Recently, it is required to transfer continuous media over networks without QoS guarantee. In these networks, network congestion will cause transmission delay variance which degrades the quality of continuous media itself. This paper proposes a new protocol using a congestion control with two level rate control in the data transfer level and the coding level. It introduces a TCP-like congestion control mechanism to the rate control of data transfer level, which can detect the QoS change quickly, and adjust the coding rate of continuous media with time interval long enough for its quality. The performance evaluation through software simulation with multiplexing continuous media traffics and TCP traffics shows that the proposed protocol works effectively in the case of network congestion.

We propose a QoS scheme for ad hoc networks by combining TDMA and IEEE 802.11 DCF, and present performance evaluation results of the scheme. In the proposed scheme, the channel time is composed of two different periods, specifically TDMA period and DCF period. The TDMA period provides contention free transmission opportunities for QoS flows, and the DCF period provides contention-based access for best effort or low priority flows. We evaluate the proposed scheme for various numbers of TCP flows and different CBR data rates with QualNet simulator. Simulation results show that the protocol is able to provide an efficient solution for QoS control in ad hoc networks.

Recently, adaptive streaming over information centric network (ICN) has attracted attention. In adaptive streaming over ICN, the bitrate adaptation of the client often overestimates a bitrate for available bandwidth due to congestion because the client implicitly estimates congestion status from the content download procedures of ICN. As a result, streaming overestimated bitrate results in QoE degradation of clients such as cause of a stall time and frequent variation of the bitrate. In this paper, we propose a congestion-aware adaptive streaming over ICN combined with the explicit congestion notification (CAAS with ECN) to avoid QoE degradation. CAAS with ECN encourages explicit feedback of congestion detected in the router on the communication path, and introduces the upper band of the selectable bitrate (bitrate-cap) based on explicit feedback from the router to the bitrate adaptation of the clients. We evaluate the effectiveness of CAAS with ECN for client's QoE degradation due to congestion and behavior on the QoS metrics based on throughput. The simulation experiments show that the bitrate adjustment for all the clients improves QoE degradation and QoE fairness due to effective congestion avoidance.

The Intelligent Network (IN) provides advanced telecommunication services by use of network components distributed over the telecommunication networks. Therefore, the distributed processing technologies are the key to IN and the distribution transparencies which are the purpose of the distributed processing technologies are useful to reduce the difficulties of the IN development. This paper proposes two approaches to apply distributed processing technologies to IN considering on which plane to use the distribution transparencies in the IN conceptual model. The first approach applies distributed processing technologies to the implementation of the functional entities defined in the IN distributed functional plane. The second approach uses distributed processing technologies as the service independent building blocks defined in the IN global functional plane: This paper also presents the results of some typical case studies of the proposed two approaches, the implementation of the functional entities using the remote procedure call and the use of the global naming service as the number translation service independent building block, which show that the distributed processing technologies are highly applicable to IN.

Most of current real time video retrieval systems use video transfer protocols such that servers simply transmit video packets in the same rate as clients play them. If any packets are corrupted during transmission, they will be lost and cannot be recovered by retransmission. In video retrieval systems, however, teh video data are stored in servers and clients can prefetch them prior to playing. So, it might be possible for the video retrieval systems to make corrupted video packets retransmitted before the play-out dead line. But the application of existing reliable protocols causes problems such that, if a packet does not arrive before the dead line due to retransmission, the packets following it will not be delivered to the upper layer even if they have already arrived. In this paper, we discuss how to apply reliable protocols to real time video retrieval systems and propose an new real time video transfer protocol over ATM network, which provides the video data prefetch, the flow control for video buffer, the selective retransmission with skipping function for video packets late for the play-out dead line, and the resynchronization function for video buffer. We have implemented an experimental system using our protocol and evaluated the performance. The results of performance evaluation shows that the proposed protocol decreases the number of unplayed video data largely when transmission errors are inserted in an ATM network.

This paper presents a method of measuring the traffic of IP multicast streaming, such as IPTV, by using IPFIX and PSAMP. The IP multicast streaming service has recently become one of the popular network services, but no IP multicast operation method has been established yet. In particular, traffic measurement of IP multicast streaming encounters two challenges: as monitoring a multicast path tree and QoS measurement. There is no method to monitor them continuously in large-scale networks. Hence, we explore measurement structure suitable for large-scale networks, and then using IPFIX and PSAMP, we propose an efficient a network-level quality measurement method for IPTV and a method to extract multicast path tree data. In addition, we demonstrate the feasibility of the measurement method by evaluating a prototype system.

Resulting from the spread of Mobile Internet, the mobile communication with QoS guarantee will be required in order to realize mobile video interactions. So far, there are some studies focusing on QoS Mobile IP communication, but they require backbone routers to maintain per-flow QoS information for all individual Mobile Nodes. So these approaches suffer from the lack of scalability. Against them, we are developing an approach which the per-flow QoS information is maintained only by Mobile IP agents such as the Home Agent and the Foreign Agent. We have adopted a hierarchical method with MPLS which MPLS paths with large bandwidth are introduced between Mobile IP related nodes, and a per-flow path with small bandwidth called Pathlet is established for individual communication between a Mobile Node and a Fixed Host. The maintenance of Pathlets is only performed by Home Agent, Foreign Agent and Fixed Host, and the network backbone MPLS routers only take care of MPLS paths with large bandwidth. In the simulation, we compare our scheme with conventional scheme by observing the total number of entries managed by routers and bandwidth prepared at individual links.

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.

In this paper, we propose a new routing protocol for named data networking applied to ad hoc networks. We suppose a type of ad hoc networks that advertise versatile information in public spaces such as shopping mall and museum. In this kind of networks, information providers prepare fixed nodes, and users are equipped with mobile terminals. So, we adopt a hybrid approach where a proactive routing is used in the producer side network and a reactive routing is used in the consumer side network. Another feature of the proposed protocol is that only the name prefix advertisement is focused on in the proactive routing. The result of performance evaluation focusing on the communication overhead shows that our proposal has a moderate overhead both for routing control messages and Interest packets compared with some of conventional NDN based ad hoc routing mechanisms proposed so far.

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 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.

Recently, information centric network (ICN) has attracted attention because cached content delivery from router's cache storage improves quality of service (QoS) by reducing redundant traffic. Then, adaptive video streaming is applied to ICN to improve client's quality of experience (QoE). However, in the previous approaches for the cache control, the router implicitly caches the content requested by a user for the other users who may request the same content subsequently. As a result, these approaches are not able to use the cache effectively to improve client's QoE because the cached contents are not always requested by the other users. In addition, since the previous cache control does not consider network congestion state, the adaptive bitrate (ABR) algorithm works incorrectly and causes congestion, and then QoE degrades due to unnecessary congestion. In this paper, we propose an explicit cache placement notification for congestion-aware adaptive video streaming over ICN (CASwECPN) to mitigate congestion. CASwECPN encourages explicit feedback according to the congestion detection in the router on the communication path. While congestion is detected, the router caches the requested content to its cache storage and explicitly notifies the client that the requested content is cached (explicit cache placement and notification) to mitigate congestion quickly. Then the client retrieve the explicitly cached content in the router detecting congestion according to the general procedures of ICN. The simulation experiments show that CASwECPN improves both QoS and client's QoE in adaptive video streaming that adjusts the bitrate adaptively every video segment download. As a result, CASwECPN effectively uses router's cache storage as compared to the conventional cache control policies.

Currently, NDN-based VANETs protocols have several problems with packet overhead of rebroadcasting, control packet, and the accuracy of next-hop selection due to the dynamic topology. To deal with these problems in this paper, we propose a robust and lightweight forwarding protocol in Vehicular ad-hoc Named Data Networking. The concept of our forwarding protocol is adopting a packet-free approach. A vehicle collects its neighbor's visual identification by a pair of cameras (front and rear) to assign a unique visual ID for each node. Based on these IDs, we construct a hop-by-hop FIB-based forwarding strategy effectively. Furthermore, the Face duplication [1] in the wireless environment causes an all-broadcast problem. We add the visual information to Face to distinguish the incoming and outgoing Face to prevent broadcast-storm and make FIB and PIT work more accurate and efficiently. The performance evaluation results focusing on the communication overhead show that our proposal has better results in overall network traffic costs and Interest satisfaction ratio than previous works.

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.

Due to vehicle movement and lossy wireless channels, providing a reliable and efficient multi-hop broadcast service in vehicular ad hoc networks (VANETs) is a well-known challenging problem. In this paper, we propose BR-NB (broadcast with neighbor information), a fuzzy logic based multi-hop broadcast protocol for VANETs. BR-NB achieves a low overhead by using only a subset of neighbor nodes to relay data packets. For the relay node selection, BR-NB jointly considers multiple metrics of the inter-vehicle distance, vehicle mobility and link quality by employing fuzzy logic. Since the expected coverage and vehicle mobility are inferred from the two-hop neighbor information which can be acquired from the hello message exchange, BR-NB is independent of position information. BR-NB provides a practical and portable solution for broadcast services in VANETs. We use computer simulations and real-world experiments to evaluate the performance of BR-NB.

The self-organizing nature of ad hoc networks is a good aspect in that terminals are not dependent on any infrastructure, that is, networks can be formed with decentralized and autonomous manner according to communication demand. However, this characteristic might affect the performance in terms of stability, reliability and so forth. Moreover, ad hoc networks face a scalability problem, which arise when the number of terminals in a network increases or a physical network domain expands, due to the network capacity limitation caused by the decentralized and the autonomous manner. Regarding this problem, some hierarchical and cluster-based routings have been proposed to effectively manage the networks. In this paper, we apply the concept of hierarchical routing and clustering to opportunistic routing, which can forward packets without using any pre-established path to achieve a path diversity gain with greater reachability. The simulation results show that the proposed method can achieve 11% higher reliability with a reasonable end-to-end delay in dense environments and 30% higher in large-scale networks.

Cognitive radio ad hoc networks can be used to solve the problems of limited available spectrum and inefficient spectrum usage by adaptively changing their transmission parameters. Routing protocol design has a significant impact on the network performance. However, an efficient protocol that takes account of primary user flows and the long-term channel assignment issue in route selection is still missing. In this paper, we propose AODV-cog, a cognitive routing protocol for CSMA/CA ad hoc networks based on AODV. AODV-cog chooses a route by considering the effect on the primary users, available channel bandwidth and link reliability. AODV-cog also takes account of future channel utilization which is an important but underexplored issue. AODV-cog switches channels for secondary user flows when network congestion occurs. We use theoretical analysis and computer simulations to show the advantage of AODV-cog over existing alternatives.

Vehicular ad hoc networks have been attracting the interest of both academic and industrial communities on account of their potential role in Intelligent Transportation Systems (ITS). However, due to vehicle movement and fading in wireless communications, providing a reliable and efficient multi-hop broadcast service in vehicular ad hoc networks is still an open research topic. In this paper, we propose FUZZBR (FUZZy BRoadcast), a fuzzy logic based multi-hop broadcast protocol for information dissemination in vehicular ad hoc networks. FUZZBR has low message overhead since it uses only a subset of neighbor nodes to relay data messages. In the relay node selection, FUZZBR jointly considers multiple metrics of inter-vehicle distance, node mobility and signal strength by employing the fuzzy logic. FUZZBR also uses a lightweight retransmission mechanism to retransmit a packet when a relay fails. We use computer simulations to evaluate the performance of FUZZBR.

It is widely recognized that IP-based mobile network will be a dominant trend. For mobile IP networks, the address starvation problem and scalable mobility management for mobile nodes are important issues. In order to cope with these issues, we propose an approach to realize mobile IP network supporting private addresses for mobile nodes. Our approach introduces regional registration of mobile nodes (Hierarchical Mobile IPv4) and coordinates NAT and DNS functions with the Mobile IP protocol. It enables a mobile node to be assigned a global address temporally in a visited network and to accept a call initiated by a correspondent node connected to the global IP network. This paper describes the detailed design of our approach and the implementation of proposed procedures based on the Mobile IPv4 software developed by the CMU Monarch project.

Along with the improvement of micro processors and local area networks, a distributed system becomes useful to realize a telecommunication system. It has potential advantage to achieve both high performance and high reliability. However, the design of a distributed system tends to be more complicated compared to a conventional centralized system. For the purpose of the standardization of distributed processing, ISO and ITU-T study the Open Distributed Processing (ODP) and are currently standardizing the Basic Reference Model of ODP (RM-ODP). To avoid dealing with the complexity of distributed systems, RM-ODP defines five viewpoints. The viewpoint approach of RM-ODP is proposed as a framework for the design of a distributed system. Although some previous works give the design methods of distributed systems based on the ODP viewpoint approach, the detailed design method has not been fully specified or all of the five viewpoints are not taken into account. In this paper, we describe a detailed design method for a distributed telecommunication system based on the ODP viewpoint approach. The method applies the five viewpoints to the three phases of design of a distributed system, that is, requirement analysis, functional design and detailed design phase. It clarifies what specifications for the target system should be made from the individual viewpoints and how the specifications are related each other. It also takes account of the platform which provides the distribution support, and gives the design method for both the platform and the application specific functions on the platform. The design method is examined by applying it to the design of a distributed MHS system supporting X.400 series protocols. In this example, the remote procedure call based on the client-server model is selected as the base of the platform. The result shows that our method is useful to simplify the complexity of the design for a distributed telecommunication system.