As we are moving toward next generation wireless networks, we are facing the integration of heterogeneous access networks. The main challenge is to provide mobile users moving freely across different radio access technologies with satisfactory quality of services for a variety of applications. Consequently, the seamless roaming over heterogeneous networks is an important concern. To minimize the disruption to the ongoing session when a mobile user is moving from one access network to another, we propose a framework that integrates IEEE 802.11 WLANs and IEEE 802.16 WMANs based on the IEEE 802.21, so-called Media Independent Handover (MIH), to facilitate both homogeneous and heterogeneous handovers. Both numerical analysis and simulation results show that seamless roaming between WLAN and WMAN can be achieved and much better performance can be obtained compared with the IEEE 802.21 standard approach.

Current IP network has become the dominant paradigm for all networking environments. The significant cause of packet losses in such heterogenous networks is no longer limited to network congestion. Traditional TCP interprets every packet loss as caused by congestion which may be not the case in the current Internet. Misinterpretation of wireless random loss as an indication of network congestion results in TCP slowing down its sending rate unnecessarily. In this paper, we propose a new variant of TCP Vegas named RedVegas. By using the innate nature of Vegas and congestion indications marked by routers, RedVegas may detect random packet losses precisely. Through the packet loss differentiation, RedVegas reacts appropriately to the losses, and therefore the throughput of connection over heterogeneous networks can be significantly improved.

The differentiated services (Diffserv) architecture is a potential solution for providing quality of service (QoS) on the Internet. Most existing studies focus on providing service differentiation among few service classes. In this paper, we propose an approach which can achieve per-flow weighted fair rate allocation in a differentiated services network. Following the design philosophy of the Diffserv model, in the proposed approach core routers do not need to keep per-flow information. An edge router adjusts the transmission rate of a flow based on the feedback carried on control packets, which are inserted by the ingress edge router and returned by the egress edge router. Core routers periodically estimate the fair share rate of each virtual flow and mark the results in control packets. We use both simulations and analysis to evaluate the performance of the proposed approach. The analytical results show that our approach allows a system to converge to weighted fair rate allocations in limited time. Through the simulation results, we can further validate the analytical results, and demonstrate that better throughput can be achieved.

Packet classification is essential for supporting advanced network services such as firewalls, quality-of-service (QoS), virtual private networks (VPN), and policy-based routing. The rules that routers use to classify packets are called packet filters. If two or more filters overlap, a conflict occurs and leads to ambiguity in packet classification. This study proposes an algorithm that can efficiently detect and resolve filter conflicts using tuple based search. The time complexity of the proposed algorithm is O(nW +s), and the space complexity is O(nW), where n is the number of filters, W is the number of bits in a header field, and s is the number of conflicts. This study uses the synthetic filter databases generated by Class-Bench to evaluate the proposed algorithm. Simulation results show that the proposed algorithm can achieve better performance than existing conflict detection algorithms both in time and space, particularly for databases with large numbers of conflicts.

In the infrastructure-less disaster environment, the application of the peer-to-peer (P2P) group conference over mobile ad hoc network (MANET) can be used to communicate with each other when the rescue crews search the survivors but work separately. However, there still are several problems of in-time multimedia delivery in P2P-MANET: (1) MANET mobility influences the maintenance of P2P overlay. (2) P2P overlay is not proximal to MANET topology, this leads to the inefficient streaming delivery. (3) The unreliable wireless connection leads to the difficulty of multi-source P2P group conferencing. Therefore, P2P conferencing cannot work well on MANET. To overcome the above disadvantages, in this paper, we present a cross-layer P2P group conferencing mechanism over MANET, called RING (Real-time Intercommunication Network Gossip). The RING uses the ring overlay to manage peers and utilizes the cross-layer mechanism to force the ring overlay to be proximal to MANET topology. Therefore, RING can lead efficient in-time multimedia streaming delivery. On the other hand, the ring overlay can deal with peer joining/leaving fast and simply, and improves the delivery efficiency with the minimum signaling overhead. Through mathematical theory and a series of experiments, we demonstrate that RING is workable and it can shorten the source-to-end delay with minimal signaling overhead.

In this paper we present two traffic control approaches, a circuit emulation traffic control (CETC) and an adaptive priority traffic control (APTC) for supporting voice services in ATM networks. Most voice services can be handled as CBR traffic, this causes a lot of wasted bandwidth. Sending voice through VBR (variable bit rate) may be a better alternative, because it allows the network to allocate voice bandwidth on demand. In CETC, the service discipline guarantees the quality of service (QOS) for voice circuits. Through mathematical analysis, we show that CETC features an adequate performance in delay-jitter. Moreover, it is feasible in implementation. We also present an APTC approach which uses a dynamic buffer allocation scheme to adjust the buffer size based on the real traffic need, as well as employs an adaptive priority queuing technique to handle various delay requirements for VBR voice traffic. It provides an adequate QOS for voice circuits in addition to improving the multiplexing gain. Simulation results show that voice traffic get satisfied delay performance using our approaches. It may fulfill the emerging needs of voice service over ATM networks.

Differentiated services (Diffserv) model is one of the possible solutions for providing quality of service (QoS) on the Internet. Most existing approaches assume that the packet loss is an indication of network congestion and thus reduce the sending rates of sources. For wireless networks, the assumption is not proper since packet losses may be caused by other reasons, such as fading and interference of the signal. Therefore, these approaches do not work well in wireless networks. In this paper, we propose an approach which is able to provide service differentiation in wireless environments. In our approach, the rate share of a connection is determined by the associated weight. By keeping a proper amount of extra data in the network, the proposed approach can achieve weighted proportional fairness, which can provide selective QoS without any particular support from the network. We use the ns simulator to evaluate our approach. Simulation results show the validity of the proposed approach.

A distributed congestion avoidance scheme, the End-to-End Rate Control Algorithm (EERCA), has been proposed for Available Bit Rate (ABR) service. In this work, we enhance EERCA by replacing the Queue Occupancy Reduction with the queuing delay control (QDC), which manipulates the queue occupancy more efficiently with less functional complexity. Furthermore, it alleviates per-VC accounting requirement and thus reduces the complexity of a switch. From both the analysis and the simulation, it shows that the proposed virtual queue occupancy precisely reflects the variation of the actual queue occupancy. This enhanced EERCA features higher efficiency and more stable queue occupancy than the original algorithm.

Due to the fast advances in wireless networking technology, there is an increasing number of hosts using TCP/IP to connect to the Internet via wireless links. However, it is known that TCP performs poorly on paths with wireless links. This paper presents an approach to address the problem. In the proposed approach, a sender controls the size of transmitted packets and observes the number of losses for each of the controlled packet sizes. It then estimates the number of congestion losses and random losses separately. The results are used to decide whether to reduce the window size or not when a packet loss is detected. We use ns-2 simulator to evaluate the proposed approach. The simulation results show that the proposed approach is able to achieve between 25% and 150% better throughput than FACK under the byte-error rate range of 510-5 to 2010-5.

A critical design issue of Transmission Control Protocol (TCP) is its congestion control that allows the protocol to adjust the end-to-end communication rate based on the detection of packet loss. However, TCP congestion control may function poorly during its slow start and congestion avoidance phases. This is because TCP sends bursts of packets with the fast window increase and the ACK-clock based transmission in slow start, and respond slowly with large congestion windows especially in high bandwidth-delay product (BDP) networks during congestion avoidance. In this article, we propose an improved version of TCP, TCP-Ho, that uses an efficient congestion window control algorithm for a TCP source. According to the estimated available bandwidth and measured round-trip times (RTTs), the proposed algorithm adjusts the congestion window size with a rate between exponential growth and linear growth intelligently. Our extensive simulation results show that TCP-Ho significantly improves the performance of connections as well as remaining fair and stable when the BDP increases. Furthermore, it is feasible to implement because only sending part needs to be modified.

Establishing peer-to-peer (P2P) live streaming for mobile ad hoc network (MANET) requires an efficient scheme to deliver the real-time data in the infrastructure-less disaster environment. However, P2P membership management is difficult in the dynamic mobility and resource limited MANET. In this paper, we present a cross-layer design for P2P-MANET which integrates P2P DHT-based routing protocol and IPv6 routing protocol. Therefore, the proposed scheme can manage and recover the P2P overlay as well as selecting efficient routing path to multicast video streaming. The simulation results demonstrate that the proposed scheme performs relatively better than the layered approach or the off-the-shelf design in terms of the playback continuity and signaling overhead.

Although ATM networks support various traffic requirements, but many data applications are unable to precisely specify traffic parameters such as bit rate. These applications generally require a dynamic share of the available bandwidth among all active connections, they are called available-bit-rate (ABR) service. Due to bursty and unpredictable pattern of an ABR data stream, its traffic control is more challenging than other services. In this paper, we present an improved ABR traffic control approach, called Offset Proportional Rate Control Algorithm (OPRCA). The proposed approach achieves high link utilization, low delay and weighted fair sharing among contenting sources according to the predefined OPR. The implementation is much simpler than that of existing schemes. OPRCA combines an end-to-end rate control with link-by-link feedback control, and employs a buffering scheme that avoids Head-of-Line (HOL) blocking. It can dynamically regulate the transmission rate of source traffic and maintain the real fairness among all active connections. Simulation results have shown the effectiveness of OPRCA in several performance aspects.

In this work, we propose an End-to-End Rate Control Approach (EERCA) for congestion avoidance in Available Bit Rate (ABR) service on Asynchronous Transmit Mode (ATM) networks. In our approach, the network estimates the number of cells stored in the switch for each VC. The source generates a specific traffic pattern, then a proper explicit rate can be derived based on the received traffic pattern at the destination. This approach is designed to reduce the rate calculation effort in the switch as well as to avoid the complexity in setting the monitoring-interval. EERCA features higher efficiency, higher utilization, more stable queue occupancy, shorter transient response time, and better fairness compared with existed schemes.

In the previous work, Lampson et al. proposed an IP lookup algorithm which performs binary search on prefixes (BSP). The algorithm is attractive, even for IPv6, because of its bounded worst-case memory requirement. To achieve fast forwarding, it may need to slow down the insertion speed. Although this can be justified, the routing-table reconstruction in BSP is too time-consuming to handle the frequent route updates. In this work, we propose a fast forwarding-table construction algorithm which can accomplish more than 4,000 route updates per second. Moreover, it is simple enough to fulfill the need of fast packet forwarding. With the enhanced multiway search tree, we further reduced the depth of the tree and eliminated the pointer storage; this reduces the forwarding table size and shortens the lookup time.