An optimal selection criterion of the modulation and coding scheme (MCS) for maximizing spectral efficiency is proposed in consideration of the signaling overhead of mobile WiMAX systems with a hybrid automatic repeat request mechanism. A base station informs users about the resource assignments in each frame, and the allocation process generates a substantial signaling overhead, which influences the system throughput. However, the signaling overhead was ignored in previous MCS selection criteria. In this letter, the spectral efficiency is estimated on the basis of the signaling overhead and the number of transmissions. The performance of the proposed MCS selection criterion is evaluated in terms of the spectral efficiency in the mobile WiMAX system, with and without persistent allocation.

IP-- is proposed as an Internet Protocol suitable for optical packet networking. As optical routers require much faster control than electric ones and lack of optical buffers other than those by fiber delay lines requires fixed time control, Internet Protocols must be at least as simple as IPv4 and much simpler than IPv6. IP-- also addresses issues of IP address space exhaustion and IP routing table explosion.

In this paper, we analyze the extended real-time Polling Service (ertPS) algorithm in IEEE 802.16e systems, which is designed to support Voice-over-Internet-Protocol (VoIP) services with data packets of various sizes and silence suppression. The analysis uses a two-dimensional Markov Chain, where the grant size and the voice packet state are considered, and an approximation formula for the total throughput in the ertPS algorithm is derived. Next, to improve the performance of the ertPS algorithm, we propose an enhanced uplink resource allocation algorithm, called the e 2rtPS algorithm, for VoIP services in IEEE 802.16e systems. The e 2rtPS algorithm considers the queue status information and tries to alleviate the queue congestion as soon as possible by using remaining network resources. Numerical results are provided to show the accuracy of the approximation analysis for the ertPS algorithm and to verify the effectiveness of the e 2rtPS algorithm.

We have developed a novel railway signal control system that operates as a distributed system. It consists of a central control unit (called LC) and terminal devices (called FC) that are distributed at the railroad wayside and operate signal devices. The Internet technologies and optical LAN technologies have been used as communication methods between the LC and the FCs. While handling enormous amount of electric cables may cause human errors, the system is expected to reduce troubles of the current signal system at construction works thanks to the Internet technologies. The FC is a distributed terminal device that has its own processor and placed at the railroad wayside to control the field signal devices. The LC is a centralized computer device that has software arranged by the function of the field devices. An optical network system and multiple communication paths between the LC and the FCs realize durable transmissions. Moreover, the assure performance of controls and transmissions have been investigated, and the autonomous distributed signal control system is also discussed as the next steps of the system.

In this paper, we propose a protocol for multicast communication called Multicast Datagram Transfer Protocol (MDTP) to provide multicast for video broadcasting service on the Internet. MDTP is a one-to-many multicast communication protocol, which is constructed based on IPv4 unicast protocol by utilizing IP Router Alert Option, and it uses unicast addressing and unicast routing protocol. A mechanism is presented to allow a router to remove identical video stream, to duplicate a video stream, and to forward each copy of the duplicated video stream to its destinations. Ordinary IP routers that do not support MDTP will treat the MDTP packets as normal unicast packets. Hence, gradual deployment is possible without tunneling technique. With a delegation mechanism, MDTP router is also able to handle request from clients, and serve the requested video stream. The simulation results show that the average bandwidth usage of MDTP is close to the average bandwidth usage of IP multicast. MDTP also has greater efficiency than XCAST, and its efficiency becomes significant for a large number of clients.

This paper proposes a new FEC scheme using backup channel to send redundant information instead of piggybacking the main packet. This is particularly applicable to the modern IP networks which are distributed all over the world. In this method only one source coder for both the main and the redundant payload is used to reduce the overall computational complexity. The Gilbert loss model (GLM) is employed to verify the improvement of the packet loss probability in this new method compared with that in a single path FEC scheme. It is shown, through simulation results that using our proposed backup channel can considerably improve the packet loss and delay performance of the VoIP networks.

Real-world IP networks are heterogeneous in terms of server and link capacities. A sophisticated and comprehensive load balancing method is essential if we are to avoid congestion in the servers and links of heterogeneous networks. If such a method is not available, network throughput is limited by bottleneck servers or links. This paper proposes an anycast technique that achieves load balancing under heterogeneity. The proposed method well suits implementation on active networks. By taking advantage of the processing ability provided by active nodes, the method can decide packet routes flexibly on the basis of various criteria to realize a variety of load balancing schemes. Some of these schemes can successfully prevent the congestion of heterogeneous networks by tackling bottlenecks in both server and link capacities. The method is also advantageous given its light control load even when using many mirrored servers. Computer simulations confirm the effectiveness of these features.

Software applications for the transfer of money or personal information are increasingly common on the Internet. These applications require user authentication for confirming legitimate users. One-time password authentication methods risk a stolen-verifier problem or other steal attacks because the authentication on the Internet server stores the user's verifiers and secret keys. The SAS-2 (Simple And Secure password authentication protocol, ver.2) and the ROSI (RObust and SImple password authentication protocol) are secure password authentication protocols. However, we have found attacks on SAS-2 and ROSI. Here, we propose a new method which eliminates such problems without increasing the processing load and can perform high security level same as S/Key systems without resetting the verifier.

In this paper, we propose a speech coding translation scheme by transferring coding parameters between GSM half rate and G.729 coders. Compared to the conventional decode-then-encode (DTE) scheme, the proposed parameter conversions provide speech interoperability between mobile and IP networks with reducing computational complexity and coding delay. Simulation results show that the proposed methods can reduce about 30% computational load and coding delay acquired in the target encoders and achieve almost imperceptible degradation in performance.

Fast packet switches for variable-size packets have become an everyday necessity with the rapid growth in the volume of Internet traffic. Such switches can be designed in two different ways, either by segmenting packets into smaller fixed-size cells and designing packet switches for such cells, or by designing generic packet switches for variable-size packets, where packet segmentation and reassembly can be omitted. The second option is investigated in this paper. The synchronous operation mode with time-limited bulk service is selected. The switching fabric is assumed to be internally non-blocking and provided with input queues. A previous maximum switch throughput analysis has been done under the assumption that the length of the time slot is fixed set to its minimum allowed value (Tmin). In this work, a so-called time-slot stretch factor (SF) is introduced. The actual time-slot length is determined by multiplying Tmin with the SF, where SF. Next, a so-called Internet traffic-source model is proposed based on findings from real IP traffic measurements. The performance implications of the proposed time-slot length modification are analyzed by discrete-event computer simulation. The maximum switch throughput is increased by increasing the SF value, e.g. for uniform packet size distribution and SF=10, the maximum switch throughput is increased from 75% to 97%. The influence of the traffic-source characteristics on the maximum switch throughput is decreased when SF value is increased. In order to prevent any possible throughput degradations, it is advisable to use integer SF values. Packet delay analysis has revealed that by increasing the SF value, the mean packet delay is also increased. Nevertheless, it is shown that the number of switch input and output ports is the most important factor to be considered when packet delay is at stake. Service class differentiation inside investigated packet switch is possible and is not affected by the increasing SF value. Such a packet switch is suitable for implementation in wide area networks, due to high transmission speeds and the small number of switch ports.

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.

In this article, we first discuss QoS metrics of the data networks, followed by raising the challenging problems for the next-generation Internet with high-performance and high-quality. We then discuss how the WDM technology can be incorporated for resolving those problems. Several research issues for the IP over WDM networks are also identified.

A wavelength division multiplexing (WDM) optical path-based Internet protocol (IP) backbone network is proposed as a cost-effective way of realizing robust IP-over-photonic systems. The WDM optical path is based on WDM transmission and wavelength routing. Between end-to-end IP backbone routers, the WDM optical path, a fat and robust optical pipe, is defined across photonic transport systems (PTS's). Tera-bit class PTS's will be required for the future IP backbone network and this level of performance is achievable. Optical layer routing is done at intermediate nodes, so the electrical packet-by-packet routing required by existing systems is eliminated. An optical signal format that permits cost-effective IP packet transmission is presented. WDM optical paths directly accommodate the IP packets via layer-2 frames. The cost-effectiveness of the proposed system, especially for heavy traffic, is demonstrated from the viewpoint of the overall network traffic transport capability and network node cost. The proposed system is as robust as existing systems; e. g. fault/degradation localization mechanism and optical layer network protection one are implemented. Thus the proposed IP-over-photonic system will create cost-effective and robust IP backbone networks.

A wavelength division multiplexing (WDM) optical path-based Internet protocol (IP) backbone network is proposed as a cost-effective way of realizing robust IP-over-photonic systems. The WDM optical path is based on WDM transmission and wavelength routing. Between end-to-end IP backbone routers, the WDM optical path, a fat and robust optical pipe, is defined across photonic transport systems (PTS's). Tera-bit class PTS's will be required for the future IP backbone network and this level of performance is achievable. Optical layer routing is done at intermediate nodes, so the electrical packet-by-packet routing required by existing systems is eliminated. An optical signal format that permits cost-effective IP packet transmission is presented. WDM optical paths directly accommodate the IP packets via layer-2 frames. The cost-effectiveness of the proposed system, especially for heavy traffic, is demonstrated from the viewpoint of the overall network traffic transport capability and network node cost. The proposed system is as robust as existing systems; e. g. fault/degradation localization mechanism and optical layer network protection one are implemented. Thus the proposed IP-over-photonic system will create cost-effective and robust IP backbone networks.

This paper proposes a protocol to support mobility in the Internet with a new encapsulation technique. IP-squared (IP2). A basic idea to support mobility is as follows; 1) to define two IP addresses for each mobile host that indicate the host itself and its geographical location (logical and geographical identifiers), 2) to maintain an association of the logical identifier with the geographical identifier and 3) to continue communications between hosts by converting their logical identifiers to the corresponding geographical identifiers dynamically wherever they migrate. The association is called mobility binding. A goal of IP2 is to propose a mobility support feature which can simultaneously realize efficient routing paths to mobile hosts and less control traffics to maintain the mobility bindings into the current Internet Protocol without any modifications to both the conventional protocols and nodes. IP2 forms the efficient routing paths by enabling intermediate nodes to process the encapsulated datagrams. The key technique for this feature is a new header detection algorithm based on CRC checksum calculation and an effective usage of a header structure. Percentage of efficient routing paths can be adaptively controlled, depending on dispositions of the nodes which can en -and de capsulate datagrams appropriately based on the detection algorithm and the mobility bindings. The mobility binding must be updated whenever a mobile host migrates to another network. IP2 adopts an updating scheme combining self refreshment and on demand updating modes with taking a mechanism to form the efficient routing paths into considerations. It is shown that IP2 can acheive both an efficient routing path formation and a low traffic for mobility binding maintenance through analytical evaluations.