Abstract: A Random Early Detection (RED) policer in accordance with the invention does not tail-drop arriving packets as being non-conforming. For instance, because the RED policer uses a running estimate such as an exponential weighted moving average (EWMA), for example, it allows the policer to absorb traffic bursts. The policer uses randomization in choosing which packets to drop; with this method, the probability of dropping a packet from a particular sending node is roughly proportional to the node's bandwidth share, hence the packets of different flows are fairly dropped.

Abstract: 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.

Abstract: In a particular embodiment, the present disclosure is directed to a data communications system. The data communication system includes a plurality of digital subscriber lines, a digital subscriber line multiplexer coupled to each of the plurality of digital subscriber lines; and a data switch coupled to the digital subscriber line multiplexer via a communication link. The data communications system is configured such that the number of digital subscriber line users supported by the digital subscriber line multiplexer is determined based on an estimated maximum number of users, the estimated maximum number of users determined based on an average peak bandwidth per user value, a data communication capacity of the communication link, and a data transmission slowdown indicator.

Abstract: In a particular embodiment, the present disclosure is directed to a data communications system. The data communication system includes a plurality of digital subscriber lines, a digital subscriber line multiplexer coupled to each of the plurality of digital subscriber lines, and a data switch coupled to the digital subscriber line multiplexer via a communication link. The data communications system is configured such that the number of digital subscriber line users supported by the digital subscriber line multiplexer is determined based on an estimated maximum number of users, the estimated maximum number of users determined based on an average peak bandwidth per user value, a data communication capacity of the communication link, and a data transmission slowdown indicator. The communication capacity is based on a user type selected from a set of available user types.

Abstract: A call admission control technique allowing flexible and reliable call admissions at an ATM switch in the case of an ATM network including both QoS-specified and QoS-unspecified virtual connections is disclosed. In the case where a QoS (Quality of Service) specified connection request occurs, an estimated bandwidth is calculated which is to be assigned to an existing QoS-unspecified traffic on the link associated with the QoS-specified connection request. A call control processor of the ATM switch determines whether the QoS-specified connection request is accepted, depending on whether a requested bandwidth is smaller than an available bandwidth that is obtained by subtracting an assigned bandwidth and the estimated bandwidth from a full bandwidth of the link.

Abstract: An improvement for a phasor fragmentation engine and method, whereby a phasor flipping algorithm is applied when determining fragment phasors for a non-constant envelope modulation signal (e.g. OFDM). The phasor flipping algorithm avoids sharp phase transitions for the fragment phasors, which cause an increase in bandwidth, by performing a comparison of the phasor angle separation between the prior and current time samples. This comparison corresponds to a determination of whether the modulation signal V has passed near or through zero.

Abstract: Methods and apparatus providing information about characteristics associated with a network are described. A network node associated with a client for selecting a server that can best provide content to a client transmits a request for a response. A plurality of content servers or content routing agents receive requests corresponding to the request from the network node associated with the client. The content servers provide a response and fragment and/or pad datagrams associated with the response. The content servers transmit these fragmented and/or padded datagrams to the network node associated with the client. The network node associated with the client can reassemble and/or trim the received datagrams and identify as optimal the server that first completes transmission of the response.

Abstract: A network system for actively controlling congestion to optimize throughput is provided. The network system includes a sending host which is configured to send packet traffic at a set rate. The network system also includes a sending switch for receiving the packet traffic. The sending switch includes an input buffer for receiving the packet traffic at the set rate where the input buffer is actively monitored to ascertain a capacity level. The sending switch also includes code for setting a probability factor that is correlated to the capacity level where the probability factor increases as the capacity level increases and decreases as the capacity level decreases. The sending switch also has code for randomly generating a value where the value is indicative of whether packets being sent by the sending switch are to be marked with a congestion indicator.

Abstract: The present invention is related to a Rake receiver (R1) for receiving a radio signal and to a searcher (S8a, S8b) for estimating the delay profile of one or more radio links. The searcher includes means for estimating delay profiles in separate windows. The searcher is arranged for being switched into a single window mode or alternatively into a multiple window mode. In single window mode the delay profiles of separate radio links are estimated in the windows, while in multiple window mode two or more windows are allocated adjacent in time for estimating the delay profile of the same radio link. Thereby, the use of the resources of the searcher can be adapted to the radio environments in which the searcher will be used. The Rake receiver comprises fingers that are grouped into two sets. Within a set of fingers the propagation paths are time aligned. The output from the two set of fingers are time aligned and then combined.

Abstract: There is provided a method for transmitting non-real time traffic in a connection oriented communications network. The network includes a network core that has a source and a destination with a path between them. The non-real time traffic is received at the source from a number of connections and each of the connections is associated with one of many classes of transmission service. The method includes the first step of aggregating at the source the non-real time traffic received from the connections onto the path. The non-real time traffic is transmitted on the path without regard to the connections with which the non-real time traffic is associated. The method includes the second step of segregating at the destination the non-real time traffic so transmitted on the path within the core according to the corresponding connections with which the non-real time traffic is associated.

Abstract: A system includes servers accessing storage through a storage area network. The storage has a plurality of ports which may differ in data transfer speed, and memory resources connected to these ports. Data transmission between a port and a server is monitored. If the amount of data transfer exceeds prescribed limits, then a different port is selected, and that port is used for further communications.

Abstract: The present packet scheduling algorithm gives cellular network operators greater flexibility in adjusting the way resources are allocated among interactive best-effort data users. Best effort data users with different radio link qualities may have different amounts of data delivered to them using the same amount of radio resource. In the context of link adaptation, this characteristic complicates the fairness issue in cellular environments and has a profound impact on the overall system performance. As a result, the present packet scheduling algorithm is capable of allocating radio resource dynamically, not only based on channel conditions, but also to achieve different performance trade-offs among users with different link qualities. According to the algorithm, channel quality is determined for each user, channel efficiency is calculated and the channel efficiency value is used as the primary factor in weighting the delivery of packets to (or from) a given user.

Abstract: A method for managing an asynchronous transfer mode (ATM) cell includes transmitting an ATM cell from a server system to a broadband modem. The ATM cell is forwarded from the broadband modem to a client system.

Abstract: A communication system that includes multiple nodes controls a flow of data from a first node of the multiple nodes to a second node of the multiple nodes without relying on an estimate of a rate at which data is drawn from a buffer of the second node and such that an overflow and an underflow of the buffer is avoided. The second node determines multiple flow control parameters, including a current occupancy (Q) of the buffer and an upper threshold (U) and a lower threshold (L) for an occupancy of the buffer and determines a desired data rate (r) based on the multiple flow control parameters. The desired data rate can be used to adjust a data rate for the flow of data. In another embodiment, the communication system further dynamically controls a rate at which flow control messages are conveyed by the second node to the first node.

Abstract: A method for handling call set-ups in a telecommunications network. The method includes receiving a request to set up a call. A dynamic load value is compared to a load threshold. The call is allowed to be set up when the dynamic load value is less than the load threshold.

Abstract: A method for adaptively partitioning a buffer in a shared buffer switch is provided. The buffer partitioning method for a shared buffer switch which has a plurality of input ports, a plurality of output ports, and a shared buffer, the method for determining whether or not to store a cell, which is newly received through one of the input ports, in the shared buffer comprises the steps of (a) determining a buffer area of the shared buffer in which the newly received cell is stored; (b) determining a cell discard threshold with respect to the total number of cells stored in the shared buffer and the changing rate, with respect to time, of the total number of the cells; and (c) determining whether or not to store the newly received cell in the shared buffer, by comparing the number of cells stored in the buffer area in which the newly received cell is to be stored, with the cell discard threshold.

Abstract: A method for dynamic division of the radio capacity in a Time Division Multiple Access (TDMA) system between different packet radio services. Each packet radio service is composed of packets with the same or different priority. Several MSs can share a time slot. Information regarding to the interference for each MS is obtained by a request from a mobile station (MS) or through a traffic measurement at the base transceiver station. This information, along with the QoS contract, is used as criteria in allocating more time slots to packet radio service. A framework of dynamic scheduling for mobile communication (DSMC) comprising a Channel Codec Unit (CCU) and a Packet Control Unit (PCU) determines the service rate upon the required QoS of a user and the interference level.

Abstract: A method and a system using the method enabling to decrease required resource for the bandwidth reservation in an inter-site connection network used for communication between communication sites. The aggregation of bandwidth is considered for allocating bandwidth in the inter-site connection network. The network consists of a plurality of nodes each having a packet switch, and a plurality of user sites each constituted by a host or an internal network respectively connected to the plurality of nodes via access lines. With respect to the plurality of user sites, when one user site is to be connected to the other site through the network, a necessary and sufficient bandwidth for interconnecting between the user site and the other is calculated according to the bandwidth of the access line connecting the user site to the network considered as a minimal bandwidth. Based on the calculated result, the bandwidth is allocated to the plurality of nodes.

Abstract: An operating method of a storage area network system which conducts transmittance of data by a plurality of servers accessing a storage through a storage area network, the storage area network system operating: obtaining and monitoring a data transfer amount between the respective servers and the storage; comparing a prescribed data transfer amount prescribed by the server and the obtained actual data transfer amount; selecting one of a plurality of ports provided in the storage according to the comparison result; and providing the relevant server an instruction to connect to the selected port.

Abstract: In a class of minimum cost flow control (MCFC) algorithms for adjusting session rates or window sizes congestion control is achieved through consideration of an incremental cost function that addresses link congestion, and an incremental cost function that addresses the cost of providing less than the desired transmission rate. A coarse version of the algorithm is geared towards implementation in the current Internet, relying on the end-to-end packet loss observations as indication of congestion. A more complete version anticipates an Internet where sessions can solicit explicit congestion information through a concise probing mechanism.

Abstract: A computer dispatcher connected to one or more respective network buffers has stored file lists that identify one or more of the files in the database that are to be transmitted over networks connected to the respective network buffer. A scheduler(s) schedule one or more portions of one or more of the files to be written to the respective network buffers by defining transmission criteria about each of the files in the file list. These transmission criteria include a quantity to transmit criteria, defining a quantity of one or more of the portions of the respective file to transmit, and one or more release times. The release times define the time at which the respective portion is to be written to the network buffer. The system includes a dispatching process that determines an available space on one or more of the network buffers and a current system time.

Abstract: It is an object of the invention to measure a communication data volume for each service type in measuring a communication data volume in a charging process and the like. The invention comprises a plurality of measuring points 131 to 133 for measuring a communication data volume and a data relaying apparatus 120 for transmitting received data to any of the plurality of measuring points 131 to 133. The data relaying apparatus 120 has a service-type judging means 121 for judging the service type of the received data, a measuring point determining means 122 for determining the measuring points 131 to 133 to which the data is to be transmitted based on the service type and a data transmission means 122 for transmitting the data to the measuring point thus determined, and each of the plurality of measuring points 131 to 133 has a communication data measuring means for measuring the communication data volume of the data received from the data relaying apparatus 120.

Abstract: In recent years adaptive modulation has emerged as a popular technique to improve data throughput and system capacity in a wireless system. The basic idea is to adapt the modulation scheme to the fading channel quality, using different schemes for different channel conditions. Therefore one primary issue is to determine the switching thresholds between the modulation schemes. Typically these thresholds are fixed according to a certain criterion. This paper introduces a novel adaptive learning approach that is capable of dynamically adjusting the thresholds so as to maximize the throughput. A key feature of the proposed self-learning scheme is that no dedicated training signal is required, instead it utilizes the long-term average throughput to continuously update the thresholds as the data is transmitted.

Abstract: A method, article of manufacture and system to determine a bit rate of a signal is described.

Abstract: A method and apparatus for controlling congestion at a node in a data network. The node includes an input for receiving traffic units from the network, an output for releasing traffic units to the network and a control unit. The control unit is responsible for estimating a level of data occupancy of at least a portion of the network by looking at the traffic units received at the input from a remote node in the network. When the data occupancy level reaches a certain threshold, the node issues a control signal to the remote node such that the remote node lowers its rate of traffic units input in the network. By estimating the network data occupancy level, congestion at the node can be effectively foreseen and controlled.

Abstract: A communication connecting device includes a data storage to which data read by a scanner are written. A packetizer packetizes the data read out of the data storage in accordance with a packet size fed from a size information storage. A rate controller counts a period of time after the start of packet transmission or a period of time consumed to send a preselected number of packets. The rate controller calculates a communication rate on the basis of the above period of time and the number of packets sent, and then compares the communication rate with a communication rate assigned to a receiving terminal and defined by a Recommendation T.30 protocol. The rate controller allows the packets to be sent only if the calculated communication rate is lower than the communication rate of the receiving terminal.

Abstract: A device that passively monitors arriving and departing data packets on one or more networks, correlates arriving data packets with departing data packets, and calculates a latency estimate based on the confidence of the correlation. The device detects and copies data packets arriving at a network device and the data packets departing from the same network device. A timestamp is stored for each arriving or departing data packet. Latency across a network device can be determined based on the timestamps for correlating data packets. Additionally, latency across a network device per protocol layer can also be calculated. Varying levels of confidence of a latency estimation depend on the operation necessarily performed on the data packet by the network device and the protocol level at which correlation between the arriving and departing data packets can be achieved.

Abstract: A traffic pattern of data packets that originate at a traffic source and are transmitted through one of multiple ports is monitored. A parameter value characterizing fluctuations a in a transmission rate of data through the port relative to a transmission rate for the monitored traffic pattern is generated, and data packets from the traffic source are allocated to at least one other port for transmission based on the first parameter value.

Abstract: The power budgeter controls the rate of data flow in one or more transceivers among a plurality of transceivers. The power budgeter monitors aggregate transceiver activity (data flow) during a predefined period, and limits transceiver data transmit and receive rates during times when the aggregate transceiver activity would otherwise exceed a predefined threshold. If total data flow during the monitored time period exceeds the threshold, data flow through the active transceiver during the next (second) time period is limited. Data flow rates are limited in a transceiver by shutting off transceivers (receiver and/or transmitter) for a specified portion of the second time period. In other types of transceivers, the transmitter and/or receiver may be forced into a low-power idle state or similar operating condition for a specified portion of the second predefined time period.

Abstract: A method of transmitting data packets, in particular of real-time or near-real-time data over internet protocol (IP) networks using a real-time protocol (RTP) for media data products and a real-time control protocol (RTCP) for control data packets. Each protocol is allocated a fraction of the available transmission bandwidth. The method comprises the steps of measuring relevant characteristics of the network link, calculating from this measurement an optimized RTCP bandwidth and transmitting control data packets using the optimized RTCP bandwidth.

Abstract: A startup management system and method, particularly adapted for use in computer and other communication networks, is presented. Rate-based flow and congestion control mechanisms have been considered desirable, including to deal with the needs of emerging multimedia applications. Explicit rate control mechanisms achieve low loss because of a smooth flow of data from sources, while adjusting source rates through feedback. However, large feedback delays, presence of higher priority traffic and varying network conditions make it difficult to ensure feasibility (i.e., the aggregate arrival rate is below the bottleneck resource's capacity) while also maintaining very high resource utilization. The invention applies entry and early warning techniques which increase the initial connect rate of newly connecting sources.

Abstract: A router and method for routing table cache population technique is disclosed. In particular, the illustrative embodiment routes packets through it more quickly than comparatively expensive routers in the prior art. The present invention recognizes that a fast router has small routing table cache that has a high hit ratio and that a high hit ratio can be achieved with a small routing table cache by predicting which entries will be needed in the routing table cache in the future and by populating the routing table cache with those entries before they are needed.

Abstract: A communication service provision system, method and device to provide a communication service which provision various communication devices to realize communication service ordered by a customer and reduce the load of an operator by utilizing a policy server. In the invention, the following are performed. A customer care server executes order processing suitable for communication service ordered by a customer. An operational flow manager activates a design server based upon contents stored in an operational flow storage when the order processing is finished and instructs the design server to execute route computing. When the operation is finished, a policy server sets configuration data to respective communication devices composing a route suitable for the order of the customer. Network design can be also performed by inputting route data and policy data from the policy server without using the design server.

Abstract: A method related to a data communication system, and more specifically to frame relay multiplexing, in which system data frames are forwarded by nodes performing inter alia multiplexing and concentration of traffic from a plurality of input sources, and wherein the sum of transmission time in question, the queue delay and acknowledge frame transmission time, should not exceed a transmission timer setting, and for the purpose of solving the problem with queuing of data frames in such a communication network, it is according to the present invention suggested to combine the following steps: a) simulating an output queue length, b) estimating an input source traffic value, and c) using fuzzy logic, for thereby deciding when a data frame has to be deleted from the data stream.

Abstract: Monitoring of customer bandwidth demand is accomplished by comparing mean access router uplink utilization to a prescribed threshold value that is computed based upon granular measurements of uplink utilization. This provides an indication of whether customer demand is met with a statistical level of confidence. Further, the actually measured access router uplink utilization can be used to make a predictive assessment of whether to provision a new customer on the access router without adversely affecting existing customers.

Abstract: An acceptor for admitting incoming requests to a server application includes a session manager that determines the class of an incoming request. The class includes a first class and a second class. A queuing module is provided to store the request if the incoming request is of the second class. A priority control module is provided to ensure that a predetermined number of requests are sent to the server application for service in each cycle. The priority control module allows (1) the predetermined number of the first class requests to be sent to the server application if the first class requests received in a cycle are at least equal to the predetermined number, and (2) a mixture of the first class requests and the second class requests to be sent to the server application if the first class requests received in a cycle are less than the predetermined number. A method of handling incoming requests to the server application is also described.

Abstract: A method/system for estimating cell rate is provided. The cell rate to be reported to an ATM (asynchronous transfer mode) source may be estimated using a previously calculated cell rate parameter weighted in a manner dependent upon whether a rate for the connection is high or low. An averaging parameter may be adjusted for each calculation so that for low rate connections newly measured data is more heavily weighted, and for high rate connections newly measured data is less heavily weighted. Relatively accurate and quick estimates for both high bandwidth and low bandwidth connections are achievable.

Abstract: A method of creating an email usage record is provided. An input signal is received. An email usage record is created in response to the received input signal. An event jacket is created. The event jacket is associated with the email usage record. A command to send an email message is received. A confirmation signal is received after the email message has been sent. Finally, an email leg jacket is created. The email leg jacket includes at least one record. The at least one record includes information relating to the email message.

Abstract: A method for analyzing traffic in a system (100) having a plurality of nodes (110, 120, 130) is provided. The method includes forecasting a total amount of traffic transmitted to and from each of the plurality of nodes (110, 120, 130) based on observed traffic statistics and an estimated growth. The method also includes generating a traffic matrix using the observed traffic statistics and the forecasted traffic associated with each of the plurality of nodes (110, 120, 130). The method may also include using the traffic matrix to determine projected traffic between each of the plurality of nodes (110, 120, 130).

Abstract: There is provided a method for calculating the call processing capacity of a mobile communication system using an Internet communication network. The present invention calculates and stores the call processing capacity of a mobile communication system using an Internet web browser and database to attain accurate call processing capacity information, thereby obtaining an optimal mobile communication system design. Furthermore, since multiple designers can share information about the calculated call processing capacity through a database, thereby making the call processing capacity information be actively effectively used.

Abstract: A system for user-space packet modification, including a set of kernel code and a user-level application programming interface (API). The system facilitates creation of a special socket for passing packets between kernel space and user space. The system in turn facilitates creation and application of a packet filter associated with the socket, in order to trap incoming or outgoing packets being processed in the kernel at a designated point in a protocol stack. Once a packet is trapped, it is moved through the socket into user space, thereby at least temporarily preventing the protocol stack from further processing the packet. In user space, an application may operate on the packet, for instance, modifying aspects of the packet or deleting the packet altogether. The system in turn facilitates injection of a packet from user space into kernel space, and into a designated point in the protocol stack for desired stack processing.

Abstract: A method for predicting the percentage of real-time a processor in a base station is being utilized (i.e., processor occupancy) based only on calling rates, given measured call control event (i.e., origination, termination, handoff, paging, registration, and locate rates) data for the base station. Constant ratios between the various call control events and the calling rates (e.g., originations+terminations) are determined for the base station. Given the actual time spent by the processor on each type of call control event and the constant ratios, the processor occupancy is accurately predicted using call originations+terminations or call origination rates as variables. The processor occupancy may be fixed at some value (e.g., 90%) and the calling capacity of the base station (i.e., the total number of call originations and terminations which will result in such processor occupancy) is determined from the determined relationship between processor occupancy and calling rates.

Abstract: A network device that is inserted in the path of traffic in a packet network, and the associated procedures and controller algorithms for monitoring the performance of aggregates of short lived Transmission Control Protocol (TCP) Connections flowing over a bottleneck link and dynamically managing their performance. TCP operates by allowing a certain window of data to be outstanding between the source and the receiver of each transfer and if many transfers attempt to share the network, congestion occurs, thus reducing the transmission rate of ongoing transfers. The method and apparatus aim at the performance of an aggregate of short-lived connections and hence measure only the RTT (Round Trip Time) for the aggregate and determine a window for the aggregate. By setting a target performance for the entire aggregate a measurement is made over the aggregate to determine the current performance level for each value of control eg. A RD (Random Drop) probability, or a value of MWA (Modified window Advertisement).

Abstract: The method for handling call set-ups in a telecommunication network includes receiving a request to set up a call. The method also includes comparing a dynamic load value for a switch to a load threshold for the switch. The method further includes allowing the call to be set up when the dynamic load value is less than the load threshold.

Abstract: The present invention includes a system for determining cell loss among on-off sources on a network by (a) receiving cells; and (b) measuring aggregate cell loss of said sources at port level. The method preferably includes determining aggregate cell loss of the sources; determining mean rate of each of the sources; determining burstiness of each of the sources; determining average burst length of each of the sources; determining buffer size; and approximating the cell loss ratio of each of the sources based substantially only upon the aggregate cell loss, the mean rate, the burstiness, and the average burst length.

Abstract: Methods and systems allowing a plurality of computer nodes to communicate using weighted transmission paths are provided. A load balancer distributes packets across weighted transmission paths according to transmission path quality, which is monitored and updated from time to time. As transmission quality on a specific transmission path decreases, the weight assigned to that transmission path is reduced. Similarly, weights may be increased if transmission quality improves. The weights assigned to various transmission paths may correspond to a relative number of packets to be transmitted on each respective transmission path. The transmission path for each packet can be selected based on the weights of the various transmission paths. Using weights based on transmission quality, transmission paths with higher transmission quality are used more often than transmission paths with lower transmission quality, resulting in more efficient communications.

Abstract: Measurement packets are sent at equi-intervals from a sending unit into a network path. The probing packets are received by a reception unit, which measures the packet transmission time to estimate an available bandwidth of a network path from a parameter Q indicative of a correlation in the packet transmission time between adjacent packets.

Abstract: A user interface and computer-implemented method for analyzing and displaying network traffic performance assessment data in a computer network. Packet trace data is organized into application-level threads for display in a visual thread diagram. The user can manipulate the visual thread diagram to configure it in a number of ways in a dynamic manner.

Abstract: A feedback-based adaptive network is described wherein at least a portion of the network elements report operating information relating to network conditions to a centralized data store. The information which is reported to the data store is analyzed by a policy engine which includes a plurality of application specific plug-in policies for analyzing selected information from the data store and for computing updated control information based upon the analysis of the information. The updated control information is fed back to selected network elements to thereby affect operation of the selected elements. Typically, when the operation of a network element has been affected, its corresponding operating information will change. The new or changed network element operating information is then reported to the data store and analyzed by the policy engine. The policy engine may then generate new or updated control information for affecting the operation of selected elements in the network.

Abstract: A system receives a flow of data packets via the link and determines a target bandwidth to be allocated to the flow on the link. In response to the flow, the receiving system transmits data to the sending system. The transmitted data control the sending system such that when the sending system transmits subsequent data packets to the receiving system, such subsequent data packets are transmitted at a rate approximating the target bandwidth allocated to the flow. In one embodiment, the rate at which the transmitted data from the receiving system arrive at the sending system determines the rate at which the sending system transmits the subsequent data packets. The receiving system can control the rate by delaying its response to the sending system according to a calculated delay factor. In another embodiment, the data transmitted from the receiving system to the sending system indicate a maximum amount of data that the receiving system will accept from the sending system in a subsequent data transmission.