To offer Web-IMS convergent services, network operators are providing interfaces for Web servers such as Parlay X on top of IMS application servers. However, this type of architecture leads to the building of at least as many interfaces as there are IMS services opened to the Web servers. This architecture also lacks consideration on media/transport processing. This paper presents a unique and generic service-independent functional entity on the IMS network border named Web Session Controller to handle interactions both at the signaling level and at the media/transport level to control Web-IMS communications. A prototype has been developed and is described that allows new types of services to be offered, e.g., push customized Web content to IMS users.

This paper proposes a link bandwidth design method considering failures in an IP network. When we design the link bandwidth, we need to consider rerouted traffic's bandwidth by failures in order to avoid the traffic congestion. If we knew the point-to-point traffic matrix, we could easily obtain the required link bandwidth after failures. However, the support in routers for measuring traffic matrices is poor and it is generally difficult to measure traffic matrices in a large network. The proposed method does not need traffic matrix measurement; instead, it uses link load measurement before failures which is generally easy to measure. Then, the proposed method calculates the logical least upper bound (in mathematical other words, supremum or sup) of the bandwidth variation at each link by failures by using Linear Programming (LP). Simulation results show that the link bandwidth amount designed by the proposed method is close to the real required amount, which needs traffic matrix measurement. That is, the proposed method provides little wasted bandwidth using link load measurement only. Compared with the simple method which also uses only link load measurement, the proposed method achieves 50% reduction in wasted bandwidth.

IP fast reroute techniques have been proposed for achieving fast failure recovery in just a few milliseconds. The basic idea of IP fast reroute is to reduce recovery time after failure by precomputing backup routes. A multiple routing configurations (MRC) algorithm has been proposed for obtaining IP fast reroute. MRC prepares backup configurations, which are used for finding a detour route after failure. On the other hand, requiring too many backup configurations consumes more network resources. It is necessary to recover more traffic flows with fewer backup configurations to ensure scalability. We propose a new backup configuration-creation algorithm for maximizing traffic flows which are fast recovered as much as possible under a limited number of backup configurations. The basic idea is to construct a spanning tree excluding failure links with higher link-loads in each backup configuration. We show that our algorithm has more robust on actual large IP networks.

This paper describes the implementation and evaluation of layer-1 bandwidth-on-demand (BoD) capabilities in the Japanese academic backbone network, called SINET3. The network has a nationwide GMPLS-based layer-1 platform and provides reservation-based and signaling-based BoD services. The overall architecture for providing BoD services including its capabilities, user interface, path calculation, and interface to drive the layer-1 platform are described. Actual examples of BoD services and evaluations of the path setup/release time in the network are also presented.

This paper describes the network design and configurations of the new Japanese academic infrastructure, called SINET3, which provides a rich variety of network services to more than 700 universities and research institutions. Since the start of full-scale operations in June 2007, the network has expanded its services to include multi-layer transfer services (IP, Ethernet, and layer-1), enriched virtual private network services (L3VPN, L2VPN, VPLS, and L1VPN), enhanced QoS services (packet-based and circuit-based), and brand-new layer-1 bandwidth-on-demand (BoD) services. This paper explains how the network provides these various network services on a single network platform by effectively configuring leading-edge networking components, such as high-performance IP routers, layer- 1 switches, and a BoD server. Evaluations of the network design and configurations confirmed that the networking functions were effectively coordinated. The procedures and techniques related to the configuration validation that covered all phases of the network design and construction are also presented.

We have been developing a network visualization technique: a network architecture technology to overlay IP networks over an optical backbone network to improve flexibility and resiliency. Virtual network topology created in the optical backbone network is used as a substrate for a IP network. In order to improve flexibility and resiliency, we need to implement carrier-grade mechanisms for operation, administration, and maintenance. For the IP network, the following management and controls are addressed: traffic measurement, route trace, topology and route optimization, and failure analysis. For the optical backbone network, the impact of the GMPLS control plane is addressed. We also discuss the integrated operation and management of the IP and optical backbone networks.

This paper describes brand-new layer-1 bandwidth on demand (BoD) services implemented in the new Japanese academic backbone network, called SINET3. SINET3 is an advanced converged network that provides multi-layer transfer, enriched VPN, enhanced QoS, and layer-1 BoD services. The layer-1 BoD services are dynamic layer-1 resource allocation services directly triggered by users and artfully achieved on the multi-service platform by using a layer-1 BoD server. This paper first explains how the network accommodates a wide variety of network services by effectively combining leading-edge technologies. The paper next describes the overall mechanism for the dynamic layer-1 path setup on the multi-service platform and details the functions of the BoD server in many aspects. The designs focus on the tangible achievement of these services over a nationwide network composed of 75 layer-1 switches and 12 IP/MPLS routers.

This paper proposes a network design scheme for emerging Layer 1 Virtual Private Network (L1VPN) services. In L1VPN services, customers can request connection setup/deletion within their own VPNs. L1VPN services require a new network design scheme to comply with this customers' ability to reconfigure their VPNs. In order to achieve this, by defining a resource demand model considering flexibility, and designing the network based on this model, amount of resources required can be computed. This paper proposes a new resource demand model by extending conventional resource demand models, and a new network design algorithm. Simulations are conducted to evaluate performance of the proposed network design scheme.

With explosive dissemination of broadband IP access services, like xDSL and FTTH, many service providers seek a chance for deploying new IP based broadband services networks for their customers. Among multiple candidates, content distribution (CD), interactive multi-media (IMM) and virtual private network (VPN) services are the most attractive broadband services for service provider and their customers. The quality of service (QoS) control and traffic engineering (TE) mechanism with strict reliability are essential functions to differentiate these services from conventional best effort IP access service and to make this broadband services network as a new communication infrastructure. Because MPLS already has all of these mechanisms in its protocol architecture, it is assumed that developing broadband IP service network by MPLS technology is the most promising way. But many of these services, such as video broadcasting, video conference and VPN multicast service require not only point-to-point (P2P) transmission capability but also point-to-multipoint (P2MP) transmission capability with much stricter QoS control, more sophisticated TE and more stable reliability control requirement. Therefore, P2MP capability extension is required to conventional MPLS. This paper proposes a scalable multicast MPLS protocol that can establish P2P and P2MP LSP simultaneously. The P2MP extension is realized on RSVP-TE (RFC3209). By introducing tree explicit route object (TERO) and enhancing conventional Path/Resv signaling procedures to enable tree based LSP setup, the proposed protocol can set any kind of P2MP LSPs; cost minimum (Steiner) P2MP LSP, delay-constrained P2MP LSP according to application demands. The proposed protocol also supports dynamic partial tree modification mechanism; Grafting and Pruning to handle dynamic multicast traffic. Therefore, P2MP LSP is modified easily. Several scalability and performance evaluations are conducted in this paper to prove its feasibility. This paper also proposes a multicast constrained shortest path fast (CSPF) mechanism to support dynamic QoS constrained P2MP LSPs setup. We confirmed that proposed schemes are scalable and has enough performance to be applied to future large scale IP service convergence network.

The design and implementation of the transport layer of a native ATM (asynchronous tranfer mode) protocol stack and its embedding into an overall architecture that provides end-to-end quality of service (QOS) is presented. Within this architecture, the typical transport functionalities are enlarged with QOS monitoring and adaptation mechanisms. A QOS-based API (application programming interface) is proposed that shields application programmers from the complexity of QOS management and control. Both unicast and multicast connections supporting interactive multimedia applications are considered.

The service specific connection oriented protocol is now being studied for use as the ATM adaptation layer protocol for connection oriented data communication and UNI/NNI signaling in the B-ISDN. The key feature of this protocol is a numbered poll mechanism that enables status information to be exchanged frequently and that is helpful for selective retransmission for frame loss. This paper examines this mechanism from the viewpoints of polling interval and frame length. Because this protocol was designed to perform selective retransmission for error recovery, some requirements, such as saving buffer, are not met. Two enhancements are therefore proposed: one consists of a Go-back-N-like procedure and nonassured credit, and the other is an SD PDU with a P bit. Guidelines to determine the POLL PDU interval are given.>