This paper studies how to extract and group essential nodes to guarantee loop freedom during the update of the network, while minimizing update time. We propose a model to establish the relationships of loops which may arise during the updating procedure. Given the model, we design a heuristic algorithm, with the recursive optimizing for the relationships, to extract essential nodes for the loop-free update, and shorten the updating time. The performance of our proposal has been examined and we verified that our algorithm would obviously reduce the update time.

By separating the control and forwarding planes, Software-Defined networking (SDN) enables the forwarding paths to be flexibly controlled by the logically centralized controllers using the global network view. To introduce SDN into existing networks, it is necessary to upgrade traditional devices to SDN- enabled ones. However, due to the business, economic and management limitations, it is difficult to realize full SDN deployment. As a result, how to migrate existing devices to SDN-compliant ones becomes the obvious dilemma for every network operator. In this paper, we address this question from the network performance perspective, and study how to leverage the capability of SDN to maximize traffic flow that can be achieved in hybrid SDNs. We formulate the maximum flow problem in networks with partial SDN deployment, and develop a fast Fully Polynomial Time Approximation Scheme (FPTAS) for solving it. Simulation results using real topologies show that hybrid SDNs outperform traditional networks, and we can obtain a near optimal network performance when 50% of SDN nodes are deployed.

Interactive multiview video streaming (IMVS) allows viewers to periodically switch viewpoint. Its user experience can be further enhanced by creating virtual views from neighboring coded views using view synthesis techniques. Dynamic adaptive streaming over HTTP (DASH) is a new standard that can adjust the quality of video streaming according to the network condition. In this paper, we propose an improved DASH-based IMVS scheme over wireless networks. The main contributions are twofold. First, our scheme allows virtual views to be generated at either the cloud-based server or the client, and can adaptively select the optimal approach based on the network condition and the cost of the cloud. Second, scalable video coding is used in our system. Simulations with the NS3 tool demonstrate the advantage of our proposed scheme over the existing approach with client-based view synthesis and single-layer video coding.

Recently, Dynamic Adaptive Streaming over HTTP (DASH) has attracted significant attention. In this paper, we consider DASH-based transmission of scalable videos in wireless broadband access networks (e.g., long-term evolution and WiMAX), and propose three methods to enhance the quality of experience of wireless DASH users. First, we design an improved mapping scheme from scalable video coding layers to DASH layers that can provide the desired bitrates, enhance the video end-to-end throughput, and reduce the HTTP communication overhead. Second, we develop a DASH-friendly scheduling and resource allocation algorithm by integrating the DASH-based media delivery and the radio-level adaptation via a cross-layer approach. It utilizes the characteristics of video content and scalable video coding, and greatly reduces the possibility of video playback interruption by considering the client buffer status. The optimization problem is formulated as a mixed binary integer programming problem, and is solved by a subgradient method. Finally, a DASH proxy-based bitrate stabilization algorithm is proposed to improve the video playback smoothness that can achieve the desired tradeoff between playback quality and stability. Simulations with the Qualnet tool demonstrate that our schemes achieve better performances than other methods in the literature.

Software Defined Networking (SDN) is an emerging networking paradigm that assumes a logically centralized control plane separated from the data plane. Despite all its advantages, separating the control and data planes introduces new challenges regarding resilient communications between the two. That is, disconnections between switches and their controllers could result in substantial packet loss and performance degradation. To achieve resilient control traffic forwarding, this paper investigates the protection of control traffic in SDNs with multiple controllers. We propose a control traffic protection scheme that combines both local rerouting and constrained reverse forwarding protections. This scheme enables switches to locally react to failures and redirect the control traffic to controllers by using standby backup forwarding options. Our goal is then to find a set of primary routes for control traffic, called protection control network, where as much control traffic as possible can benefit from the proposed protection scheme. We formulate the protection control network problem and develop an algorithm to solve it. Simulation results on real topologies show that our approach significantly improves the resilience of control traffic.

Recently dynamic adaptive streaming over HTTP (DASH) has gained significant attentions. In this paper, we study DASH-based transmission of scalable videos in wireless broadband access networks (e.g. LTE, WiMAX), and propose a DASH-friendly scheduling and resource allocation scheme (DFSRA) to enhance the Quality-of-Experience (QoE) of wireless DASH users. By integrating DASH-based media delivery and radio-level adaptation in a cross-layer manner, the scheme can determine the optimal wireless resource allocation. A gradient-based algorithm is proposed to solve the optimization problem, which has the following advantages over traditional algorithms: (1) The characteristics of video content and scalable video coding are utilized. (2) The client buffer status is considered to reduce video playback interruption and buffer overflow. Simulations with the Qualnet tool demonstrate that our scheme achieves better performance than existing methods in the literature.

In cloud data centers, different mapping relationships between virtual machines (VMs) and physical machines (PMs) cause different resource utilization, therefore, how to place VMs on PMs to improve resource utilization and reduce energy consumption is one of the major concerns for cloud providers. The existing VM placement schemes are to optimize physical server resources utilization or network resources utilization, but few of them focuses on optimizing multiple resources utilization simultaneously. To address the issue, this paper proposes a VM placement scheme meeting multiple resource constraints, such as the physical server size (CPU, memory, storage, bandwidth, etc.) and network link capacity to improve resource utilization and reduce both the number of active physical servers and network elements so as to finally reduce energy consumption. Since VM placement problem is abstracted as a combination of bin packing problem and quadratic assignment problem, which is also known as a classic combinatorial optimization and NP-hard problem, we design a novel greedy algorithm by combining minimum cut with the best-fit, and the simulations show that our solution achieves better results.

In order to provide low latency, pay-as-you-go infrastructure, Cloud service provider has built a large number of geo-distributed data centers. Nowadays, energy cost has become increasingly important fraction of data center operating cost. Researchers' attentions are attracted to propose many methods to reduce the electricity cost for data centers. However, another significant operating cost, Internet bandwidth cost, has been neglected. Because both the electricity prices and bandwidth prices are varied with time and regions, it is not easy to minimize the operating cost while response to workload timely. As a solution, based on Lyapunov optimization framework, we propose an online control policy to achieve close-to-optimal performance with tradeoff between cost and delay. The advantage of our solution is that we do not need any future knowledge on the stochastic model (e.g., bandwidth prices and electricity prices). The results from our simulations on real workload trace and price data demonstrated that our solution is effective.

Cloud infrastructure providers (CIPs) usually build a large number of geo-distributed data centers to cater to the recent proliferation of Cloud Computing. The CIPs commonly use multiple internet service providers (ISPs) to interconnect their geo-distributed data centers. Diurnal patterns and leftover bandwidth were effectively exploited by past studies to improve bandwidth utilization and minimize cost on inter-data center traffic. However, the growing number of inter-domain traffic was neglected. Inter-domain communications may become the potential bottleneck of inter-data center bulk transfers. Moreover, the rising inter-domain traffic increases the ISP operational cost, which will be not beneficial to reduce the CIPs' bandwidth cost or improve the quality of services over a long run. In this paper, we present a scheduling scheme that considers both bandwidth utilization and ISPs friendliness via a store-and-forward mechanism. The problem is modeled on a time-expanded graph. We compared our scheme with general bulk transfers mechanism under several different simulation settings. The results demonstrate that our strategy reduces the inter-domain traffic tremendously and achieves the ISP-friendliness significantly.

Live migration of virtual machine plays an important role in data center, which can successfully migrate virtual machine from one physical machine to another with only slight influence on upper workload. It can be used to facilitate hardware maintenance, load balancing, fault-tolerance and power-saving, especially in cloud computing data centers. Although the pre-copy is the prevailing approach, it cannot distinguish which memory page is used, resulting in transferring large amounts of useless memory pages. This paper presents a novel approach Memory Exploration and Encoding (ME2), which first identifies useful pages and then utilizes Run Length Encode algorithm to quickly encode memory, to efficiently decrease the total transferred data, total migration time and downtime. Experiments demonstrate that ME2 can significantly decrease 50.5% of total transferred data, 48.2% of total time and 47.6% of downtime on average compared with Xen's pre-copy algorithm.

Online video service has become one of the main Internet applications, especially video-on-demand (VoD). Most VoD systems are based on P2P technology currently. As the number of users and media files grows rapidly on demand, the storage capacity of source media files becomes limited. Besides, the system performance and user experience drops obviously when online users do not share the same data. In this paper, we present a reliable and high-performance distributed storage system for P2P-VoD service based on Hadoop Distribute File System (HDFS). The reliability and scalability of the system are proved through experiments. The results also verify that the data response time and server utilizations get improved, which means the system gets a better performance.

The key infrastructure of Cloud Computing is data center which is shared by many tenants. Each tenant's application competes for acquiring more network bandwidth in order to maximize its utility. However, this may cause interference among these diverse applications. Malicious competition not only degrades its performance, but also makes the overall performance of the data center poor and ineffective. To ensure the Quality of Services (QoS) and achieve high network utilization, in this paper, we propose a bandwidth allocation scheme for data center networks (DCNs), which is based on an application utility-based model. In our scheme, multi-path feature of DCN is leveraged to improve the network utilization, and utility functions are constructed to differentiate the throughput and delay sensibilities of different applications. Moreover, our scheme is suitable for arbitrary DCN topologies and without modification on current hardware. The numerical simulation shows that our scheme can provide bandwidth guarantee, fine-grained service differentiation and achieve high network utilization.

In cognitive radio networks, when the primary user (PU) who has a minimum transmission rate requirement is experiencing a bad channel condition, it can resort to the cooperation with the secondary users (SUs) in exchange for a fraction of its licensed spectrum. However, due to the selfish nature, the SUs may not act as cooperatively as to meet the PU's Quality of Service (QoS) requirement. On the other hand, the SUs may not exploit efficiently the benefit from cooperation if they compete with each other and collaborate with the PU independently. Therefore, how to stimulate the SUs to form a group to cooperate with the PU while guarantee the PU's QoS requirement are critical challenges. In this paper, we propose a two-level bargain framework to address the aforementioned problems. In the proposed framework, the interaction between the PU and the SUs are modeled as the first level bargain game while the second level bargain game is used to formulate the SUs' decision making process on spectrum sharing. We analyze the optimal actions of the PU and the SU and derive the theoretic results for the one-PU one-SU scenario. To find the solutions for the one-PU multi-SU scenario, we propose a revised numerical searching algorithm and prove its convergence. Finally, we demonstrate the effectiveness and efficiency of the proposed scheme through simulations.

End-to-end delay is an important QoS metric and has received much research attention. Recently, the multifractal detrended fluctuation analysis (MFDFA) is widely used as a robust tool to investigate the scale behavior of non-stationary time series. In this paper, we use MFDFA to analyze the scale behavior of end-to-end delay series. Based on ping series of both home and international paths, we find the 2-order correlation of delay series is analysis scale, path, and (collection) time dependent: may be biscaling or even multiscaling, persistent or anti-persistent. In addition, we observe that delay series may be multifractal at both fine and coarse scales. We also examine the source of multifractality for delay series and find the source is much complex.

Peer-to-Peer (P2P) file sharing applications use multiple TCP connections between peers to transfer data. The aggressiveness and robustness of P2P technology remarkably improve transfer efficiency and network bandwidth utilization. However, while the network bottleneck link is congested, P2P applications tend to unfairly steal bandwidth from other traditional Internet applications (Client/Server mode), which deteriorates the performance of traditional Internet applications. The paper proposes a friendlyP2P system with new application-level approaches for congestion detection and avoidance to keep fairness between P2P traffic and traditional Internet traffic. friendlyP2P, which is friendly to ISPs, namely to Internet networks and traditional Internet traffic, detects network congestion via throughput measurements and alleviates network congestion by optimization of the number of P2P connections from the viewpoint of P2P users. friendlyP2P system requires neither network node support nor TCP modification, which makes it easy to deploy. Simulation experiments demonstrate that fairness and congestion avoidance can be achieved in presence of congestion, and network bandwidth can be effectively utilized in absence of congestion with friendlyP2P technology.

Deploying multiple mobile sinks is an attractive approach to enhance the performance of wireless sensor networks. In this paper, we address the scenario of two mobile sinks. The two sinks can travel in the same region or in the divided regions separately. The problems are formulated by a lattice-based network model. We deduce the network lifetime and delay of data delivery in different mobility patterns. The relation between network performance and number of mobile sinks is also discussed. Simulation results are provided to validate our theoretic analysis followed by performance comparison of different mobility patterns.

Managing trust is a key issue for a wide acceptance of P2P computing, particularly in critical areas such as e-commerce. Reputation based trust management has been identified in the literature as a viable solution to the problem. However, the mechanism faces the challenges of subjectively, experiential weighting referrals when aggregating recommendation information. Furthermore, not considering some malicious attacks when building trust relationship between peers in the existing schemes make trust model very vulnerable. This paper presents P2PTrust-a new trust framework based on reputation for unstructured P2P networks. In P2PTrust, except reputation value, reputation revision value is also taken into account in order to deal with the dynamic or spoiling behavior of peers, which makes P2PTrust differ from other trust models based on the reputation only. Considering the credibility of peers' referrals, a credibility quantification and update scheme is proposed in the paper as reliable means of seeking honest feedbacks. Subsequent experimental results show that, compared to the existing trust models, our scheme is robust in systems where the vast majority of users are malicious and more advanced in successful transaction rate.

Video streaming over wireless networks is becoming increasingly important. In this paper, we study the problem of multiple real-time video surveillance streaming over time-varying, error-prone wireless mesh networks (WMNs). The video streams are transmitted on the Internet through Internet Gateway (IGW). For better transmission, we propose a video streaming design by multiple flows gathering with opportunistic relay (OR) and network coding (NC): FORNEC. With meticulous design, FORNEC avoids multiple video streams aggregating around the most congested region of IGW. Further, with NC and OR, FORNEC provides robust packets forwarding with minimum loss rate to overall video streams.

Prolonging the network lifetime and reducing the delay of event delivery are both important in event-driven multi- hop wireless sensor networks. In this paper, an optimum predefined mobility trajectory of the sink was explored to balance the two performance metrics. We focus on a lattice-based analytical model for understanding performance as system and mobility parameters are scaled. Based on the model, the theoretical analysis and the simulation results validate our statements. The case of multiple mobile sinks are also discussed and simulated.

In peer-to-peer file sharing networks, files can be regarded as public goods. Based on the economic game theory, we firstly analyze the performance of file sharing in P2P networks in this paper, and demonstrate that being free-rider is the best strategy for most users without external incentives. Furthermore, another factor is considered in our research that each user has its own willingness to share the file resources (willing-to-share) which is his private information unobserved by other users. Based on these premises, we design an incentive mechanism to induce all users to share the file as many as possible according to his true willingness. In this mechanism, users are inspired to provide commons by taxing them. The correctness and efficiency of this mechanism have been proved by mathematic. At the end of this work, a logical P2P management system has been proposed to make this mechanism running

In IEEE 802.16e, the energy saving mechanism is an important problem. Excessive listening operations in sleep-mode will waste a lot of energy and shorten the lifetime of a Mobile Subscriber Station (MSS). In this paper, we propose an enhanced energy saving mechanism to overcome this problem. An embedded Markov chain model is adopted to analyze the enhanced energy saving mechanism analytically. Meanwhile, a closed-form expression of the average energy consumption in the sleep-mode for the suggested mechanism is presented. We evaluate and validate the suggested scheme via analytical results and simulation results. Simulation results illustrate that the proposed mechanism can obtain better effects of energy conservation to minimize MSS power usage and to extend the lifetime of MSS effectively while not compromise the performance of Service Data Unit (SDU) response time.

A modeling approach is proposed in this paper to build the bipartite fault propagation model (FPM) for Internet services. The FPM is layered as Internet services involve multiple layers. Two fault localization algorithms, MCA (Max-Covering Algorithm) and MCA+, are designed for the bipartite FPM. MCA+ is an extension of MCA, taking lost and spurious symptoms into account. Simulation results show that MCA+ achieves high detection rate, low false positive rate and has polynomial computational complexity even in the presence of lost and spurious symptoms.

As a well-known multimedia stream transport protocol, TFRC provides smooth transfer rate under stable network conditions, and achieves good fairness to TCP. However, it is not flexible in heterogeneous mobile networks because the available bandwidth varies rapidly. This paper proposes a measurement based TFRC (MBTFRC) protocol, which uses passive bandwidth measurements at the receiver to improve the flexibility of TFRC. In addition, a window-based EWMA filter with two weights is used to achieve stability and fairness simultaneously. Simulation results verify the flexibility, stability and fairness of MBTFRC.

The problem of topology control in multi-rate multihop ad hoc networks is studied and a multi-rate aware topology control algorithm (MATC) is proposed in the paper. MATC fully takes the inherent multi-rate property of wireless networks into account. It removes some low rate links while preserving the network connectivity. On the basis of MATC, on-demand routing protocols can find the optimal route in the one-hop range. Extensive simulations show that MATC can significantly improve the network throughput.

One-way delay (OWD) traces are important measurements for analyzing end-to-end performance on the Internet. It is still a great challenge to provide a scalable solution for large-scale OWD measurement. Because the clocks at end systems are usually not synchronized, the OWD measurements are often inaccurate. For the more challenging case with clock resets to some reference times during the measurement, the OWD measurements are more inaccurate. Furthermore, the measurement data often exhibit considerable network-induced noise when the network is heavily loaded. All the existing OWD measurement techniques, such as linear programming and convex-hull approach (CHA), try to solve this problem by deterministic mathematics model. However, they often fail to distinguish clock resets from temporary Internet congestion. Based on the fuzzy-clustering analysis, this paper proposes a new algorithm to estimate and remove the clock skews and resets from measurement results. This algorithm has been implemented as a tool called fuzzy-based OWD corrector (FOC). The paper then presents OWD measurements of several Internet paths using FOC. Numerical experiments demonstrate that FOC is more accurate and robust than the existing techniques. FOCs computation complexity O(N) is similar to that of CHA and its computing time is much less than that of convex-hull technique.

Packet one-way delay (OWD) traces are important measurements for analyzing end-to-end performance in the Internet. Due to the fact that the clocks at the end systems are usually not synchronized, the OWD measurements can be quite inaccurate. For the more challenging case with clock resets, i.e., the clocks are reset to some reference times during the measurement period, the OWD measurements can be even worse. We propose a new algorithm to estimate and remove the relative clock skews and clock resets from measurements, based on fuzzy-clustering analysis. Numerical experiments shows that, compared with existing techniques such as linear programming and the convex-hull approach, the new algorithm is more accurate and robust.

Future communication systems might be characterized by an integration of multiple access systems, and people are expecting excellent multi-access solutions for better services. However, currently proposed solutions in different layers are unable to completely fulfill the requirements. In this paper, we put forward the stream control transport protocol (SCTP) as a transport layer approach for multi-access, and propose an efficient SCTP load-sharing enhancement. The evaluation of performance has been investigated in the key multi-access scenarios with theoretical analyses, simulations in NS and some Linux-kernel experiments. Since SCTP supplies an effective multi-homing mechanism and the transport layer solutions for multi-access can quickly adapt to the fluctuating network conditions, our SCTP approach is able to bring more throughput for the simultaneous multi-access perform seamless behavior during the vertical handover between heterogeneous access networks, and provide better robustness and connectivity than other solutions in the wireless multi-access scenarios.

Congestion control for large multicast groups has been a challenging issue for widespread deployment of multicast services. Single-rate multicast congestion control schemes address both TCP-friendly and scalability issues with low complexity. However, they do not solve the well-known "drop-to-zero" problem, i.e., to prevent a slow receiver from slowing down faster receivers in the same multicast group. For this purpose, we propose a new single-rate multicast congestion control mechanism for heterogeneous receivers which helps each multicast receiver to receive at its desired rate. The basic idea is as follows. Each multicast receiver maintains and adjusts a window at itself just like what TCP source does. They calculate a desirable receive rate with this window and the RTT to the source and feed back this rate to the source and intermediate routers. The sender tunes its rate according to the fastest receiver in the group. Each multicast router does not copy each incoming multicast packet to all egress links in the multicast tree unconditionally but copy packet to egress links according to the rate information obtained from the downstream receivers. 'Good' egress links receive more packets while 'bad' egress links receive less packets. The proposed strategy is shown to be still TCP-friendly and scalable while eliminating the "drop-to-zero" problem.

Design the multimedia stream transport protocol in heterogeneous mobile environments faces great challenges due to two objectives: smooth transfer rate, i.e., stability objective, and fast response to the changes of available bandwidth due to vertical handoff in heterogeneous networks, i.e., flexibility objective. TFRC is an equation-based congestion control scheme, which fits the stability objective well, but fails in the flexibility objective. This paper proposes to use passive bandwidth measurement at the receiver to improve the performance of TFRC. In addition, a window based EWMA filter with two weights is used to achieve stability and flexibility at the same time. Based on these considerations, a TFRC enhancement (MBTFRC) is presented. Its stability, flexibility and fairness are verified by simulations.

The paper presents a more complete analytical model of multiple TCP connections established between a busy server and multiple receivers. In such a case, the server may become the bottleneck of the whole system, and TCP behaviors are different from the model presented before. Based on the analysis of working flows in the system and a M/G/1 queueing model, the RTT and long-term TCP throughput formulae are derived in terms of number of TCP connections, packet loss rate, and end-to-end delay. The effect of maximum window size is also investigated. Finally, simulation results confirm the effectiveness of the analytical model.

Numerous researches have shown that the original DCF access mode of IEEE 802.11 has low performance in case of heavy traffic environment. Based on the analysis of previous IEEE 802.11 DCF, we have proposed a parameter that closely relative to the traffic load of the current wireless LAN and formed a new DCF-based protocol with the name of NSAD (new self-adapt DCF-based protocol). NSAD is proved effective in the error-free environment. In this paper, we proposed some new mechanisms to further enhance the performance of NSAD and formed a new MAC mechanism RSAD (robust self-adapt DCF-based protocol). We have done simulation in a more real environment than we do in NSAD. The simulation results have shown that the proposed RSAD can achieve better performance when compared with the original NSAD algorithm in an error-prone environment (e.g. better property in goodput and window tuning accuracy).

Accurate measurement of network bandwidth is crucial for many flexible Internet applications and protocols. Based on a new multi-packet delay model, this paper presents a new algorithm called ALBP (asymmetric link bandwidth probing) to measure each hop's link bandwidth in both directions in an IP network, and uses the kernel density estimator to produce a final estimate. Compared with previous techniques, ALBP can eliminate (at least partially) the errors introduced by ICMP replies through the use of interarrival time of two probing packets, and avoid the problem of error accumulation. Simulation results show that ALBP is usually more accurate than both one-packet and packet tailgating techniques, and the bandwidth consumption of ALBP is close to that of packet tailgating.

Several handover schemes, such as smooth handover and fast handover, have been proposed to realize seamless handovers in mobile environment. However in our simulation, even if smooth handover in mobile IPv6 are implemented, TCP can not always be ensured to achieve better performance due to packets forwarding burst. Based on the performance analysis of TCP with the support of smooth handover, this paper proposes an enhanced buffer forwarding control scheme in smooth handover to improve TCP performance. In our scheme, a packet-pair probing technology is adopted to estimate the bandwidth of the new path from previous router (Prtr) to mobile node (MN), which is used by Prtr to control the buffered packets forwarding. Finally, simulation results demonstrate that smooth handover with our scheme can achieve better TCP performance than that without it.

Due to the fact that future wireless network will support multiclass services, so it is imperative to differentially treat various kinds of traffic types. An adaptive wireless resource management mechanism based on priority is proposed in this paper. Firstly, the mechanism estimates the reserved resource that is needed to support the necessary QoS in terms of handoff call dropping probability, then sets the thresholds according to the priority of various traffic types. The differentiated admission control scheme satisfies QoS requirement of various traffic types and admits high-priority traffic preferentially under high traffic load. The scheme proposed in this paper is simple and adaptive. Finally the paper presents the Markov model of this differentiated admission control, uses the numerical method to analyze the performance of this scheme, and compares the numerical results with simulation data. The results match relatively well.

This paper has first presented an in-depth analysis on the distributed coordination function (DCF) access mode of IEEE802.11 protocol. Based on the result of our study, we have concluded a new self-adapt wireless LAN MAC algorithm. Numerous simulation results have shown that the new algorithm can achieve the self-adapt character with the growing of node number and is superior to the original DCF algorithm in the characters concerned (e.g. throughput, fairness).

The queuing/scheduling algorithm is one of the most important mechanisms to provide guaranteed quality of service (QoS) in high speed packet-switched networks. By computing the system virtual time and per packet/connection virtual start/finish time, a number of packet fair queueing (PFQ) algorithms are designed to simulate GPS (generalized processor sharing). The difference in computation complexity is due to the variable ways to compute system time. Many algorithms, including WFQ/WF/sup 2/Q, SCFQ, SPFQ, WF/sup 2/Q+, VC, etc., have been proposed to use different system virtual time (also known as system potential) function. This paper proves that all the latency-rate (LR) servers only need to calculate their system virtual times once per packet service time, no matter how many packet arrivals occur in this interval. Thus, it is a general scheme that benefits all the well-known LR PFQ algorithms.

Being a part of IEEE project 802, the 802.11 medium access control (MAC) is used to support asynchronous and time bounded delivery of radio data packets. It is proposed that a distributed coordination function (DCF), which uses carrier sense multiple access with collision avoidance (CSMA/CA) and binary slotted exponential backoff, be the basis of the IEEE 802.11 WLAN MAC protocols. This paper proposes a throughput enhancement mechanism for DCF by adjusting the contention window (CW) resetting scheme. Moreover, an analytical model based on Markov chain is introduced to compute the enhanced throughput of 802.11 DCF. The accuracy of the model and the enhancement of the proposed scheme are verified by elaborate simulations.

IEEE 802.11 medium access control (MAC) is proposed to support asynchronous and time bounded delivery of radio data packets in infrastructure and ad hoc networks. The basis of the IEEE 802.11 WLAN MAC protocol is a distributed coordination function (DCF), which is a carrier sense multiple access with collision avoidance (CSMA/CA) with a binary slotted exponential back-off scheme. Since IEEE 802.11 MAC has its own characteristics that are different from other wireless MAC protocols, the performance of reliable transport protocol over 802.11 needs further study. This paper proposes a scheme named DCF+, which is compatible with DCF, to enhance the performance of reliable transport protocol over WLAN. To analyze the performance of DCF and DCF+, this paper also introduces an analytical model to compute the saturated throughput of WLAN. Compared with other models, this model is shown to be able to predict the behavior of 802.11 more accurately. Moreover, DCF+ is able to improve the performance of TCP over WLAN, which is verified by modeling and elaborate simulation results.

IETF proposes the differentiated services (DiffServ) architecture to implement QoS in the Internet. However, recent studies have shown that under various conditions, the existing DiffServ mechanism may have problems of unfairness and inefficient resource utilization, thereby failing to achieve the desired QoS for TCP flows running over assured services. This paper considers that the idea of TCP friendly congestion control should be used in traffic conditioning (TC) for AF PHB (assured forwarding per-hop behavior), especially when the customer expects to achieve better performance by out-of-profile traffic. This paper also proposes using a direct congestion control scheme (DCCS) and dynamic traffic conditioning (DTC) to achieve fairness between responsive and unresponsive aggregates as well as better allocations for responsive flows (TCP) with different packet lengths, micro-flow number, round trip time (RTT) etc. Moreover, a better resource utilization can be achieved at the same time.

In this paper, we present an effective scheduling algorithm based on the RPS model, called simple weighted fair queueing (SWFQ). In SWFQ, computation of the system potential function does not require such division or multiplication operations as in MD-SCFQ. Compared with MD-SCFQ, SWFQ has lower complexity and can be easily implemented in chips. We verify the effectivity of proposed SWFQ through strict theoretical analysis.

As an important mechanism to provide QoS guarantee in packet-switched networks, queueing scheduling algorithms have been widely researched. The rate-proportional server (RPS) is a good fluid model that covers the general processor sharing (GPS) models. We can design corresponding packet-level queueing scheduling algorithms through choosing different system potential correction function of RPS. We present an enhanced rate-proportional server (ERPS), which gives the upper and lower bounds of system potential correction function and compute corresponding fairness index. Thus, we can easily devise the needed packet-level queueing scheduling algorithm based on ERPS. We also give a packet-level queueing scheduling algorithm based on ERPS.

Precision and efficiency are the two most important goals of simulation. As far as communication networks are concerned, for the great complexity of their topologies, protocols and traffic, how to simulate them efficiently has been a hot spot in the recent research. This article first emphasizes the importance of the survey before the simulation begins, because it can make the target of simulation clear. Also, during the modeling phase prior to simulation, we should consider adequately the characteristics of the communication network to be simulated and adopt some methods in order to improve the simulation efficiency.

CBT and PIM-SM are two typical intradomain multicast routing protocols. However, CBT lacks an efficient member management mechanism such as admission control and authentication for senders of multicast group, and PIM-SM lacks robustness. This paper proposes a new one, which can effectively solve the both problems through admitting senders at multiple cores and transporting packets in a uni-direction. Furthermore, the protocol is easy to implement.

Traffic prediction is a new research subject in ATM traffic management. It is the basis of dynamic and adaptive traffic control. This paper, based on the theory of fuzzy system, presents a fuzzy traffic predictor using nearest neighborhood clustering learning algorithm. We examine its prediction effect through autoregressive (AR) process and real video traffic. The results show that the approximator/predictor is accurate and flexible. The applications of the traffic predictor in ATM networks include dynamic bandwidth allocation, traffic shaping, traffic smoothing, service synchronization etc. We mainly discuss dynamic bandwidth allocation based on the fuzzy traffic prediction. The research results show that the dynamic bandwidth allocation can reduce cell loss ratio, buffer requirement and improve the bandwidth utilization remarkably.

This paper addresses an optimal link capacity design problem for self-healing SDH networks. By taking advantages of the binary search and inducing two parameters, we proposed an enhanced algorithm, binary search optimisation algorithm (BSOA), which results in a great reduction on the computation time and network installation cost (spare capacity). The proposed algorithm was tested on three sample networks for several typical traffic demands. The results indicate that this algorithm is more effective than before.

Network/service survivability is an important problem demanding prompt solution in the design and planning of new telecommunications networks (e.g., SDH networks and ATM networks). Different restoration mechanisms for SDH and ATM networks have already been developed. The previous literature comparing and evaluating the restoration mechanisms only uses a simple method. In order to overcome the shortcomings in the traditional method, this paper proposes the customer-oriented and network engineer-oriented definitions of restoration mechanism availability. In the second definition, the extra spare capacity requirement in the network is taken into account in the availability analysis. Because an exact measurement of the availability is not feasible due to the large number of failure states in the network, an ORDER algorithm (Li and Silverter 1984) to get a reasonable subset to approximately measure the availability is recommended. Then, applications of the definitions are given. One application is how this definition is used by network planner/engineer to construct a cost-effective survivable network. Another application is that a new group of parameters-survivability quality of service (SQoS) is defined and added to CAC (connection admission control) of the ATM survivable network. Finally, a simulation using two definitions is performed. The simulation results prove that two availabilities can evaluate restoration mechanisms more sufficiently and more meaningfully.

There are different priority classes of cells in ATM networks. The policing mechanisms seen in the literature only can police total traffic amount or total rate because they only use one token generator and one token pool. We present six mechanisms using priority leaky buckets to police traffic with different priority-classes of cells. All the policing mechanisms use two token generators and two token pools. They can distinguish the classes of arriving cells. They can police the rate of class-0 traffic as well as the total rate of the priority traffic. Their performance are studied through simulations.