Recently routing protocols for QoS aware multicast are actively studied, but there are few studies focusing on the scalability of link state advertisement when the available bandwidth of a link is updated along with the QoS aware multicast tree construction. This paper proposes a new QoS aware multicast routing protocol that is scalable in terms of the link state advertisement exchange. Our protocol has the following features; (1) A multicast network is divided into domains, and the advertisement of information on links within a domain is limited within the domain. (2) Among the border multicast routers, only the link state information of inter-domain links is advertised. As a result, the number of link state advertisement messages will be drastically reduced. (3) When a multicast tree spreads over multiple domains, the tree construction needs to be performed without information on links in other domains, and it is possible that the construction may fail. In order to cope with this problem, the crank back mechanism of a tree construction is introduced. This paper describes the detailed procedures and the message formats of our protocol. It also describes the evaluation of the number of exchanged link state advertisement messages and shows that our protocol can reduce the number comparing with the conventional protocols.

In the Mobile IP handoff procedure, mobile node movement is detected from advertisements of foreign agents that differ from previously received advertisement and the new "care-of" address is registered with the home agent. However, user packets are not forwarded to the new foreign agent until a registration is completed and this interruption may degrade the quality of service especially in real-time applications such as audio and video, or may lower the TCP throughput due to retransmission timeout. To tackle these issues, we propose a new low latency handoff method, where access points used in a wireless LAN environment and a dedicated MAC bridge are jointly used to alleviate packet loss without altering the Mobile IP specifications. In this paper, we present design architecture of the proposed method and evaluate its performance in an actual network environment to verify the effectiveness of our approach.

Although TCP is widely used in the Internet, new specifications are still proposed and implemented. In the circumstance above, it is highly possible that some errors are detected on the communication between new and old implementations. Several test tools were developed so far. However, they do not have enough functions to allow test operators to modify test sequences suitable for their test purposes. We have developed a TCP tester which generates test sequence using test scenario. The tester performs exceptional TCP protocol behavior only when the condition specified in the test scenario is satisfied. Otherwise, it performs normal TCP behavior. The tester is implemented by modifying TCP module of NetBSD with SACK code developed by Pittsburgh Supercomputing Center. We have also evaluated implementations of congestion control and SACK algorithms using the tester.

With the increase of the number of wireless sensing or metering devices, the collection of sensing data using wireless communication becomes an important part of a smart grid system. Cognitive radio technology can be used to facilitate the deployment of smart grid systems. In this paper, we propose a data collection and dissemination framework for cognitive radio smart grid systems to fully utilize wireless resources while maintaining a reliably connected and efficient topology for each channel. In the proposed framework, each sensor node selects a channel considering the primary user (PU) channel utilization and network connectivity. In this way, the data collection and dissemination can be performed with a high reliability and short delay while avoiding a harmful effect on primary users. We use computer simulations to evaluate the proposed framework.

In Vehicular Ad hoc Networks (VANETs), general purpose ad hoc routing protocols such as AODV cannot work efficiently due to the frequent changes in network topology caused by vehicle movement. This paper proposes a VANET routing protocol QLAODV (Q-Learning AODV) which suits unicast applications in high mobility scenarios. QLAODV is a distributed reinforcement learning routing protocol, which uses a Q-Learning algorithm to infer network state information and uses unicast control packets to check the path availability in a real time manner in order to allow Q-Learning to work efficiently in a highly dynamic network environment. QLAODV is favored by its dynamic route change mechanism, which makes it capable of reacting quickly to network topology changes. We present an analysis of the performance of QLAODV by simulation using different mobility models. The simulation results show that QLAODV can efficiently handle unicast applications in VANETs.

In order to make the implementation of network components flexible and cost effective, it is required to use widely available technologies as the implementation platform. The distributed operating systems can be adopted as such a platform, because they allow to implement a network component using multiple computers connected through a local area network. In this paper, we focus on the Intelligent Network (IN) whose network components are modelled as Functional Entities (FEs), and describe an implementation method of FEs using distributed operating systems. Our method is summarized as follows: The remote procedure call (RPC) is used for the access transparent inter-process communication. The lightweight process mechanism is used for handling concurrent requests. CCF/SSF (Call Control Function/Service Control Function) and SDF (Service Data Function) are implemented as an SSF server and an SDF server, respectively. SCF (Service Control Function) is composed of a Service Dispatcher and a set of Service Executors. The Service Dispatcher accepts all the requests for IN call processing and dispatches them to appropriate Service Executors. Service Executors are created for the individual IN services and execute the service logics. SDF server and Service Executor may be replicated for load partitioning.This paper has also described the implementation of experimental system supporting "Freephone" service based on our method, and showed the performance evaluation of the experimental system in terms of the real-time and concurrent call processing of IN services. We used Mach and SUN OS as a platform for implementing the servers for FEs. The experimental system using four workstations shows that it can handle up to 170IN calls in one second with the additional response time of less than 200msec, which is small enough compared with the response time for the basic connection control. Those results prove that our method is feasible for implementing practical FEs.

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.