This paper describes a proposed propagation estimation method and TCP/IP-based evaluations for mobile communications employing a stratospheric platform. To estimate a wireless channel, a realistic and detailed description of its physical environment must be accurately defined. Therefore, a building distribution model characterizing the physical environment in areas in Japan is presented. The analyses of the propagation estimation method are based on the "ray-tracing" model. The results from the proposed method are derived depending on elevation and azimuth angles. In order to validate our results, comparisons between the proposed method and our previous measurement are made for a typical semi-urban area in Japan. The comparisons show close agreement between the estimation results and the measurement results. Finally and interestingly, we present communication performance evaluations based on TCP/IP protocol by using the results achieved from our channel estimation with semi-analytical and simulation approach.

Overlay networks are expected to be a promising technology for the realization of QoS (Quality of Service) control. Overlay networks have recently attracted considerable attention due to the following advantages: a new service can be developed in a short duration and it can be started with a low cost. The definition and necessity of the overlay network is described, and the classification of various current and future overlay networks, particularly according to the QoS feature, is attempted. In order to realize QoS control, it is considered that routing overlay and session overlay are promising solutions. In particular, session and overlay networks are explained in detail since new TCP protocols for QoS instead of current TCP protocols that control congestion in the Internet can be used within overlay networks. However, many open issues such as scalability still need further research and development although overlay networks have many attractive features and possess the potential to become a platform for the deployment of new services.

Using the Proportional Integral Derivative (PID) principle of classical feedback control theory, this paper develops two general congestion control algorithms for routers implementing Active Queue Management (AQM) while supporting TCP/IP traffic flows. The general designs of non-interacting (N-PID) and interacting (I-PID) congestion control algorithms are tailored for practical network scenarios using the Ziegler-Nichols guidelines for tuning such controllers. Discrete event simulations using ns are performed to evaluate the performance of an existing F-PID and new N-PID and I-PID algorithms. The performance of N-PID and I-PID is compared mutually as well as with the F-PID algorithm. It reveals that N-PID and I-PID have higher speed of response but lower stability margins than F-PID. In general the accurate following of the target queue size by the PID principle congestion control algorithms, while providing high link utilization, low loss rate and low queuing delays, is also demonstrated.

This paper deals with optimizing end-to-end window control algorithms that achieve proportional fairness in the long run, and also investigating their steady state and transient or short-term fairness. An abstracted stochastic model of a bottlenecked connection is employed and the window control is designed to minimize the buffer queue variance while keeping the mean queue level at a target value, taking the round-trip delay into account. An optimal minimum variance window control algorithm and its generalized version are derived with the major aim of reducing the window size fluctuations. A relation between the generalized version and the (p, 1) proportionally fair algorithm is investigated. Also we derive the stability limits of the generalized minimum variance window control algorithm which can be useful during network design. The effects of both the target queue length and a weighting parameter on the steady state and short-term fairness, as measured by Jain's fairness and short-term fairness indices, calculated over short time intervals, are investigated by simulation. The generalized algorithm has better fairness properties as compared to existing algorithms such as that employed in TCP Vegas.

The current congestion control mechanism for TCP with ECN suffers from the oscillation of the window size and the unfair sharing of network resources. These problems are caused by the insufficient information about the congestion level of a network and the different round trip time (RTT) among the connections respectively. In this paper, we propose a new window control algorithm for TCP with ECN to avoid the oscillation of the window size and to achieve the fairness among the connections.

We have developed an OC-48c (2.4 Gbps) PCI-compliant network interface card and drivers with the aim of evaluating the effectiveness of our proposed link layer protocol MAPOS. In this paper, we study the effectiveness of MAPOS particularly from the viewpoint of the influence of packet sizes up to the 64-kbyte jumbo MTU size and the effectiveness of our new implementation of the non-interrupt-driven sending process and interrupt batching receiving process deployed to improve the throughput in short-packet transmissions. Our main findings are as follows; Enlarging the packet size up to 64-kbyte MTU improves the performance in transmission. OC-48c wire speed is achieved with packet sizes larger than 16 kbytes. Implementation of the non-interrupt-driven sending process and the interrupt batching receiving process improves the performance of short-packet transmission. In particular, the transmission throughput is improved by 50% when 64-byte short packets are used. The maximum loss-free receive rate is also raised by 50% when 4-kbyte packets arrive. With a high-performance CPU, the data-transfer speed of the DMA controller for jumbo packets cannot keep up with the packet-queueing speed of the CPU. Our proposed procedure for adaptive algorithm switching method can resolve this problem. The maximum TCP throughput observed in our measurement using the latest PCs and MAPOS OC-48c PCI card was 2342.5 Mbps. This throughput represents the highest performance in a legacy-PCI-based system according to the results database of the benchmarking software.

We proposed a new buffer management scheme for GFR services through FIFO queuing discipline. Proposed scheme can provide minimum bandwidth guarantee for GFR VCs as well as improve the fairness among the competing GFR VCs on a single FIFO queue. From simulation result, we demonstrate the proposed scheme fulfills the requirements of GFR services as well as improves the TCP throughput

During TCP/IP communications, MobileIP routing optimization functions causes out-of-order TCP packet sequences. To solve this problem, we propose a dynamic delayed ACK control scheme in which the wireless link-state management part notifies the upper TCP/IP layer of base-station hand-over, and at this time the TCP/IP layer sends dynamic delayed ACKs in response by using two-level-timer (i.e., hard-timer and soft-timer) processing. Simulation results confirm that applying dynamic delayed ACK control to MobileIP networks improves average throughput.

This paper proposes an idea for modeling aggregated TCP/IP traffic arriving at a bottleneck link by focusing on its scaling behavior. Here, the aggregated TCP/IP traffic means the IP packet traffic from many TCP connections sharing the bottleneck link. The model is constructed based on the outcomes of our previous works investigating how the TCP/IP networking mechanism affects the self-similar scaling behavior of the aggregated TCP/IP traffic in a LAN/WAN environment. The proposed traffic model has been examined from the perspective of application to network performance estimation. The examinations have shown that it models the scaling behavior and queueing behavior of actual traffic, though it neglects the interaction among TCP connections that compete with each other for the single bottleneck link bandwidth.

We proposed an improved Hierarchical Packet Fair Queueing (H-PFQ) mechanism, using ACK Spacing, for efficient bandwidth management of TCP traffic over Internet. According to the pre-determined bandwidth sharing and the class hierarchy of all TCP sessions, we design an algorithm to calculate the required time intervals between consecutive ACK packets of each TCP session to avoid packet drops due to buffer overflow. We demonstrated via computer simulations that the proposed improvement techniques may result in much better performance than merely original H-PFQ mechanism used in the forward direction in the sense that not only effective throughput of the bottleneck link is improved but also the fairness among TCP sessions can be maintained.

Nowadays, many IP based communication systems are connected by high-speed networks to realize high-speed TCP/IP communication. However, since the mechanism of TCP/IP protocols is based on a best effort service, the quality of the communication may change by the time or the route of the traffic. Therefore, it is important for network providers to investigate the quality of their users' communication. In order to analyze the traffic on an Internet backbone, the realtime analysis is one of the important factors. So far, several tools were developed for the purpose of the traffic measurement. However, none of them can analyze sufficient statistics to evaluate the quality for each end user's communication in realtime. Therefore, we have designed and implemented a performance monitor, which collects the statistics representing the performance such as TCP throughput while capturing the traffic. The statistics are collected by every pair of IP addresses and by every application. The monitor also provides the function to analyze effectively for the statistics records such as sorting and filtering of the records and the graphical user interface to operate the software tool. This paper describes the design and implementation of the performance monitor.

We describe a control scheme for wireless-link layers and the TCP/IP layers in which wireless link states, such as signal strength and transmission rate, are transparent to the upper network layers and govern the behavior of the upper layers. Monitoring and notifying functions for wireless link states are incorporated into mobile hosts, and prefetching functions for the mobility agent lists are added to the present MobileIP functionalities of the network layers. In addition, we give the mobile hosts functions for controlling the TCP advertised windows in the transport layers and give the base-stations buffering functions to deal with the variable signal strength of the wireless links. A simulation (using ns-2) of this control scheme shows that mobile agent can be switched at the network layers soon after base-station hand-over and that errors in packet routing, packet loss and communication throughput reduction can be avoided. Moreover, communications can be interrupted without having to shrink the size of the congestion windows of the TCP senders, which improves overall throughput.

This paper discusses and clarifies effectiveness of data-driven implementation of protocol handling system to access TINA (Telecommunications Information Networking Architecture) network and internet. TINA is a networking architecture that achieves networking services and management ubiquitously for users and networks. Many TINA related ACTS (Advanced Communication Technologies and Services) projects have been organized in Europe. In Japan, The TINA Trial (TTT) to achieve ATM network management and services based on TINA architectures was done by NTT and several manufactures from April 1997 to April 1999. In these studies and trials, much effort is devoted to development of software based on service architecture and network architecture being standardized in TINA-C (TINA Consortium). In order to achieve TINA environment universally in customers and network sides, we have to consider how to deploy TINA environment onto user side and how to use access transmission capacity as efficiently as possible. Recent technology can easily achieve application and environment downloading from the network side to user side by use of e. g. , JAVA. In accessing the network, there are several possible bottlenecks in information exchange in customer side such as PC processing capability, access protocol handling capability, intra-house wiring bandwidth. Authors, in parallel with TINA software architecture study, have been studying versatile requirements for hardware platform of TINA network. In those studies, we have clarified that the stream-oriented data-driven processor authors have been studying and developing have high reliability, high multiprocessing and multimedia information processing capability. Based on these studies, this paper first shows Von Neumann-based protocol handler is ineffective in case of multiprocessing through mathematical and emulation studies. Then, we show our data-driven protocol handling can effectively realize access protocol handling by emulation study. Then, we describe a result of first step of implementation of data-driven TCP/IP protocol handling. This result proves our TCP/IP hub based on data-driven processor is applicable not only for TINA/CORBA network but normal internet access. Finally, we show a possible customer premises network configuration which resolves bottleneck to access TINA network through ATM access.

This paper proposes a new simple method for network measurement. It extracts 6-bit control flags of TCP (Transmission Control Protocol) packets. The idea is based on the unique feature of flag ratios which is discovered by our exhaustive search for the new indexes of network traffic. By the use of flag ratios, one can tell if the network is really congested. It is much simpler than the conventional network monitoring by a network analyzer. The well-known monitoring method is based on the utilization parameter of a communication circuit which ranges from 0% to 100%. One cannot tell the line is congested even if the factor is 100%. 100% means full utilization and does not give any further information. To calculate the real performance of the network, one should estimate the throughput or effective speed of each user. The estimation needs much calculation. Our new method tries to correlate ratios of TCP control flags and network congestion. The result shows the usefulness of this new method. This paper analyzes the reason why the flag ratios show the unique feature.

The TCP/IP protocol suite is the standard requirement for all applications that need to communicate over the Internet. As TCP/IP applications are unable to specify the QoS parameters needed for most Asynchronous Transfer Mode (ATM) services, they tend to use the unspecified bit rate (UBR) service category when running across ATM networks. The UBR service utilizes any bandwidth that is left unused by the rest of the ATM services. This has led the ATM Forum's Traffic Management Group to define a new service category called guaranteed frame rate (GFR). GFR is intended to provide minimum cell rate guarantees and fair access to excess bandwidth left over from higher-priority services. This article first presents a tutorial overview of GFR and then presents a survey of the research work that has been carried out toward the design and implementation of associated ATM switch mechanisms.

It is important to establish the technology to accommodate best effort TCP/IP traffic over wide area ATM networks. The UBR (Unspecified Bit Rate) service category is the most typical service category for the best effort traffic, especially in the LAN environment. On the other hand, the VBR (Variable Bit Rate) service category with SCD (Selective Cell Discard) option is considered as the service category which is appropriate for wide area networks due to its fairness and minimum guarantee of the cell transmission using not only PCR (Peak Cell Rate) but SCR (Sustainable Cell Rate) and MBS (Maximum Burst Size). However, there is no actual evaluation for such service. We have, therefore, performed the experimental studies on TCP/IP over VBR with SCD along with UBR and VBR without SCD by VC (Virtual Channel) level policing when each TCP connection is mapped to a different VC. Through these experiments, we measured the link utilization of the effective data and the fairness between each obtained TCP throughput during the congestion of the ATM switch. From the results of the link utilization, the value is over 95% under the various conditions. Therefore, even in the case of the cell losses due to SCD or buffer overflow in ATM switch congestion, average throughput is almost the same as the value which equals the trunk line speed divided by the number of the accommodated TCP connections. From the results of the fairness, VBR with SCD per VC is better than UBR and also obtains better TCP throughput than VBR without SCD. Furthermore, to confirm those characteristics more generally, we adopt the accommodated TCP connections not only with the same TCP send/receive socket buffer size but with different sizes. Finally, we discuss the effectiveness between VBR with SCD and the other service categories, such as UBR and ABR (Available Bit Rate) and GFR (Guaranteed Frame Rate), and conclude that VBR with SCD is one of the most suitable ATM service categories for accommodating best effort traffic.

In this paper, we present the FB (Fair Buffering) mechanism for the efficient support of TCP traffic over UBR connections in the subnet ATM model. We show that both throughput and fairness may be improved by using this congestion control mechanism. The FB mechanism is founded on the observation that the performance of TCP over UBR connections is optimal when the buffer space is allocated in proportion to the connection's bandwidth-delay product. We compare the performance of the existing drop-tail, EB (Equal Buffering) and the proposed FB buffer management schemes, with and without RR (Round-Robin) scheduling, and show through simulation the effectiveness of the proposed FB mechanism when used with the RR scheduling scheme.

In ATM Network, the VBR (Variable Bit Rate) service category is used to accommodate TCP/IP traffic. In an international ATM network with large propagation delay, higher TCP throughput can be obtained by use of window scale option. In order to accommodate TCP traffic with window scale option effectively, it is required to select appropriate values of VBR parameters, i. e. SCR (Sustainable Cell Rate) and MBS (Maximum Burst Size), and to evaluate the impact of UPC (Usage Parameter Control) function on TCP throughput. We have studied those technical issues for the conventional TCP, but the results cannot be applied to TCP traffic with the window scale option due to the TCP terminal performance and the large window size. In this paper, we proposed VBR parameter determination method for TCP with the window scale option and evaluated the values in each condition. These results show that the determined MBS is much smaller than the burst length of TCP segments especially using low performance TCP terminals. Furthermore, we also discuss some experimental results of TCP throughput degradation due to UPC function. It shows that the throughput of TCP with large window size is degraded when the SCR and MBS values used in ATM switch are smaller than the determined values.

QoS of a multimedia interactive satellite communications system, which uses a satellite circuit for downlinks and a terrestrial network for uplinks, has been measured and evaluated under inferior satellite circuit conditions. Our attention in this paper is focused on TCP throughput as the most significant QoS, and experiments and analysis on TCP throughput were performed. The measured results are in good agreement with theoretical values calculated by taking congestion avoidance algorithms of TCP into consideration. As a result, we were able to verify that the TCP throughput declines radically when the low signal quality of a satellite circuit causes packet retransmission to occur. In this case, congestion avoidance works to reduce network congestion. The congestion avoidance increases the value of a retransmission timer and decreases the window size even if the network is not congested. Assuming that throughput deterioration is caused by rain fade, the total time when the throughput declines is less than 4 hours a year in Tokyo, and the results show the system can be put to practical use.

This paper proposes and performs the primary feasibility evaluation on Flow Attribute Notification Protocol (FANP), which is a protocol between neighbor CSR (Cell Switch Router) nodes for the management of cut-through packet forwarding, in order to apply label switching paradigm. In cut-through packet forwarding with label switching, a router doesn't have to perform conventional IP packet processing for the received packets. FANP indicates the mapping information between a data-link connection and a packet flow to the neighbor node. FANP defines two key procedures, i. e. , one is the VCID Notification Procedure, and the other is the Flow-ID Notification Procedure. The VCID Notification Procedure lets the label switching paradigm over the label swapped data-link, such as ATM link, though the other label switch architecture can not work over the label swapped data-link. The primary evaluation of FANP has been performed using the prototype system and with the actual packet statistics. The result shows that, with a corporate backbone level, the label switch router system with FANP would work well.