In this paper, a parallel flow monitoring technique that achieves accurate measurement of end-to-end delay of networks is proposed. In network monitoring tasks, network researchers and practitioners usually monitor multiple probe flows to measure delays on multiple paths in parallel. However, when they measure an end-to-end delay on a path, information of flows except for the flow along the path is not utilized in the conventional method. Generally, paths of flows share common parts in parallel monitoring. In the proposed method, information of flows on paths that share common parts, utilizes to measure delay on a path by partially converting the observation results of a flow to those of another flow. We perform simulations to confirm that the observation results of 72 parallel flows of active measurement are appropriately converted between each other. When the 99th-percentile of the end-to-end delay for each flow are measured, the accuracy of the proposed method is doubled compared with the conventional method.

String-topology multi-hop network is often selected as an analysis object because it is one of the fundamental network topologies. The purpose of this paper is to establish expression for end-to-end delay for IEEE 802.11 string-topology multi-hop networks. For obtaining the analytical expression, the effects of frame collisions and carrier-sensing effect from other nodes under the non-saturated condition are obtained for each node in the network. For expressing the properties in non-saturated condition, a new parameter, which is frame-existence probability, is defined. The end-to-end delay of a string-topology multi-hop network can be derived as the sum of the transmission delays in the network flow. The analytical predictions agree with simulation results well, which show validity of the obtained analytical expressions.

We investigate delay analysis of multi-hop wireless mesh network (WMN) where nodes have multi-channel and multiple transceivers to increase the network capacity. The functionality of the multi-channel and multiple transceivers allows the whole WMN to be decomposed into disjoint zones in such a way that i) nodes in a zone are within one-hop distance, and relay node and end nodes with different CWmins contend to access the channel based on IEEE 802.11e EDCA, ii) different channels are assigned to neighbor zones to prevent the hidden node problem, iii) relay nodes can transmit and receive the packets simultaneously by multi-channel and multiple transceivers. With this decomposition of the network, we focus on the delay at a single zone and then the end-to-end delay can be obtained as the sum of zone-delays. In order to have the location-independent end-to-end delay to the gateway regardless of source nodes' locations, we propose two packet management schemes, called the differentiated CW policy and the strict priority policy, at each relay node where relay packets with longer hop count are buffered in higher priority queues according to their experienced hop count. For the differentiated CW policy, a relay node adopts the functionality of IEEE 802.11e EDCA where a higher priority queue has a shorter minimum contention window. We model a typical zone as a one-hop IEEE 802.11e EDCA network under non-saturation condition where priority queues have different packet arrival rates and different minimum contention window sizes. First, we find the PGF (probability generating function) of the HoL-delay of packets at priority queues in a zone. Second, by modeling each queue as M/G/1 queue with the HoL-delay as a service time, we obtain the packet delay (the sum of the queueing delay and the HoL-delay) of each priority queue in a zone. Third, the average end-to-end delay of packet generated at end node in each zone is obtained by summing up the packet delays at each zone. For the strict priority policy, we regard a relay node as a single queueing system with multiple priority queues where relay packets in priority queues are served in the order of strict priority. Relay node has smaller CWmin than end node has and relay node competes with end nodes in a zone. Using the PGF of HoL-delay of packet at relay node and end node, we obtain the packet delay in a zone. The average end-to-end delay to the gateway generated at end node in each zone is obtained. Finally, for both the differentiated CW policy and strict priority policy, by equating all end-to-end delays to be approximately equal, we find the minimum contention window sizes of each priority queue numerically by trial and error method so that end-to-end delays of packets are almost equal regardless of their source's location, respectively. Numerical results show that proposed two methods obtain almost same end-to-end delay of packets regardless of their generated locations and our analytical results are shown to be well matched with the simulation results.

Recent research on overlay networks has revealed that user-perceived network performance could be improved by an overlay routing mechanism. The effectiveness of overlay routing is mainly a result of the policy mismatch between the overlay routing and the underlay IP routing operated by ISPs. However, this policy mismatch causes a "free-riding" traffic problem, which may become harmful to the cost structure of Internet Service Providers. In the present paper, we define the free-riding problem in the overlay routing and evaluate the degree of free-riding traffic to reveal the effect of the problem on ISPs. We introduce a numerical metric to evaluate the degree of the free-riding problem and confirm that most multihop overlay paths that have better performance than the direct path brings the free-riding problem. We also discuss the guidelines for selecting paths that are more effective than the direct path and that mitigate the free-riding problem.

Recent research on overlay networks has revealed that user-perceived network performance, such as end-to-end delay performance, could be improved by an overlay routing mechanism. However, these studies consider only end-to-end delay, and few studies have focused on bandwidth-related information, such as available bandwidth and TCP throughput, which are important performance metrics especially for long-lived data transmission. In the present paper, we investigate the effect of overlay routing both delay and bandwidth-related information, based on the measurement results of network paths between PlanetLab nodes. We consider three metrics for selecting the overlay route: end-to-end delay, available bandwidth, and TCP throughput. We then show that the available bandwidth-based overlay routing provides significant gain, as compared with delay-based routing. We further reveal the correlation between the latency and available bandwidth of the overlay paths and propose several guidelines for selecting an overlay path.

Wireless mesh networks are attracting more and more attention as a promising technology for the next generation access infrastructure. QoS support is a unavoidable task given the rising popularity of multimedia applications, and also a challenging task for multi-hop wireless mesh networks. Among the numerous QoS factors, end-to-end delay is one of the most critical and important issues, especially for the real time applications. Over multi-hop wireless mesh networks, end-to-end delay of a flow is highly dependant on the number of hops as well as congestion condition of the hop nodes that the flow traverses through. In this paper, we propose QoS priority control schemes based on the end-to-end QoS delay metrics in order to increase traffic accommodation, i.e., the numbers of real-time flows which satisfy the requirements of end-to-end delay and packet delivery ratio over multi-hop wireless mesh networks. The first scheme enables source and forwarding nodes to perform priority control based on the number of hops of routes. The second scheme enables nodes to perform priority control based on the congestion condition of the hop nodes, where the flow traverses through. The effectiveness of the proposed schemes is investigated with NS-2 network simulator for voice and video traffics over multi-hop wireless mesh networks. Simulation results show that the scheme greatly improves the traffic accommodation for voice and video applications in multi-hop wireless mesh networks.

In the emerging networks, routing may be performed at various levels of the TCP/IP stack, such as datagram, TCP stream or application level, with possibly different message forwarding modes. We formulate an abstract quickest path problem for the transmission of a message of size σ from a source to a destination with the minimum end-to-end delay over a network with bandwidth and delay constraints on the links. We consider six modes for the message forwarding at the nodes reflecting the mechanisms such as circuit switching, store and forward, and their combinations. For each of first five modes, we present O( m2 + mn log n ) time algorithms to compute the quickest path for a given message size σ. For the last mode, the quickest path can be computed in O(m + n log n ) time.

End-to-end delay and loss measurement is an efficient way for a host to examine the network performance. Unnoticed clock errors that influence the accuracy of the timestamp may result in fatal system errors. In this paper, we discuss the characteristics and defects of the existing clock distortion adjustment algorithms. Those algorithms are not applicable to process a long-term delay trace, which contains periodical NTP clock adjustment. Therefore, we propose a relatively robust algorithm to resolve the problem. The algorithm employs window function to partition the long-term trace into short segments, improves the precision of the estimation of the time and amount of NTP clock adjustment To evaluate the performance of our proposed algorithm, we practice it in adjusting the clock distortion of the real delay traces collected from Internet. The results indicate that our proposed algorithm has excellent effect on the removal of the clock distortion from the long-term delay traces.

In this paper, we propose an adaptive video frame rate control method, called AFCON, that video encoders use in conjunction with explicit rate based congestion control in the network. First, an encoder buffer constraint which guarantees the end-to-end delay of video frames is derived under the assumption of bounded network transmission delay for every frame data. AFCON is based on the constraint and consists of future channel rate prediction, frame discarding, and frame skipping. Recursive Least-Squares (RLS) is used to predict the low-frequency component of the channel rate. Frame discarding prevents the delay violation of frames due to the prediction error of the channel rate. Frame skipping adapts the encoder output rate to the channel rate while avoiding abrupt quality degradation during the congestion period. From the simulation results, it is shown that AFCON can adapt to the time-varying rate channel with less degradation in temporal resolution and in PSNR performance compared to the conventional approach.

From our previous studies, we derived the worst case cell delay within an ATM switch and thus can find the worst case end-to-end delay for a set of real-time connections. We observed that these delays are sensitive to the priority assignment of the connections. With a better priority assignment scheme within the switch, the worst case delay can be reduced and provide a better network performance. We extend our previous work on the closed form analysis to conduct more experimental study of how different priority assignments and system parameters may affect the performance. Furthermore, from our worst case delay analysis on a regulated ATM switch, network traffic can be smoothed by a leaky bucket at the output controller for each connection. With the appropriate setting on the leaky bucket parameter, the burstiness of the network traffic can be reduced without increasing the delay in the switch. Therefore, fewer buffers will be required for each active connection within the switch. In this paper, our experimental results have shown that the buffer requirement can be reduced up to 5.75% for each connection, which could be significant, when hundreds of connections are passing through the switches within a regulated ATM network.

To implement the PGPS packet scheduling algorithm in high speed networks is more difficult since it is based on real time simulation of an equivalent fluid-model system leading to a higher implementation time complexity. A modified approach to PGPS is the SCFQ scheme. This scheme is easy to implement, but has an increasing end-to-end delay bound. The VC packet scheduling algorithm provides the same end-to-end delay bound as PGPS does, but has the disadvantage of unfairness. As SCFQ, SFQ is much easier to implement than PGPS and achieves the same fairness, but has a higher end-to-end delay bound than PGPS. We propose a new packet scheduling algorithm, called Minimum Starting-tag Fair Queueing (MSFQ), which assigns the virtual time to be the minimum starting tag over all backlogged connections. MSFQ is much easier to implement than PGPS and provides the same end-to-end delay bound for each connection and fairness as PGPS. In this paper, we will show the end-to-end delay bound and fairness of MSFQ and compare 5 rate-based packet scheduling algorithms including PGPS, VC, SCFQ, SFQ, and MSFQ focusing on end-to-end delay bound, fairness, and implementation time complexity.