In this paper, we survey the fairness issues in the congestion control mechanisms of TCP, which is one of most important service aspects in the current and future Internet. We first summarize the problems from a perspective of the fair service among connections. Several solution methods are next surveyed. Those are modifications of TCP congestion control mechanism and router support for achieving the fair service among TCP connections. We finally investigate the fair share of resources at endhosts.

Over the future Internet, the real time communication generating such as CBR (Constant Bit Rate) traffic will widely spread, whereas the current Internet has no ability to provide QoS (Quality of Service) assurance for real time communication so far. In QoS networks, CBR traffic will have priority for its stringent QoS requirement over non-real time traffic such as TCP connections, which use the unused bandwidth left by CBR connections. Therefore, there is possibility that CBR traffic with priority causes TCP throughput degradation in QoS networks. For this reason, the performance of Tahoe TCP has been examined in that context, but other TCP variants such as Reno TCP, NewReno TCP and TCP with SACK option, which are now very common, have not yet been investigated clearly. In the present research, we will clarify how these TCP variants behave in QoS networks by means of simulations and compare their performance. From the results, SACK TCP can adapt very well to the changing bandwidth available and is very robust against the fluctuation, i.e., burstness, of CBR packet arrival process.

By the growth of the Internet, the client's access to the Internet have been increasing. In order to meet those demands, asymmetric networks have become a great attraction. However, asymmetric networks composed of satellite and terrestrial links have not been enough investigated. Therefore, this paper proposes a new formula for TCP performance evaluation for the asymmetric networks. Using this evaluation formula, we calculate the throughput of TCP Reno over the asymmetric networks taking Slow Start into account. The calculation results are compared with the following: (1) the value based on an existing theoretical formula, (2) the outputs of simulation by NS (Network Simulator), and (3) the experimental results using VSAT (Very Small Aperture Terminal) satellite communication system for satellite links and the Internet for terrestrial links. As a result, it is shown that the new formula is more precise than the one already proposed.

The differentiated services (diffserv) architecture has been proposed for implementing scalable service differentiation in the Internet. Expedited forwarding and assured forwarding have been standardized as Per-Hop Behaviors (PHB) in diffserv. Assured forwarding can be utilized to realize the service, which provides each user with a minimum guaranteed rate and a fair share of the residual bandwidth. We call it guaranteed rate (GR) service. With GR service, each packet for flow i is marked in or out based on comparison between the sending rate and the minimum guaranteed rate. When congestion occurs in networks, out packets are dropped more aggressively than in packets. Recently, several fair queuing schemes have been proposed for core stateless networks. They can achieve fairer bandwidth allocation than random early detection (RED). However, there have not been any studies that consider in/out bit usage to support GR service. This paper proposes how to extend the schemes that have been proposed for core stateless networks to allow the support of in/out bit usage. We present the performance of one of the extended schemes and compare the scheme to RED with in/out bit (RIO) in terms of fair bandwidth allocation.

We have developed a new scheme to provide Diffserv-based QoS over ATM access networks. Well-known Diffserv over ATM scheme requires some extension for conventional routers with ATM interfaces. The routers must map their Diffserv classes of services into ATM QoS classes and forward IP packets into prioritized VCs based on DSCP (DiffServ Code Point). The purpose of this work is to provide Diffserv-based QoS over ATM network using conventional IP over ATM interfaces on routers. We propose DSCP snooping at ATM edge nodes, which differentiates services over a single VC between two IP domains. A prototype circuit was used to evaluate this scheme.

A packet transmission delay is an important quality characteristic for various applications including real-time and data applications. In particular, it is necessary to investigate not only a whole distribution of the packet transmission delay, but also the tail part of the distribution, in order to detect the packet loss. In this paper, we analyze the characteristics of the tail part of packet delay distributions by statistical analytic approach. Our analytic results show that the Pareto distribution is most appropriate in 95-99.9% region of the cumulative distribution of packet transmission delays. Based on our statistical analysis, we next propose an adaptive playout control algorithm, which is suitable to real-time applications. Numerical examples show that our algorithm provides the stable packet loss ratio independently on traffic fluctuations.

Recently, traffic of real-time communication such as video or IP telephony is increasing in Internet. Low delay and low jitter communication is required to perform these kinds of services smoothly. CoS (Class of Services) scheme which controls traffic by the class unit such as application is currently being paid attention. It is considered to be efficient control mechanism which utilizes characteristics of Internet traffic. One of effective methods for queueing control to realize CoS is CBQ (Class Based Queueing). Efficient and dynamic bandwidth allocation can be realized by using CBQ. However, CBQ mechanism is not necessarily efficient for real-time communication. In this paper, We clarify the effect of dynamic CBQ control scheme on real-time communication by experiment. In addition, we show that low delay and low jitter communication is possible by reducing unnecessary control overhead which is caused by traffic change. We propose new mechanism which changes bandwidth allocation of CBQ dynamically suppressing non real-time traffic and giving priority to real-time traffic on using bandwidth. We also show its efficiency by using real system.

Network dimensioning is an important issue to provide stable and QoS-rich communication services. A reliable estimation of bandwidths of links between the end-to-end path is a first step towards the network dimensioning. Pathchar is one of such tools for the bandwidth estimation for every link between two end hosts. However, pathchar still has several problems. If unexpectedly large errors are included or if route alternation is present during the measurement, the obtained estimation is much far from the correct one. We investigate the method to eliminate those errors in estimating the bandwidth. To increase the reliability on the estimation, the confidence interval for the estimated bandwidth is important. For this purpose, two approaches, parametric and nonparametric approaches, are investigated to add the confidence intervals. Another important issue is the method for controlling the measurement period to eliminate the measurement overheads. In this paper, we propose a measurement method to adaptively control the number of measurement data sets. Through experimental results, we show that our statistical approaches can provide the robust estimation regardless of the network conditions.

One drawback of Integrated service architecture is the scaling problem. Therefore, flow aggregation is an important solution for supporting quality of service in large-scale network. In advance reservation, priori information of advance-reserved requests can be used for flow aggregation before their initiation time. However, an impolitic aggregation can lead to violate admission control. In this paper, we propose an effective algorithm to aggregate advance-reserved requests with guaranteed delay. The proposed algorithm not only can reduce the amount of state in core network but also minimize the bandwidth consumption. The simulation result indicates that the state in the core network can be reduced as low as 17.3% even in the worst case.

In order to make the ATM network fault-tolerant and the network service flexible, a method for the setting up of backup virtual paths (VP's for short) using multiagents is effective with respect to adaptability to change of network resource and user requirements, examples of which are failure of nodes and links and addition of VP's, respectively. In this method, under the assumption that candidates of backup VP's between different pairs of source and destination nodes are given, the optimum backup VP's are obtained by exchanging information among agents autonomously. First, this paper proposes measures for determining backup VP's between different pairs of source and destination nodes. Next, this paper presents simulation results to evaluate the adaptability of the method. The results show that the method efficiently obtains the optimum backup VP's even when the number of backup VP's increases and that different idle time at each destination node enables to shorten the total processing time while keeping complete detection of shared links.

In QoS networks, routing algorithms for QoS traffic have to provide the transmission path satisfying its QoS requirement while achieving high utilization of network resources. Therefore, server-based QoS routing algorithms would be more effective than distributed routing ones which are very common on the Internet. Furthermore, we believe that rerouting function enhances the advantage of their algorithms in which an already accepted flow with established path is replaced on some other path in order to accept newly arriving transmission request if it can not be accepted without doing so. Thus in this paper, we will propose a rerouting algorithm with the server-based QoS routing and evaluate its performance in terms of the blocking probability by computer simulation. In addition, we will investigate the impact of the amount of traffic with high-priority on the performance in some network topologies. Through some simulation results, we also discuss some issues arising in improving the effectiveness of rerouting.

In server load balancing where replicated servers are dispersed geographically and accesses from clients are distributed to replicated servers, a way of distributing the accesses from clients to an adequate server plays an important role from the viewpoint of load balancing. In the paper, we propose a new network paradigm for server load balancing using active anycast. In active anycast, an end user only sends its request to group of servers using an anycast address. When this request arrives at an active router, it selects an adequate server from the viewpoint of load balancing and changes the anycast address of a packet to the unicast address of the selected server. The decision which server is the best one from the viewpoint of server load balancing is made by an active router rather than an end user, so active anycast is a network-initiated method. Simulation results show that active anycast can accomplish efficient server load balancing, even when a small part of routers are equipped with active network technology.

In order to increase the flexibility and the extensibility for packet transmission, the active network approach, that makes network nodes programmable, is very promising. Every packet includes a program specifying behavior of the packet at network nodes, and the network nodes only have the functions to execute programs of the packet. In an active network, network congestion management is achieved more flexibly and intelligently than that in the conventional internet. A number of mechanisms for congestion management are easily developed because various network functions such as QoS routing and congestion detection are easily combined in the same network nodes. By executing different routing protocols that use different levels of the link information, a congestion management scheme can be customized by users. This paper shows how QoS routing and congestion detection achieve a congestion management over an active internetwork system.

In wireless ad hoc mobile network, a host desires to communicate with another host may need some intermediate nodes to relay data packets. To maximize the channel resource utilization and minimize the network transfer delay along the path, the shortest path with minimum hops approach is often adapted. However, by considering the employing medium access control (MAC) protocol, the minimum transfer delay from source to destination may be achieved by choosing a longer path but with less contention delay. In this paper, we will propose an efficient delay-oriented routing protocol for mobile ad hoc wireless networks. The expected access contention delay of IEEE 802.11 protocol is analyzed to support the routing decision. Simulation results show that the derived path length in proposed delay-oriented routing protocol is slightly higher than that of conventional shortest path with minimum hops approach but it can significantly reduce both average transfer delay and packet loss rate.

In this letter, performance evaluation of a system that combines between Code-Division Multiple Access (CDMA) and ALOHA protocol in multimedia networks is presented. In our analysis, we compare the performance between the two basic techniques of ALOHA protocol, i.e., Slotted-ALOHA (S-ALOHA) and Unslotted-ALOHA (U-ALOHA), when combined with CDMA scheme to support voice and data users operating in same CDMA channel. The quality of service (QoS) required for voice and data media is completely taken care of. We obtain the throughput of data media, and the outage probability for voice considering both voice and data offered loads. Throughput performance of S-ALOHA technique is almost twice of that of U-ALOHA. However, we show in this letter that when we combine CDMA with the two basic techniques of ALOHA to accomplish multimedia transmission, both techniques have almost same performance. And, thus, CDMA U-ALOHA can be a good candidate for multimedia networks.

This paper proposes a λ-ring system that is a wavelength-based self-healing-ring application unlike ordinary fiber-based ones. To design survivable networks of interconnected such self-healing ring systems, a virtual mesh network scheme is used, in which wavelength assignment in virtual links can be considered according to the λ-ring-system or fiber-ring-system applications of the bidirectional wavelength-path switched architecture. Integer-programming-based design problems are then formulated that minimize the total fiber length in these self-healing-ring applications. Numerical examples show that the λ-ring-system application is always superior to 4-fiber and 2-fiber-ring-system applications and 1+1 end-to-end path protection.

This paper focuses on the comparison of OFDM system channel estimation using time domain techniques and using frequency domain techniques. The channel model is based on the Taiwan DTV field-testing results, with static and dynamic multipath distortion. The simulation results prove that the channel estimation performance of the OFDM system in the time domain is better than in the frequency domain.

Sequence estimation (SE) of narrow-band digital FM signals, such as CPFSK and GMSK, with non-coherent limiter/discriminator (L/D) and integrate and dump (I&D) detection is investigated in detail using both analysis and simulation. The BER is studied from approximate upper and lower bounds obtained through Chernoff bounding techniques and minimum error event path probability along with a Gaussian noise assumption for high SNR. Various IF filters and the dependence of the error probability upon modulation index are considered. The results show an optimum modulation index around h 0.55, and clearly demonstrate the effectiveness and limitations of sequence estimation.

The inter-cell interference between uplinks and downlinks in CDMA packet communication systems, employing a shared-TDD scheme, is evaluated under cellular environments. It is found that interference between base stations rarely degrades uplink throughput, but interference between mobile stations substantially degrades downlink throughput. A transmission power control scheme is proposed to improve the downlink throughput. The proposed scheme increases the transmission power of downlink packets when they are re-transmitted, and thus, improves the signal-to-interference ratio of the downlink re-transmission packets. Computer simulation shows that this scheme increases downlink throughput without sacrificing the uplink throughput until the uplink throughput reaches a maximum value.

The stage 3/2 decoding scheme, originally suggested by U. Timor, is modified for a Rayleigh fading channel to improve the performance of a fast frequency-hopped multiple access/multilevel frequency shift keying system. When signal-to-noise ratio per bit is 30 dB, the simulation results show that the modified stage 3/2 decoding scheme increases the spectral efficiency by 11% compared to the modified stage 1 decoding scheme at bit error rate of 10^-3. Further, the performance comparisons are made between the modified multistage decoding scheme and the diversity combining methods, where the modified stage 3/2 decoding scheme shows better performance.

We describe a simplified receiver structure having several receiving antennas (i.e., an adaptive array antenna system) and using time-division-multiplexing (TDM) signal processing. Three simplified receiver structures were investigated for use in the antenna system. To confirm the feasibility of using a TDM receiver, both a TDM receiver and a conventional adaptive array receiver were constructed for testing. In our proposed system, several repetitions of the constant modulus algorithm (CMA) are used to reduce co-channel interference (CCI). The frame format used for both receivers was the same as that of the personal handy phone system in Japan. The laboratory testing was done using a fading simulator to enable measurement of the bit error rate. The results are very promising and show the feasibility of the TDM receiver.

To improve the throughput efficiencies of ARQ protocols over a high random packet-error channel, contiguous multiple copy-transmission (CMCT) strategy for which each packet is (re-)transmitted by sending its multiple copies in contiguous slots has been used so far. However, in burst error environments, all copies may be damaged in an error burst resultting the performance degradation of CMCT. To cope with this situation, we propose, in this paper, a new strategy called intermittent multiple copy-transmission (IMCT) whereby multiple copies are sent at a fixed interval. The throughput efficiency of go-back-N ARQ using CMCT or IMCT is analyzed and considered under a two-state Markov channel model expressing burst error property of a channel. As a result, it is shown that (i) the degree of improvement of throughput efficiency by CMCT or IMCT depends on the degree of error burst and (ii) the proposed IMCT can improve the throughput efficiency of go-back-N ARQ for high and burst error channels.

In distributed network systems, it is one of the most important problems how to assign the files to servers in view of cost and delay. It is obvious that there is a trading-off relationship between costs and delays in these systems. In order to evaluate the optimization that the total cost is minimized subject to the total delay, we have presented the Optimal File Allocation Model as 0-1 integer programming, and have investigated the general characteristics in distributed systems. In this model, we have introduced many cost and delay parameters to evaluate the total cost and delay in the system more exactly. In constructing practical systems, it is necessary to investigate the weight and the contribution of each parameter to the total cost. It is very useful to show how to estimate cost and delay parameters on the basis of this analysis. In this paper, we analyze the sensitivity of these parameters and make clear the influence between principal parameters.

We propose a fast and compact longest match table look-up method for very long network addresses like IP version 6. This method uses two ideas for a routing-table arranged in a tree-structure. The first idea is to make table look-up fast by caching pointers to intermediate nodes in the tree, reducing the number of node traversals. The second idea is to reduce the memory size required for each node in the tree by one-third by eliminating common parts of addresses of adjacent nodes. Evaluating the performance of this method by using actual routing table data of an IP backbone network, we found it was five to ten times faster than a conventional method.

An AGC Applebaum array, which is a modified Applebaum array employing an automatic gain controller (AGC), is proposed. When the eigenvalues of the input covariance matrix of an array system are spread by orders of magnitude, conventional adaptive arrays are unable to remove all the interference signals quickly. The proposed array increases the cross-correlation between the low-power interference signals at the array input and output through the use of an AGC block in the feedback loop. As a result, the weight vector is adapted for the removal of both low-power and high-power interference signals. Computer simulations were performed to demonstrate that the proposed array can produce high output signal to interference plus noise ratio (SINR) with a fast convergence speed.

A new charge pump is proposed which provides improved jitter characteristics for a phase-locked loop (PLL). The PLL with the proposed charge pump is implemented with 0.6 µm CMOS technology. The measured RMS output jitter is as much as 28% smaller than that of a PLL with a previously reported charge pump structure.

In this paper, we investigate the bit error rate (BER) of OMC-CDMA (Orthogonal Multi-Carrier-Code Division Multiple Access) with Maximal Ratio Combining (MRC) scheme on two-ray Rayleigh multipath fading channels, and compare the MRC scheme with the Equal Gain Combing (EGC) scheme according to not only the number of users (M) and subcarriers (N) but also the power of delayed signal. We found that the MRC scheme outperforms the EGC scheme, and the BER of OMC-CDMA depends greatly on the power of delayed signal. If the delayed path power is increased from 10% to 50%, as compared with the direct path power, CDMA capacity is decreased approximately by 30% at BER=10^-3, N=256 and SNR=15 dB. When the system is heavily loaded by users in large decaying parameter, the performance of OMC-CDMA with the MRC scheme can not increase the BER significantly compared to that with the EGC scheme.

This paper presents closed form expressions to evaluate the average bit error rate (BER) of coherent binary phase shift keying (BPSK) and quadrature PSK (QPSK) systems in the presence of Nakagami-m fading channel and noisy phase reference. Performance degradation due to noisy phase reference is investigated versus both the fading parameter m and the maximum phase error φ. When m is increased from 1 to 9 and φ = 30, the degradation at the average BER of 10^-3 for BPSK is increased from 0.3 dB to 0.48 dB. For φ increasing from 10 to 40 and m=5, the degradation is increased from 0.06 dB to 0.92 dB. Degradation thus increases with increasing φ and m.

This paper proposes a new MAC scheme for ATM based Passive Optical Network (APON) which follows the ITU-T recommendation G.983.1. MAC schemes for APON mostly adopt single-side scheduling in which only OLT (Optical Line Termination) schedules upstream access order for ONUs (Optical Network Units) that inform cell arrival information to OLT. In this paper, we propose a MAC scheme called double-side scheduling protocol (DSP) in which both OLT and ONU participate in scheduling process. Through the computer simulation we find out that the CDV (cell delay variation) performance of the real-time service can be improved remarkably using fewer overhead slots than that of the MAC schemes adopting CDV control algorithm. For the proposed MAC scheme, we present grant field format, minislot format, divided slot format, and scheduling algorithm.

This paper proposes a framework for building a customized comparison-shopping system in which the users are allowed to add their own shopping stores dynamically. In this framework, a shopping agent is implemented with a robust inductive learning method that automatically constructs wrappers for semi-structured online stores. During learning, strong biases assumed in many existing systems are weakened so that the real stores with reasonably complex document structures can be handled.