward explicit congestion notification (“BECN”) indicating an overbalanced condition in the logical circuit. An overbalanced condition is indicative of network congestion in the logical circuit. A logical failover circuit including an alternate communication path for communicating the data in the overbalanced logical circuit is then identified. The data in the overbalanced logical circuit is rerouted to the logical failover circuit without manual intervention. After rerouting the data to the logical failover circuit, a determination is made as to whether the overbalanced condition in the logical circuit has been cleared. If it is determined that the overbalanced condition in the logical circuit has been cleared, then the data from the logical failover circuit is rerouted to the logical circuit in the data network without manual intervention.

A communication device according to the present invention enhances transfer of time-critical data between one or more LANs and a device (e.g., edge router, etc.) coupled to a backbone network. A virtual bottleneck in the form of a queue is introduced by the communication device at the customer premises or customer end of a backbone network access line where the network congestion or bottleneck resides. The virtual bottleneck delays and/or discards time insensitive traffic prior to time-critical or voice traffic being delayed in the edge router. This is accomplished by the virtual bottleneck queue including a storage capacity or length less than that of a queue utilized by the edge router. A traffic manager or scheduler controlling the virtual bottleneck dynamically adjusts the virtual bottleneck based on the bandwidth required for time-critical packets to ensure sufficient bandwidth is available for those packets.

The invention discloses a media stream self-adapting transporting method based on the internet comprising, a. establishing RTP/UDP/IP protocol structure, wherein the data plane of the protocol structure is in charge of using the media video stream and audio stream packaged by RTP, while the control plane of the protocol structure transmits the control instructions simultaneously using the RTCP feed back network information and RTSP, establishing a congestion control structure based on the transmitting end that can be applied both in the real time coding and in the video frequency transfer storage. The invention can realize the best playback quality possible on the user end under the changes of the network operation.

A distributed computer system includes links and routing devices coupled between the links and routing frames between the links. Each of the routing devices includes a congestion control mechanism for detecting congestion at the routing device and responding to detected congestion by gradually reducing an injection rate of frames routed from the routing device.

The present invention is a delay based model and in fact uses queueing delay as a congestion measure, providing advantages over prior art loss based systems. One advantage is that queueing delay can be more accurately estimated than loss probability. This is because packet losses in networks with large bandwidth-delay product are rare events under TCP Reno and its variants (probability on the order 10⁻⁷ or smaller), and because loss samples provide coarser information than queueing delay samples. Indeed, measurements of delay are noisy, just as those of loss probability. Thus, another advantage of the present invention is that each measurement of queueing delay provides multi-bit information while each measurement of packet loss (whether a packet is lost) provides only one bit of information for the filtering of noise. This makes it easier for an equation-based implementation to stabilize a network into a steady state with a target fairness and high utilization. In addition, the dynamics of queueing delay provides scaling with respect to network capacity. This helps maintain stability as a network scales up in capacity.

Methods and systems for processing network data are disclosed herein and may include receiving from a switching device, a congestion indicator that indicates congestion. In response to the congestion indicator, latency of reaction by a source end point, may be reduced by preventing introduction of queued up new frames to affected flow or CoS before the local stack adjusts its rate to congestion conditions and/or by rate limiting the processing of unprocessed network frames in hardware. The unprocessed network frames may include unprocessed network frames of a particular type. In response to the received congestion indicator, by a destination end point, congestion indicator flags may be set in processed network frames of the particular type, faster than an expected reaction of a local stack. The congestion indicator flags may be explicit congestion notification (ECN)-Echo flags.