Abstract: A method is presented for determining optimal or preferred configuration settings for wireless or wired network equipment in order to obtain a desirable level of network performance. A site-specific network model is used with adaptive processing to perform efficient design and on-going management of network performance. The invention iteratively determines overall network performance and cost, and further iterates equipment settings, locations and orientations. Real time control is between a site-specific Computer Aided Design (CAD) software application and the physical components of the network allows the invention to display, store, and iteratively adapt any network to constantly varying traffic and interference conditions. Alarms provide rapid adaptation of network parameters, and alerts and preprogrammed network shutdown actions may be taken autonomously. A wireless post-it note device and network allows massive data such as book contents or hard drive memory to be accessed within a room by a wide bandwidth reader device, and this can further be interconnected to the internet or Ethernet backbone in order to provide worldwide access and remote retrieval to wireless post-it devices.

Abstract: The paper investigates the effects of packet loss and delay jitter on speech quality in voice over Internet protocol (VoIP) scenarios. A new formula is proposed to quantify these effects and incorporated into ITU-T G.107, the E-model. In the simulation, codecs ITU-T G.723.1 and G.729 are used; random packet loss and Pareto distributed network delay are introduced. The prediction errors range between -0.20 and +0.12 MOS (mean opinion score). The formula extends the coverage of the current E-model, and is very useful in MOS prediction as well as network planning.

Abstract: Network calculus is known to apply in general only to feedforward routing networks, i.e., networks where routes do not create cycles of interdependent packet flows. In this paper, we address the problem of using network calculus in networks of arbitrary topology. For this purpose, we introduce a novel graph-theoretic algorithm, called turn-prohibition (TP), that breaks all the cycles in a network and, thus, prevents any interdependence between flows. We prove that the TP-algorithm prohibits the use of at most 1/3 of the total number turns in a network, for any network topology. Using analysis and simulation, we show that the TP-algorithm significantly outperforms other approaches for breaking cycles, such as the spanning tree and up/down routing algorithms, in terms of network utilization and delay bounds. Our simulation results also show that the network utilization achieved with the TP-algorithm is within a factor of two of the maximum theoretical network utilization, for networks of up to 50 nodes of degree four. Thus, in many practical cases, the restriction of network calculus to feedforward routing networks may not represent a too significant limitation.

Abstract: Techniques for dynamically and optimally selecting one or more computing systems, e.g., a server set, to which another computing system, e.g., a client device, is to be directed. The computing systems may be part of a distributed computing network. For example, such techniques may include the following steps/operations. First, input data is obtained. An assignment is then computed based on at least a portion of the obtained input data. In one embodiment, the input data is represented as a graph, wherein the graph represents client-based content request information as flow data and fees charged by server sets as cost data. One or more optimization operations are then applied to the flow data and the cost data so as to maximize flow, minimize cost, and ensure client cluster to server set assignments are within a specified network delay threshold. A reference list, e.g., map, is then computed based on results of the optimization operations. The reference list is useable for selecting, upon request, a server set to which a client device is to be directed, e.g., so as to provide computing services.

Abstract: A method and system for controlling traffic on a network. A congestion indicator is generated by network components in response to the flow of network traffic. The congestion indicator is received by a network peripheral node that has a threshold variable which controls the flow of traffic flowing from the network peripheral node. The threshold variable corresponding to the congestion indicator will be reduced in order to restrict the flow of traffic flowing from that network peripheral node. If more than one congestion indicator is received by the network peripheral node, then the threshold variable will continue to be reduced thereby further restricting network traffic. If no further congestion indicators are received, then the network peripheral node will terminate the Back-Off Period state of the threshold variable such that the threshold variable can then be increased and network traffic can increase.

Abstract: Method and Apparatus for delivering audio signals from a source node to a destination node on a network. The apparatus uses a number of switches that transmit prioritized data on a packet network. The switches are coupled to a number of send/receive nodes for sending and receiving digital audio signals on the data network. The audio packet size and the receive buffers are sized to store a minimum possible number of audio samples to minimize latency in processing audio signals arriving at said receive node, but still ensure audio delivery without interruption due to packet data network delay. An additional feature of the invention is recovery of clock synchronization over the same data network by novel arrangement of transmission of timing packets on the network. By sending a multiplicity of packets at irregular intervals a minimum network transit delay can be determined by each of the receive nodes which allows the receive nodes to filter out packet network transit delay error and maintain accurate local clocks.

Abstract: Packets in a multi-hop wireless network are routed based on the available link throughputs, network node congestion and the connectivity of the network in a manner that minimizes the use of radio resources and minimizes delay for packets in multi-hop system. The routing method also avoids congestion in the access network, especially near the network access points as provided by network access nodes. Each wireless network node maintains a link table for storing link conditions and associated route costs. Packets are routed according to the low cost route. Subsequent wireless network nodes evaluate whether a lower cost route is available and, if so, route the data packet according to the lower cost route. Every wireless network node transmits the data packet, a specified route and a time stamp indicated a time of the last data entry in the link table that was used to calculate the low cost route.

Abstract: In a wireless network, two or more devices can determine how close they are from one another in a peer-to-peer way, by exchanging the network characteristics they observe in real time A method for determining device proximity in a wireless network, includes characterizing at a first device any detected wireless network radio signals; receiving any broadcast network characteristics from at least one other device on the network; comparing the first device network characteristics with the received network characteristics from the at least one other device on the network; if the network characteristics are within a predetermined relationship, the first device and the at least one other device are in proximity with one another Network characteristics may include signal strength, noise level and MAC address of the transmitting device The method eliminates the need to calibrate WiFi signal strength as a function of location in a particular area

Abstract: The delay limited capacity of an ad hoc wireless network confined to a finite region is investigated. A transmission and relaying strategy making use of the nodes' motion to maximize the throughput is constructed. An approximate expression for the capacity as a function of the maximum allowable delay is obtained. It is found that there exists a critical value of the delay such that: (1) for values of the delay d below critical, the capacity does not benefit appreciably from the motion, (2) for moderate values of the delay d above critical, the capacity that can be achieved by taking advantage of the motion increases as d2/3, (3) the dependence of the critical delay on the number of nodes is a very slowly increasing function (n1/14). Finally, asymptotic optimality of the proposed strategy in a certain class is shown.

Abstract: We show that the current TCP Vegas algorithm can become unstable in the presence of network delay and propose a modification that stabilizes it. The stabilized Vegas remains completely source-based and can be implemented without any network support. We suggest an incremental deployment strategy for stabilized Vegas when the network contains a mix of links, some with active queue management and some without.

Abstract: The end-to-end delay (also referred to as latency) experienced by gaming users has a significant impact on the quality of online games. In this paper we concentrate on the delay introduced in access networks. This access network delay depends on the access technology used, the network load, the link rate configured on the access links (also referred to as the last mile link) and the size of the packets generated by the games. We characterize this access network delay by means of measurements. First, we focus on this delay in actually deployed access networks: dial-up, cable and Asymmetric Digital Subscriber Line (ADSL) access. In these access networks the access network delay shoots up as soon as the gaming user (or somebody else on the user's home network) saturates the user's last mile link with traffic generated by applications (e.g. web browser) other than the games. Therefore, we also characterize the access network delay in a laboratory set-up of a QoS-enabled ADSL network. In this set-up we show that it is possible to (logically) segregate the game traffic from the other traffic to such an extent that the game packets are not excessively delayed while at the same time a large part of the link capacity can be consumed by the other traffic.

Abstract: Mobile-agent systems allow applications to distribute their resource consumption across the network. By prioritizing applications and publishing the cost of actions, it is possible for applications to achieve faster performance than in an environment where resources are evenly shared. We enforce the costs of actions through markets, where user applications bid for computation from host machines. We represent applications as collections of mobile agents and introduce a distributed mechanism for allocating general computational priority to mobile agents. We derive a bidding strategy for an agent that plans expenditures given a budget, and a series of tasks to complete. We also show that a unique Nash equilibrium exists between the agents under our allocation policy. We present simulation results to show that the use of our resource-allocation mechanism and expenditure-planning algorithm results in shorter mean job completion times compared to traditional mobile-agent resource allocation. We also observe that our resource-allocation policy adapts favorably to allocate overloaded resources to higher priority agents, and that agents are able to effectively plan expenditures, even when faced with network delay and job-size estimation error.

Abstract: A “multicast code constructor” facilitates network based coding in a multicast environment by determining efficient codes for optimizing network flows, thereby increasing reliable network throughput. The network code constructor processes incoming data at each node on a byte-by-byte level to produce outgoing packets to each node in the network. Network coding is provided in which arithmetic operations can occur in any finite field with more than N-1 elements, where N represents the number of receivers in the network. Further, the complexity of arithmetic employed by the coder is independent of the network capacity, and dependent only on the number of receivers in the network. In addition, in one embodiment, multicast codes are restricted to the portion of the network obtained by a union of unicast flows from a sender node to each receiver node to produce codes which do not flood the network excessively, thereby producing a lower code design complexity.

Abstract: The poor quality of Voice over IP can be improved by adaptive playout buffering at the receiver. This technique dynamically adapts the playout deadline to network conditions, thus minimizing both late packet loss and buffering time. A standard playout buffer strategy uses an estimate (Exponentially Weighted Moving Average) of the mean and variance of network delay to set the playout deadline. This estimation is characterized by a fixed, constant weighting factor. We show that tuning of this parameter so that the strategy works very well for all network conditions is not feasible. Therefore we propose to extend this standard buffer strategy by replacing the fixed, constant weighting factor with a dynamic one. In our solution, the weighting factor is dynamically adjusted according to the observed delay variations. When these variations are high (which implies that the network conditions are changing), the parameter is set low, and vice-versa. This allows rapid adaptation to network variations and reduces the frequency of late packets (or buffering time). Simulations and experimental results show that with our strategy, the trade-off between buffering delay and late packet loss at the receiver is improved significantly.

Abstract: A communications network is provided with a self-healing topological architecture. Each node in the network detects continuously transmitted beacon packets that are propagated through the network at regular intervals by a root node. Upon failure to detect a new beacon packet after a predetermined time from a parent node, a network node determines a network isolation condition and searches for another node that is still actively connected in the network. Algorithms are provided for registering with and identifying active parent node candidates during a network failure so as to prevent the creation of network loops.

Abstract: With the rising popularity of delay-sensitive real-time multimedia applications (video, voice, data) in wireless local area networks (WLANs), it is becoming important to study the delay performance of WLANs. When the medium access control (MAC) protocol is taken into consideration, the access contention delay is a key problem. In this paper, based on a Markov model, we analyze delay performance of the IEEE 802.11 distributed coordination function (DCF). In addition, through extensive simulations, we calculate the delay performance of both basic access and RTS/CTS access mechanism of the 802.11 protocol.

Abstract: A data traffic network element for a line switched ring network. The network has a data ingress point for receiving ingress payload data having multiple working connections transported in respective timeslots, at least one of the connections being passed through the network element. The network also has a data egress point for releasing egress payload data in the ring network including the connection. The traffic network element includes a timeslot interchange entity for interchanging the connection to a timeslot in the egress payload data different from the timeslot of the connection in the ingress payload data, wherein data over the connection is transported to and from said network element in different timeslots. One advantage of the network element with the timeslot interchange is the ability to provide a better bandwidth utilization by allowing to set connections over otherwise stranded bandwidth. In a specific example of implementation, the ring network is a BLSR ring network that uses the SONET protocol for data transmission. The timeslot interchange entity has the ability to store information identifying currently implemented timeslot interchanges. This information is available for access by protection switching logic during protection switching operations.

Abstract: Data traversing packet networks experience varying delays, resulting in interarrival jitter. This can result in degraded performance in real-time multimedia communications applications if the jitter delays are large or unaccounted for in the receiver application. This paper examines modeling and simulation of network jitter delay for real-time multimedia communications applications. We examine the multistructure characteristics of network delay and develop a model for simulation of jitter. The model is confirmed empirically using collected packet network jitter delay statistics.

Abstract: The paper investigates the relationship between IP network performance criteria and voice quality for VoIP service by means of extensive experiments on a test bed network. The IP network performance is examined in terms of the utilization of the bottleneck link and the following statistical factors for VoIP packets: the average of delay variations (jitter); the standard deviation of jitter; the standard deviation of packet interarrival times; the packet loss ratio. The experiments are performed by varying the following network related parameters: bandwidth of the bottleneck link; size of the bottleneck buffer; propagation delay; the average of data sizes transmitted. Statistical analyses of the experimental results reveal a high correlation between each of the IP network performance criteria and the corresponding voice quality. Moreover, the correlation depends on the bandwidth of the bottleneck link, but appears insensitive to the other network related parameters. These outcomes suggest that it may be possible to manage effectively the quality of service (QoS) of VoIP by monitoring the corresponding network performance criteria.

Abstract: An overlay network is used to logically represent an underlying physical network. A network condition is monitored by a node in a target region of the network. If the network condition occurs, the node transmits a notification to a source node including location information for nodes physically close in the physical network. The source node may select a routing node in the target region based on the location information.

Abstract: In this paper we investigated the problem of remote stabilization via communication networks. This problem arises when the control law is remotely implemented. The exchange of data between the controller and the system is done via a data communication network with known dynamics. This leads to the problem of stabilizing an open-loop unstable system with timevarying delay. Assuming a known model for the timedelay dynamic, we develop a time-varying horizon predictor, which is used as a basis to build the stabilizing control law. Simulation results are also presented.

Abstract: As the Internet is a best-effort delivery network, audio packets may be delayed or lost en route to the receiver due to network congestion. To compensate for the variation in network delay, audio applications buffer received packets before playing them out. Basic algorithms adjust the packet play out time during periods of silence such that all packets within a talkspurt are equally delayed. Another approach is to scale individual voice packets using dynamic time-scale modification. In this work, an adaptive playout algorithm based on the normalized least mean square algorithm, is improved by introducing a spike-detection mode to rapidly adjust to delay spikes. Simulations on Internet traces show that the enhanced bi-modal playout algorithm improves performance by reducing both the average delay and the loss rate as compared to the original algorithm.

Abstract: We consider the throughput/delay tradeoffs for scheduling data transmissions in a mobile ad-hoc net- work. To reduce delays in the network, each user sends redundant packet information along multiple paths to the destination. Such redundancy improves delay at the cost of increasing network congestion. Assuming the network has a cell partitioned structure and users move according to a simplified iid mobility model, we compute the exact net- work capacity and delay when no redundancy is used. The capacity achieving algorithm is a modified version of the Grossglauser-Tse 2-hop relay algorithm and provides O(N) delay (where N is the number of users). We then show that redundancy cannot increase capacity, but can signifi- cantly improve delay. A lower bound on delay of O( p N) is computed for any algorithm (with or without redundancy) which restricts packets to 2-hop paths. A scheduling pro- tocol which uses redundancy is presented and shown to achieve this delay bound when data rates of all sessions are reduced to O(1/ p N).

Abstract: A system, method, apparatus, means, and computer program code for providing a delay guarantee for a wireless network is provided.

Abstract: In this paper, we revisit a general class of multicriteria multi-constrained network design problems and attempt to solve, in a novel way, with Evolutionary Algorithms (EAs). A major challenge to solving such problems is to capture possibly all the (representative) equivalent and diverse solutions. In this work, we formulate, without loss of generality, a bi-criteria bi- constrained communication network topological design problem. Two of the primary objectives to be optimized are network delay and cost subject to satisfaction of reliability and flowconstraints. This is a NP-hard problem so we use a hybrid approach (for initialization of the population) along with EA. Furthermore, the twoobjective optimal solution front is not known a priori. Therefore, we use a multiobjective EA which produces diverse solution space and monitors convergence; the EA has been demonstrated to work effectively across complex problems of unknown nature. We tested this approach for designing networks of different sizes and found that the approach scales well with larger networks. Results thus obtained are compared with those obtained by two traditional approaches namely, the exhaustive search and branch exchange heuristics.

Abstract: In this paper we consider the problem of collecting a large amount of data from several different hosts to a single destination in a wide-area network. Often, due to congestion conditions, the paths chosen by the network may have poor throughput. By choosing an alternate route at the application level, we may be able to obtain substantially faster completion time. This data collection problem is a nontrivial one because the issue is not only to avoid congested link(s), but to devise a coordinated transfer schedule which would afford maximum possible utilization of available network resources. In this paper we present an approach for computing coordinated data collection schedules, which can result in significant performance improvements. We make no assumptions about knowledge of the topology of the network or the capacity available on individual links of the network, i.e., we only use end-to-end information. Finally, we also study the shortcomings of this approach in terms of the gap between the theoretical formulation and the resulting data transfers in wide-area networks. In general, our approach can be used for solving arbitrary data movement problems over the Internet. We use the Bistro platform to illustrate one application of our techniques.

Abstract: Sensor networks have many potential applications in biology, physics, medicine, and the military. One major challenge is to maximize network life under the constraint of an extremely limited power supply. This is especially true for biomedical applications, which require large numbers of nodes that may be implanted in a subject; frequent battery changes are impractical. The paper proposes a new data gathering protocol, and analyzes a potential application in biomedical sensing technology. The protocol, named hybrid indirect transmission (HIT), is based on an architecture consisting of one or more clusters that cooperatively compute multiple, multi-hop, indirect transmission routes. In order to minimize both energy consumption and network delay, parallel transmissions with a collision avoidance guarantee are used throughout the network; adjacent clusters do not prevent this mechanism from working. To accomplish this, each sensor independently computes a medium access controlling TDMA schedule. HIT was simulated along with three existing protocols; a comparison of their performance in terms of energy efficiency, delay, and network lifetime is provided. Results show that HIT greatly reduces both energy consumption and network delay; it also maintains longer network life compared to the other three protocols. The proposed protocol is promising and would contribute to the use of wireless micro sensor networks in future biomedical technologies.