In content oriented networks (CON), routers in a network are generally equipped with local cache storages and store incoming contents temporarily. Efficient utilization of total cache storage in networks is one of the most important technical issues in CON, as it can reduce content server load, content download latency and network traffic. Performance of networked cache is reported to strongly depend on both cache decision and content request routing. In this paper, we evaluate several combinations of these two strategies. Especially for routing, we take up off-path cache routing, Breadcrumbs, as one of the content request routing proposals. Our performance evaluation results show that off-path cache routing, Breadcrumbs, suffers low performance with cache decisions which generally has high performance with shortest path routing (SPR), and obtains excellent performance with TERC (Transparent En-Route Cache) which is well-known to have low performance with widely used SPR. Our detailed evaluation results in two network environments, emerging CONs and conventional IP, show these insights hold in both of these two network environments.

Multicast transmission, which can send the same information simultaneously to multiple users, is a key technology in content delivery networks. In this paper, we discuss a new multicast architecture with network coding proposed by Li et al. , which breaks limitation of existing IP multicast in terms of network resource utilization. Network coding based multicast can achieve the max-flow, which is the theoretical upper bound of network resource utilization. However, the max-flow transmission is not always effective and may not be robust against congestion because it maximally uses link capacity of multicast distribution tree. In this paper, we first introduce a load balancing method of network coding as an alternative use to the max-flow transmission. Next, we study the feasibility of network coding based multicast architecture from performance aspect and evaluate the network coding in terms of the max-flow and load balancing with a computer simulation. There has been no evaluation of network coding in practical network environment with packet losses and propagation delay. We also describe required key techniques and technical problems to implement network coding on the current IP networks. Our results will offer valuable insight for designing the future Internet with higher and more effective network utilization.

The window flow control based end-to-end TCP congestion control may cause unfair resource allocation among multiple TCP connections with different RTTs (round trip times) at a bottleneck link. In this paper, in order to improve this unfairness, we propose the active ECN which is an ECN based active queue mechanism (AQM). A bottleneck router with the proposed mechanism marks TCP segments with a probability which depends on the RTT of each connection. By enabling the TCP senders to reduce their transmission rate when their packets are marked, the proposed mechanism can realize the same transmission rate among TCP connections with different RTTs. Furthermore, the active ECN can directly mark ACKs from TCP receivers, while the conventional ECN marks TCP segments coming from the TCP senders. As a result, the queue length distribution at the bottleneck link gets stabilized, because the sender can quickly react to the marking according to variation of the queue length.

Congestion control is a hot topic in named data networking (NDN). Congestion control methods for NDN are classified into two approaches: the rate-based approach and the window-based approach. In the window-based approach, the optimum window size cannot be determined due to the largely changing round-trip time. Therefore, the rate-based approach is considered to be suitable for NDN and has been studied actively. However, there is still room for improvement in the window-based approach because hop-by-hop control in this approach has not been explored. In this paper, we propose a hop-by-hop widow-based congestion control method for NDN (HWCC). The proposed method introduces a window-size control for per-hop Interest transmission using hop-by-hop acknowledgment. In addition, we extend HWCC so that it can support multipath forwarding (M-HWCC) in order to increase the network resources utilization. The simulation results show that both of HWCC and M-HWCC achieve high throughput performance, as well as the max-min fairness, while effectively avoiding congestion.

In this paper, we discuss TCP performance in a wireless overlay network where wireless LANs and cellular networks are integrated. In the overlay network, vertical handover, where a mobile node changes its access link during a session, is one of the most important technologies. When a vertical handover occurs, throughput performance of a TCP flow is degraded due to not only packet losses during the handover, but drastic change of its bandwidth-delay product. In this paper, we propose an ACK-pacing mechanism for TCP congestion control to improve the performance degradation. The proposed system is receiver-driven, so no modification is required to the mechanism of TCP sender. In the proposed system, a TCP receiver adjusts a transmission rate of ACKs according to the relationship between bandwidth-delay products before and after a handover. Since the ACK-clocking mechanism of TCP adjusts the transmission rate of TCP segments, the TCP receiver can seamlessly adjust its congestion window size to the new bandwidth-delay product. Computer simulation results show that the proposed system can improve the TCP performance during the vertical handover.

One of the objectives of ABR traffic control is a fair allocation of transmission rate to every connection. Algorithms which have been proposed can achieve the fair allocation of transmission rate in persistent traffic, even though the connections have different number of hops. In bursty traffic, however, the transmission rate during the burst period, called average burst throughput, may become degraded when the number of hops increases. In this paper, we evaluate the average burst throughput characteristics of ERICA switch algorithm and investigate the effect of the number of hops. We propose a new bandwidth allocation method which can improve the performance degradation by taking the number of hops into account. The effectiveness of our method is evaluated by computer simulation.

In server load balancing where replicated servers are dispersed geographically and accesses from clients are distributed to replicated servers, a way of distributing the accesses from clients to an adequate server plays an important role from the viewpoint of load balancing. In the paper, we propose a new network paradigm for server load balancing using active anycast. In active anycast, an end user only sends its request to group of servers using an anycast address. When this request arrives at an active router, it selects an adequate server from the viewpoint of load balancing and changes the anycast address of a packet to the unicast address of the selected server. The decision which server is the best one from the viewpoint of server load balancing is made by an active router rather than an end user, so active anycast is a network-initiated method. Simulation results show that active anycast can accomplish efficient server load balancing, even when a small part of routers are equipped with active network technology.

A new transfer mode and a switching architecture which can support loss free and no delay jitter service class with shorter switching delay compared with "stop and go queueing scheme" is proposed. This scheme combines ATM scheme with hierarchical STM framing concept.

This paper proposes a traveling maintenance method based on the resource pool concept, as a new network maintenance model. For failure recovery, the proposed method utilizes permissible time that is ensured by shared resource pools. In the proposed method, even if a failure occurs in a communication facility, maintenance staff wait for occurrence of successive failures in other communication facilities during the permissible time instead of immediately tackling the failure. Then, the maintenance staff successively visit the communication facilities that have faulty devices and collectively repair them. Therefore, the proposed method can reduce the amount of time that the maintenance staff take for fault recovery. Furthermore, this paper provides a system design that optimizes the proposed traveling maintenance according to system requirements determined by the design philosophy of telecommunication networks. Through simulation experiments, we show the effectiveness of the proposed method.

The message level performance of error controls in data communication on ATM network is analyzed. Three layers, "a cell"(a unit of transmission), "a block"(a unit of error controls) and "a message"(a unit of transmission of user level) are considered. The error controls treated in this paper are GBN (Go-Back-N) and FEC+GBN. The cell loss process is assumed to be the two state Markov chain considering the cell loss process in ATM networks. Numerical results show that (1) the improvement of the message forwarding delay is saturated in some environments when the interface rate becomes high, (2) FEC is efficient when the burstiness of the cell loss process is small, the message length is large and the interface rate is high.

In this paper, we evaluate the performance of flow control mechanisms for reliable multicast under several retransmission approaches in terms of scalability. The mechanisms examined are a window-based flow control mechanism for ACK-based retransmission approaches and a rate-based flow control mechanism for NAK-based retransmission approaches. Our simulation results show that the NAK-based flow control mechanism has better scalability and higher throughput than the ACK-based flow control mechanism, and the delay incurred by a NAK suppression mechanism does not affect the performance of multicast flow control.

Application-level multicast has drawn a lot of attention as an alternative to IP multicast. In application-level multicast, multicast related features, such as group membership management, packet replication and packet forwarding are implemented at end-hosts instead of routers. The host perceived transmission quality and multicast forwarding responsibility depend on its position in the multicast distribution tree. This nature of application-level multicast motivates selfish members to alter their position by unrightful means to maximize their private benefits. Uncooperative behaviors of these selfish members, i.e. cheating, increase unfairness between selfish members and faithful members. In the context of bulk data distribution, this unfairness between members significantly impacts the receiver throughput. In this paper, to alleviate the negative impact of cheating members, we propose a new tree building protocol which builds dual multicast trees. Our proposed protocol constructs a shortest-widest path tree as the 1st tree. The members having lower position in the 1st tree are located at higher position in the 2nd tree in exchange for their unfairness. To investigate performance of our proposed protocol, it is compared with the existing application-level multicast protocol. Our simulation results show that our protocol outperforms the existing protocol from the view point of throughput and resource utilization against member cheating.

For recent datacenter networks, RDMA (Remote Direct Memory Access) can ease the overhead of the TCP/IP protocol suite. The RoCEv2 (RDMA over Converged Ethernet version 2) standard enables RDMA on widely deployed Ethernet technology. RoCEv2 leverages priority-based flow control (PFC) for realizing the lossless environment required by RDMA. However, PFC is well-known to have the technical weakness of head-of-line blocking. Congestion control for RDMA is a very hot research topic for datacenter networks. In this paper, we propose a novel congestion control algorithm for RoCEv2, TIDD (Timer-based Increase and Delay-based Decrease). TIDD basically combines the timer-based increase of DCQCN and delay-based decrease of TIMELY. Extensive simulation results show that TIDD satisfies the high throughput and low latency required for datacenter networks.

In the paper, we propose a congestion control scheme for reliable multicast communication which enables TCP fairness and prevents a drop-to-zero problem. The proposed congestion control scheme is rate-based one based on NAKs from receivers and cooperatively works with a flow control scheme. The congestion control scheme consists of two components of a rate-based controller and a selection mechanism of a representative. The rate-based controller runs between the sender and the representative and achieves TCP fairness and fast response to losses at the representative. The selection mechanism of the representative allows the sender to select the representative in a scalable manner, in which the sender makes use of NAKs from receivers to select it. In the paper, we also propose the switchover mechanism of the flow and congestion control schemes which enables the sender to use either of them adaptively based on network situations. When the network is congested, the congestion control scheme works to share network resources fairly with competing TCP flows. Otherwise, the flow control scheme works to adapt the transmission rate to the slowest receiver. We verify the performance of our proposed schemes by using computer simulation.

Large parts of the Internet are still incapable of native multicast, and ubiquitous deployment of multicast will take a long time. There are two approaches to provide wide-area multicast service in today's Internet. One is tunneling approach and the other is application-level multicast approach. In this paper, we focus on application-level multicast approach and propose a new scheme which improves the performance penalties of application-level multicast by making use of network support. Because in application-level multicast, endhosts provide multicast functionality instead of routers, application-level multicast delivery tree is inherently less efficient than IP multicast tree. Therefore, in our scheme, the router on the application-level multicast delivery tree alters the tree based on network-level delivery path. We evaluate our scheme with simulation experiment. Our simulation results indicate that our scheme improves the performance of application-level multicast. Further we compare our scheme to the tunneling approach from the viewpoint of transmission performances. The results reveal applicable domains of both approaches.

Content oriented network is expected to be one of the most promising approaches for resolving design concept difference between content oriented network services and location oriented architecture of current network infrastructure. There have been proposed several content oriented network architectures, but research efforts for content oriented networks have just started and technical issues to be resolved are still remained. Because of content oriented feature, content data transmitted in a network can be reused by content requests from other users. Pervasive cache is one of the most important benefits brought by the content oriented network architecture, which forms interconnected caching networks. Caching network is the hottest research area and lots of research activities have been published. This paper surveys recent research activities for caching networks in content oriented networks, with focusing on important factors which affect caching network performance, i.e. content request routing, caching decision, and replacement policy of cache. And this paper also discusses future direction of caching network researches.

CCN/NDN (Content-Centric Networking/Named-Data Networking) is one of the most promising content-oriented network architectures. In CCN/NDN, forwarding information base (FIB) might have multiple entries for a same content name prefix, which means CCN/NDN potentially supports multi-source download. When a content is obtained from multiple sources, the technical knowledge obtained for congestion control in the current Internet cannot be simply applied. This is because in the current Internet, FIB is restricted to have only one entry for each IP address prefix, which causes quite different path feature from CCN/NDN. This paper proposes a new congestion control for CCN/NDN with multi-source content retrieval. The proposed congestion control is composed of end-to-end window flow control and router assisted Interest forwarding control, and enables transmission rate regulation only on a congested branch.

In wireless environment, TCP suffers from significant performance degradation due to bit errors on wireless link and handovers because it responds to all packet losses by invoking congestion control even though packet losses are not related to congestion. Several schemes have been proposed to improve the performance degradation due to each cause. They have been evaluated in a specific network environment where either bit errors or handover occurs, i.e. they do not occur at the same time. In this paper, we reveal the packet recovery mismatch problem in an environment where both of bit errors and handover can cause the performance degradation. We pick up one scenario that TCP traffic is transmitted in the situation that ARQ (Automatic Repeat reQuest) and packet forwarding are implemented together. They are proposed to reduce the influence of bit errors and handover respectively and are natural approaches from the viewpoint of protocol layering. Computer simulation shows that in that scenario both techniques cannot perform efficiently due to interaction of each other. We also propose two buffer control approaches to resolve the packet recovery mismatch problem in our scenario according to applicability of cross-layer operation between layer 2 and layer 3.

Data center network is composed of high-speed Ethernet extended in a limited area of a data center building, so its RTT is extremely small of µsec order. In order to regulate data center network delay large part of which is queuing delay, QCN is proposed for layer 2 congestion control in IEEE 802.1Qau. QCN controls transmission rate of the sender by congestion feedback from a congested switch. QCN adopts probabilistic feedback transmission to reduce the control overhead. When the number of flows through a bottleneck link increases, some flows might receive no feedback even in congestion phase due to probabilistic feedback transmission. In this situation, queue length might be significantly fluctuated. In this paper, we propose a new delay-based congestion detection and control method. Our proposed delay-based congestion control is cooperated with the conventional QCN so as to detect and react congestion not detected by QCN.