In this paper, we propose an energy and cost efficient WDM/OFDMA PON system architecture, referred to as NEWOPS. Based on a two-tiered semi-tree topology, NEWOPS allows a large number of ONUs to be connected to OLT using only a handful of colorless transceivers. NEWOPS' unique architecture not only facilitates high spectral efficiency and flexible sub-channel bandwidth allocation, but also enables the system to achieve high energy and cost efficiency. We conduct a theoretical study to identify and resolve a new data-rate performance issue due to the colorless and wavelength re-use design of the system. Experimental results show that when only 5-GHz bandwidth is allotted for each wavelength, NEWOPS successfully achieves symmetric 20-Gb/s downstream and upstream OFDM-16QAM transmissions via the same fiber trunk. As a demonstration of our system, we show how NEWOPS achieves energy efficiency that is two to twelve times higher than that of a typical tree-based WDM/OFDM PON when supporting 40 ONUs.

In this paper, we propose an integrated traffic control scheme (ITCS), which facilitates circuit emulation service (CES) for wireless backhaul over a previously proposed experimental optical packet-switched WDM ring network, HOPSMAN. To ensure efficient setup of CES connections, ITCS employs a simple mean-rate-based distributed admission control followed by a novel slot-marking reservation. The total connection setup delay is comprised of the setup queueing delay and slot-marking delay. While the slot-marking delay is nearly a ring time, the mean setup queueing delay is formally computed through an accurate approximation based on an M/G/m queueing analysis. Simulation results show that ITCS accommodates remarkably high CES traffic loads while satisfying a wide range of delay and jitter requirements for wireless backhaul.

On behalf of the IEEE Communications Society and members of the organizing committee, it is our pleasure to welcome you to the 2013 IEEE 14th International Conference on High Performance Switching and Routing (IEEE HPSR 2013) in Taipei, Taiwan. The conference continues to be a leading forum for practitioners and researchers in this area.

OFDM PON has been considered to be one of the most prominent next-generation broadband wired access solutions. However, based on the current tree-architecture, existing OFDM PON systems face severe challenges when increasing the scalability and data-rate performance. In this paper, we propose a novel virtual-tree architecture of an OFDMA PON system (VTOPS). Coupled with the use of inexpensive direct modulation, VTOPS simultaneously achieves high reliability, scalability, and spectrum efficiency (i.e., high data rate) in a cost-effective manner. Theoretical and experimental results demonstrate that, by applying the pre-emphasis algorithm and subchannel adaptive modulation, VTOPS achieves 40-Gb/s downlink and uplink transmissions using low-cost 10-GHz directly modulated lasers.

In this paper, we propose a QoS parallel incremental scheduling (QPIS) algorithm for a variant of an earlier-proposed 10-Gb/s QoS optical packet switching system. The variant system consists of optical pseudo-Banyan space switches, a handful of fiber-delay-line (FDL)-based optical buffers, and four-wave-mixing (FWM) wavelength converters. The QPIS minimizes the loss probability for high-priority packets while maximizing system throughput and satisfying two constraints (switch-contention free, and buffer-contention free). Most notably, we prove that QPIS is incremental, i.e., the computed packet sets within each time slot are monotonically non-decreasing, and then derive the computational complexity. From simulation results that pit the QPIS algorithm against two other sequential algorithms, we show that QPIS outperforms these algorithms on packet loss probability, QoS differentiation, and computational complexity.

In this paper, we propose a novel integrated traffic control scheme, called ITCS, which seamlessly supports connection-oriented and connectionless services for a previously proposed experimental optical packet-switched metro WDM network, HOPSMAN. ITCS aims to efficiently satisfy the diverse quality-of-service (QoS) requirements while retaining maximal network throughput. Through a flexible reservation scheme, ITCS avoids the receiver-contention problem inhered in current WDM network, achieving high statistical multiplexing gain for isochronous traffic. By employing constant mean rate reservation and with an effective distributed call admission control (CAC), ITCS establishes isochronous traffic connections only within a single ring time under normal loads and achieves exceedingly low VBR delay. Furthermore, we develop a novel approximation to acquire the accurate results of the expected connection setup delay by means of an M/G/m queueing analysis. Extensive simulation results show that the mean setup queueing time is in profound agreement with the analytic results, and that ITCS accommodates maximal isochronous traffic with remarkable QoS performance and still retaining exceptional aggregate system throughput.

Signaling control schemes for wireless multimedia services are expected to maximize the number of newly-accepted mobile terminals (MTs), i.e., system throughput, while maintaining the quality of service (QoS) requirements for the existing MTs in the system. In this paper, we propose a γ-percent time-bound signaling control (γ-TBSC) scheme that resolves a maximum number of MTs mostly within a given bounded time, subject to assuring γ-percent soft guarantee for existing MTs. The γ-TBSC scheme performs signaling control in two phases: the call-invitation phase, and the contention-resolution phase. In the first phase, an access point invites a number of so-called potential MTs to make reservations during a call invitation interval (CII). If any collision occurs, γ-TBSC performs contention resolution in the second phase until all potential MTs are resolved successfully. Most importantly, prior to the two-phase operation, γ-TBSC analytically and statistically determines the potential MT number and the CII length such that the system throughput is maximized and the QoS is soft guaranteed. From simulation results that pit γ-TBSC against several variants of carrier sense multiple access with collision avoidance (CSMA/CA) schemes, we delineate that γ-TBSC outperforms these schemes with respect to throughput and delay performance.

Broadband access networking is an important form of basic infrastructure that supports competitive social and economic development. Emerging broadband applications such as IPTV, real-time network gaming, backhauling of broadband wireless connection data etc., demand new technological innovations to meet their ever-increasing performance-to-cost requirements. Optical fiber has been established as a future-proof medium capable of facilitating the next-generation network applications. For instance, in Feb. 2010 Google just announced the plans to build an experimental gigabit-per-second fiber-to-the-home (FTTH) network to households in North America for testing out new concepts in technologies and applications. Worldwide, FTTH deployment had surpassed thirty million users in 2009 and is still continuing to grow at a rapid pace.

In this paper, we present the design of an experimental optical-header processing and access control system (OPACS) for an optical packet switched (OPS) WDM metro slotted-ring network. Each control header is in-band time-division-multiplexed with its payload. OP ACS enables the optical headers across all parallel wavelengths to be efficiently received, modified, and re-transmitted based on a wavelength-time conversion technique. Bandwidth access in OP ACS is governed by a versatile access control scheme, called the distributed multi-granularity and multi-window reservation (DMGWR) mechanism. With DMGWR, the OP ACS network is shown to outperform two existing networks with respect to throughput, access delay, and fairness under various traffic patterns. Experimental results demonstrate that all optical headers are removed and combined with the data in a fully synchronous manner, justifying the viability of the system.

In this paper, we present the access and control design of a high-performance optical packet-switched WDM metro ring network (HOPSMAN). HOPSMAN has been designed for networks and nodes to be unconstrained by the number of wavelengths. It includes a handful of nodes that are equipped with fast optical slot erasers making bandwidth reusable and achieving greater bandwidth efficiency. In essence, HOPSMAN incorporates a versatile medium access control (MAC) scheme, which embodies efficient and fair bandwidth allocation in accordance with a quota being exerted probabilistically. The quota is analytically derived with the number of slot-eraser-nodes taken into account. The scheme also employs a new notion of credit to regulate flexible access of remaining bandwidth that is suitable for the metro environment with bursty traffic. With the MAC scheme, HOPSMAN is shown to achieve exceptional throughput, delay, and fairness performance under a wide range of traffic settings via simulation results.

Optical Packet Switching (OPS) has been envisioned as a prominent future optical networking technology for datacentric IP over Wavelength Division Multiplexing (WDM) networks, or optical Internet. Such OPS technology however raises significant transport and Quality of Service (QoS) challenges due to technological limitations. To circumvent OPS limitations, we have proposed a new Optical Coarse Packet Switching (OCPS) paradigm, which uses in-band-controlled per-burst switching and advocates traffic control enforcement to achieve high packet-loss performance and differentiated services. Based on OCPS, we have constructed an experimental IP-over-WDM network, referred to as OPSINET. OPSINET consists of two major types of nodes-edge routers, and Optical Label Switched Routers (OLSRs), and is facilitated with an out-of-band Generalized Multi-protocol Label Switching (GMPLS) control network. In this paper, we first introduce the OCPS paradigm. We then present the architecture of OPSINET, describe the in-band header/payload modulation technique, and detail the operations of the edge routers, OLSRs, and GMPLS control.

In this paper, we propose an efficient approximation approach, called Lagrangean relaxation with heuristics (LRH), aimed to resolve routing and wavelength assignment (RWA) for multi-granularity WDM networks facilitating fiber, waveband, and lambda switching capabilities. The task is first formulated as a combinatorial optimization problem in which the bottleneck link utilization is to be minimized. The LRH approach performs constraint relaxation and derives a lower-bound solution index according to a set of Lagrangean multipliers generated through subgradient-based iterations. In parallel, using the generated Lagrangean multipliers, the LRH approach employs a new heuristic algorithm to arrive at a near-optimal upper-bound solution. With lower and upper bounds, we delineate the performance of LRH with respect to accuracy and convergence speed under different parameter settings. We further draw comparisons between LRH and a typical linear programming (LP) approach via experiments over the widely-used NSFNET and three randomly generated networks. Numerical results demonstrate that LRH outperforms the LP approach in both accuracy and computational time complexity particularly for larger sized networks.

Due to MAC collisions and mobility, quality of service (QoS) provisioning in mobile ad hoc networks is very challenging. In this paper, we propose an innovative scheme to grant high priority to communications between some important nodes in TDMA based ad hoc networks. In addition to bandwidth reservation in previous QoS routing approaches, our scheme adopts a new approach: bandwidth prereservation. Some bandwidth is prereserved at intermediate nodes in a quadrangle-shaped area formed between important nodes. The key idea is to utilize each node's geographic location information and minimize potential scheduling conflicts for transmissions. In this way, time slot collisions in adjacent wireless links along a path can be reduced so that more high priority connections can be accepted into the network. Extensive simulations show that our scheme can successfully provide better communication quality at a relatively low price.

Many rate adaptation schemes at the medium access control (MAC) layer have been proposed to utilize the multirate capability offered by the IEEE 802.11 wireless MAC protocol through automatically adjusting the transmission rate to best match the channel conditions. In this paper, we present the Opportunistic packet Scheduling and Auto Rate (OSAR) protocol to exploit the channel variations. The basic idea of OSAR is as follows: rather than just matching the channel condition for a node pair in communications, our protocol takes advantage of the multiuser diversity as much as possible and adapt the rate accordingly, i.e., based on the channel conditions to its neighboring nodes, the sender chooses the neighboring node with channel quality better than certain level to schedule the transmissions of packets in its queue, then the overall system throughput may be increased. The key mechanisms of OSAR are channel aware media access, rate adaptation and packet bursting. We carry out several sets of ns-2 simulations and evaluate the impact of various factors such as channel condition, network topology and traffic load on the throughput of OSAR. Simulation results show that our proposed protocol can achieve much better performance than other auto rate schemes.

Optical wavelength-division multiplexed (WDM) networks often include optical cross-connects with multigranularity switching capability, such as switching on a single lambda, a waveband, or an entire fiber basis. In addition, it has been shown that routing and wavelength assignment (RWA) in an arbitrary mesh WDM network is an NP-complete problem. In this paper, we propose an efficient approximation approach, called Lagrangean relaxation with heuristics (LRH), aimed to resolve RWA in multigranularity WDM networks particularly with lambda and fiber switches. The task is first formulated as a combinatorial optimization problem in which the bottleneck link utilization is to be minimized. The LRH approach performs constraint relaxation and derives a lower-bound solution index according to a set of Lagrangean multipliers generated through subgradient-based iterations. In parallel, using the generated Lagrangean multipliers, the LRH approach employs a new heuristic algorithm to arrive at a near-optimal upper-bound solution. With lower and upper bounds, we conduct a performance study on LRH with respect to accuracy and convergence speed under different parameter settings. We further draw comparisons between LRH and an existing practical approach via experiments over randomly generated and several well-known large sized networks. Numerical results demonstrate that LRH outperforms the existing approach in both accuracy and computational time complexity, particularly for larger sized networks.

For IP-over-WDM networks, optical coarse packet switching (OCPS) has been proposed to circumvent optical packet switching limitations by using in-band-controlled per-burst switching and advocating traffic control enforcement to achieve high bandwidth utilization and quality-of-service (QoS). In this paper, we first introduce the OCPS paradigm. Significantly, we present a QoS-enhanced traffic control scheme exerted during packet aggregation at ingress nodes, aiming at providing delay and loss class differentiations for OCPS networks. Serving a dual purpose, the scheme is called (/spl psi/,/spl tau/)-Scheduler/Shaper, where /spl psi/ and /spl tau/ are the maximum burst size and burst assembly time, respectively. To provide delay class differentiation, for IP packet flows designated with delay-associated weights, (/spl psi/,/spl tau/)-Scheduler performs packet scheduling and assembly into bursts based on their weights and a virtual window of size /spl psi/. The guaranteed delay bound for each delay class is quantified via the formal specification of a stepwise service curve. To provide loss class differentiation, (/spl psi/,/spl tau/)-Shaper facilitates traffic shaping with larger burst sizes assigned to higher loss priority classes. To examine the shaping effect on loss performance, we analytically derive the departure process of (/spl psi/,/spl tau/)-Shaper. The aggregate packet arrivals are modeled as a two-state Markov modulated Bernoulli process (MMBP) with batch arrivals. Analytical results delineate that (/spl psi/,/spl tau/)-Shaper yields substantial reduction, proportional to the burst size, in the coefficient of variation of the burst interdeparture time. Furthermore, we conduct extensive simulations on a 24-node ARPANET network to draw packet loss comparisons between OCPS and just-enough-time (JET)-based OBS. Simulation results demonstrate that, through burst size adjustment, (/spl psi/,/spl tau/)-Shaper effectively achieves differentiation of loss classes. Essentially, compared to JET-based OBS using out-of-band control and offset-time-based QoS strategy, OCPS is shown to achieve invariably superior packet loss probability for a high-priority class, facilitating better differentiation of loss traffic classes.

Most existing contention access schemes are inherently unstable resulting in exponentially deteriorating throughput under increased traffic loads. In this paper, we propose a wide-sense stable (WSS) efficient hexanary-feedback contention access (HFCA) scheme, capable of providing signaling traffic high performance while retaining maximal throughput for wireless access networks. HFCA performs incremental contention resolution, managing a small subset of users at a time via a two-phase process. In the first phase, a group of users is probabilistically admitted, with a negligible probability of the group size greater than five. In the second phase, all users in the group are efficiently resolved. The two-phase process is augmented with hexanary feedback control facilitated by a probability density function (pdf)-based multiuser estimator (PMER) implemented at the physical layer. Basically, PMER measures the exact number of transmitting users (zero to five) in a contention slot by matching the envelope-phase pdf's histograms of received signals to a preconstructed pdf's library. To formally justify the performance of HFCA, we present throughput and stability analysis in which HFCA is shown WSS and the strict-sense stability condition is derived. Finally, analytic and simulation results delineate that, HFCA is highly robust against estimation discrepancy. Significantly, HFCA achieves high performance with respect to maximum stable and saturated throughputs, access delay, and blocking probability.

For WDM networks with multi-granularity switching, optical tunnel allocation (OTA) deals with the real-time establishment of optical tunnels between optical nodes through various optical multi-granularity switching devices. OTA is in principle a dynamic routing and wavelength assignment (RWA) problem with multi-granularity switching devices taken into account. In this paper, we propose a novel approximation approach, called Lagrangean relaxation with heuristics (LRH), aimed to resolve RWA considering both fiber and lambda switches. Such RWA is first formulated as a combinatorial optimization problem in which the bottleneck link utilization is to be minimized. To tackle the problem, the LRH approach performs constraint relaxation and derives a lower-bound solution index according to a set of Lagrangean multipliers generated through subgradient-based iterations. In parallel, using the generated Lagrangean multipliers, the LRH approach employs a new heuristic algorithm to arrive at a near-optimal upper-bound solution. Through numerical results and comparisons, we delineate that the LRH approach achieves a near-optimal solution, which is profoundly tight to its lower bound, at the expense of low computational time complexity.

Most existing contention access schemes are inherently unstable resulting in exponentially deteriorating throughput under increased traffic loads. In this paper, we propose a wide-sense stable, efficient hexanary-feedback contention access (HFCA) scheme, capable of providing signaling traffic high performance while retaining maximal throughput for wireless access networks. HFCA performs incremental contention resolution via a two-phase process. The two-phase process is augmented with hexanary feedback control facilitated by a pdf-based multiuser estimator (PMER) implemented at the physical layer. We present throughput and stability analyses in which HFCA is shown wide-sense stable, and the strict-sense stability condition is derived. Finally, analytic and simulation results delineate that, HFCA achieves high performance with respect to maximum stable and saturated throughputs, access delay, and blocking probability.

The random access channel (RACH) incurs a collision if more than one user make signaling requests over the same time slot, spreading code, or antenna element, in TDMA, CDMA, or SDMA wireless networks, respectively. We propose a hexanary-feedback collision resolution paradigm (HFCRP), capable of leveraging access efficiency of the RACH for all above wireless networks. HFCRP performs incremental contention resolution, managing a small subset of users at a time via a two-phase process. In the first phase, a group of users is probabilistically admitted, with a negligible probability of the group size greater than five. In the second phase, all users in the group are efficiently resolved. The two-phase process is augmented with hexanary feedback control facilitated by an envelope-PDF-based multi-user estimator (MR) implemented at the physical layer. Significantly, the MR measures the exact number of transmitting users (zero to five) in a contention slot by matching the envelope PDFs histograms of received signals to a library of pre-constructed PDFs. These six outcomes comprise the hexanary feedback being broadcast to users during the real-time operation of the two-phase process. Analytic and simulation results delineate that, performing in conjunction with the MR, HFCRP achieves high performance with respect to maximum stable throughput, saturated throughput, access delay, and blocking probability.

Prevailing weight-based scheduling algorithms advocate static or coarse-grained simplification of timestamp computation, resulting in performance bottlenecks and unnecessary downgrades under heavy and normal computation loads, respectively. Our goal is to design a scheduler that enables quantitative trade-off balance between performance and complexity. In this paper, we propose a stepwise QoS scheduler (SQS), enabling packet to burst scheduling under normal to heavy computation loads, based on a new notion of window. Window sizes of unity and greater than one correspond to packet and burst scheduling, respectively. SQS exerts simple FIFO service within the window and guarantees stepwise weight-proportional service at the window boundary. As a result, quantitative trade-off balance can be facilitated via the adjustment of the window size. We further formally specify the SQS service guarantee for a flow by a stepwise service curve, which is a function of W and /spl theta//sub min/-the least-upper-bound delay incurred by the first packet of the busy period of the flow. Finally, we demonstrate through experimental results that, by applying a small window (W/spl les/10), SQS outperforms weighted fair queueing (WFQ) and performs as superior as worst-case fair weighted fair queueing (WF/sup 2/Q) with respect to throughput fairness, mean delay, and worst-case delay fairness.

We propose a QoS-guaranteed MAC protocol (QMAC) augmented with dynamic granularity (frame size) control, aiming to provide bandwidth-on-demand while retaining maximal throughput. QMAC supports three types of ATM traffic-CBR, VBR, ABR, along with reservation request (RVR) traffic for making connection reservation. While CBR/VBR/ABR is governed by reservation access over the reservation bandwidth, RVR is conducted by random access over the contention bandwidth. QMAC essentially exerts dynamic granularity control on per-frame-based allocation of the reservation and contention bandwidth. The reservation is allocated following a weight-based scheduling policy. For the allocation of the contention bandwidth, based on a neural-fuzzy traffic prediction technique, QMAC derives the bandwidth implying maximum throughput. The smaller value between the favorable and remaining unreserved is designated as the contention bandwidth. Simulation results demonstrate that, compared to a set of MAC alternatives under M-Pareto-distributed and dynamic traffic settings, QMAC invariantly achieves superior performance with respect to throughput, access delay, and blocking probability.

In this paper, we propose a MAC protocol, called QMAC+, which adopts variable frame granularity, aiming to provide bandwidth-on-demand and QoS guarantees while retaining maximal network throughput. QMAC+ supports three traffic types-BR, VBR, and reservation request (RVR) for making CBR/VBR connection reservation. Essentially, QMAC+ governs weight-based reservation access (for VBR/CBR) and premier contention access (for RVR) over dynamically allocated reservation and premier contention bandwidths, respectively. In particular, the premier contention access performs incremental contention resolution by means of feedback control using an envelope-pdf-based estimator for the estimation of the number of concurrently transmitting users. As a result, the frame granularity is dynamically changed adapting to traffic fluctuations. Simulation results demonstrate that QMAC+ achieves superior performance with respect to throughput, delay, and blocking probability.

Existing TDM-based contention access schemes exhibit the unstable or load-susceptible problem in which throughput exponentially deteriorates as the traffic offered load increases or fluctuates. In this paper, we propose a persistently stable, efficient hexanary-feedback-based contention access (HFCA) scheme, aiming to provide QoS guarantee while retaining stable and maximal network throughput. Unlike existing schemes that consider all contending mobile terminals as a whole, HFCA incrementally resolves a small group of terminals each time via a two-phase selection/resolution process. The incremental resolution repeats until the minimum non-blocking probability (QoS for signaling traffic) is satisfied. To accelerate resolution within the process, the number of simultaneously transmitting terminals is estimated based on an envelope-pdf-based estimator (EPER), by matching the corresponding normalized envelope histogram of the receiving signal with the priori envelope-pdf library. Facilitated with incremental resolution and the EPER, HFCA achieves the highest maximum throughput (0.574) and saturated throughput (0.52) reported to date irrespective of traffic increase and variation.

Marginal reliability importance (MRI) of a link with respect to terminal-pair reliability (TR) is the rate to which TR changes with the modification of the success probability of the link. It is a quantitative measure reflecting the importance of the individual link in contributing to TR of a given network. Computing MRI for general networks is an NP-complete problem. Attention has been drawn to a particular set of networks (reducible networks), which can be simplified to source-sink (2-node) networks via 6 simple reduction rules (axioms). The computational complexity of the MRI problem for such networks is polynomial bounded. This paper proposes a new reduction rule, referred to as triangle reduction. The triangle reduction rule transforms a graph containing a triangle subgraph to that excluding the base of the triangle, with constant complexity. Networks which can be fully reduced to source-sink networks by the triangle reduction rule, in addition to the 6 reduction rules, are further defined as reducible/sup +/ networks. For efficient computation of MRI for reducible/sup +/ networks, a 2-phase (2-P) algorithm is given. The 2-P algorithm performs network reduction in phase 1. In each reduction step, the 2-P algorithm generates the correlation, quantified by a reduction factor, between the original network and the reduced network. In phase 2, the 2-P algorithm backtracks the reduction steps and computes MRI, based on the reduction factors generated in phase 1 and a set of closed-form TR formulas. As a result, the 2-P algorithm yields a linearly bounded complexity for the computation of MRI for reducible/sup +/ networks. Experimental results from real networks and benchmarks show the superiority, by two orders of magnitude, of the 2-P algorithm over the traditional approach.

We propose a new contention access protocol (CAP) augmented with a dynamic bandwidth allocator (DBA) for wireless ATM networks supporting ABR and signaling control (SCR) traffic. CAP incorporates a dynamic tree-splitting collision resolution algorithm parameterized by an optimal splitting depth (SD). DBA performs estimation and on-line prediction of ABR self-similar traffic characteristics. It in turn determines the optimal SD per frame, satisfying ABR throughput and SCR blocking probability requirements while retaining maximal aggregate throughput. Simulation results postulate the optimal SDs under various ABR and SCR traffic conditions. These results are then off-line trained and constructed by a backpropagation neural network (BPNN), which is used on-line for optimal bandwidth allocation.

Two major challenges pertaining to wireless asynchronous transfer mode (ATM) networks are the design of multiple access control (MAC), and dynamic bandwidth allocation. While the former draws more attention, the latter has been considered nontrivial and remains mostly unresolved. We propose a new intelligent multiple access control system (IMACS) which includes a versatile MAC scheme augmented with dynamic bandwidth allocation, for wireless ATM networks. IMACS supports four types of traffic-CBR, VBR, ABR, and signaling control (SCR). It aims to efficiently satisfy their diverse quality-of-service (QoS) requirements while retaining maximal network throughput. IMACS is composed of three components: multiple access controller (MACER), traffic estimator/predictor (TEP), and intelligent bandwidth allocator (IBA). MACER employs a hybrid-mode TDMA scheme, in which its contention access is based on a new dynamic-tree-splitting (DTS) collision resolution algorithm parameterized by an optimal splitting depth (SD). TEP performs periodic estimation and on-line prediction of ABR self-similar traffic characteristics based on wavelet analysis and a neural-fuzzy technique. IBA is responsible for static bandwidth allocation for CBR/VBR traffic following a closed-form formula. In cooperation with TEP, IBA governs dynamic bandwidth allocation for ABR/SCR traffic through determining the optimal SD. The optimal SDs under various traffic conditions are postulated via experimental results, and then off-line constructed using a back propagation neural network (BPNN), being used on-line by IBA. Consequently, with dynamic bandwidth allocation, IMACS offers various QoS guarantees and maximizes network throughput irrelevant to traffic variation.

Marginal reliability importance (MRI) of a link is a quantitative measure reflecting the significance of the link in contributing to the terminal-pair reliability (TR) of a given network. The computation of MRI for general networks has been shown to be an NP-complete problem. Moreover, attention has been drawn to a particular set of networks, called reducible networks, which can be simplified to source-sink (two-node) networks via six simple reduction axioms. The computational complexity of the MRI problem for such networks has been shown to be polynomial bounded. In this paper, we first propose a new reduction axiom, referred to as triangle reduction. Networks which can be fully reduced to source-sink networks by the triangle reduction axiom, in addition to the six reduction rules, are further defined as reducible/sup +/ networks. For efficient computation of MRI for reducible/sup +/ networks, we further propose a two-phase algorithm. The algorithm performs network reduction in the first phase, and backtracks the reduction steps and computes MRIs in the second phase. The two-phase algorithm, as shown, yields a linearly bounded complexity for the computation of MRI for reducible/sup +/ networks.

Quality of service (QOS) for diverse multimedia applications can be satisfied by the provision of synchronization platforms furnishing intramedia/intermedia synchronization and efficient process multitasking at end systems. As a result, the design of viable platforms should confront two concerns: network delay and jitter, and system multitasking. In this paper, we propose an intelligent synchronization platform, called SYNC*, satisfying given QOSs in terms of skew and throughput. SYNC* is composed of three components: intelligent video smoother (IVS), synchronizer, and multi-thread priority manager (MPM). While IVS and synchronizer deal with the network delay/jitter concern, MPM manages the challenge of system multitasking. We demonstrate via prototyping results that SYNC* achieves superior playout with respect to skew and throughput. We also observe that, the absence of either IVS/synchronizer or MPM results in unsatisfactory playout.

When transporting voice data with silence suppression over the Internet, the problem of jitter introduced from the network often renders the speech unintelligible. It is thus indispensable to offer intramedia synchronization to remove jitter while retaining minimal playout delay (PD). We propose a neural network (NN)-based intravoice synchronization mechanism, called the intelligent voice smoother (IVoS). The IVoS is composed of three components: (1) the smoother buffer; (2) the NN traffic predictor; and (3) the constant bit rate (CBR) enforcer. Newly arriving frames, assumed to follow a generic Markov modulated Bernoulli process (MMBP), are queued in the smoother buffer. The NN traffic predictor employs an online-trained back propagation NN (BPNN) to predict three traffic characteristics of every newly encountered talkspurt period. Based on the predicted characteristics, the CBR enforcer derives an adaptive buffering delay (ABD) by means of a near-optimal simple closed-form formula. It then imposes the delay on the playout of the first frame in the talkspurt period. The CBR enforcer in turn regulates CBR-based departures for the remaining frames of the talkspurt, aiming at assuring minimal mean and variance of distortion of talkspurts (DOT) and mean PD. Simulation results reveal that, compared to three other playout approaches, the IVoS achieves superior playout, yielding negligible DOT and PD, irrespective of traffic variation.

Terminal-pair reliability (TR) in network management determines the probabilistic reliability between two nodes (the source and sink) of a network, given failure probabilities of all links. It has been shown that TR can be effectively computed by means of the network reduction technique. Existing reduction axioms, unfortunately, are limited to simple rules such as valueless link removal and series-parallel link reduction. We propose a novel reduction axiom, referred to as triangle reduction. The triangle reduction axiom transforms a graph containing a triangle subgraph to that excluding the base of the triangle. The computational complexity of the transformation is as low as O(1). The paper further provides an assessment of the effectiveness of triangle reduction on partition-based TR algorithms with respect to the number of subproblems and computation time. Experimental results demonstrate that, incorporating triangle reduction, the partition-based TR algorithms yield a substantially reduced number of subproblems and computation time for all benchmarks and random networks.

For transporting voice data with silence suppression over the Internet, the problem of jitter introduced from the network often renders the speech unintelligible. It is thus indispensable to offer intramedia synchronization to remove jitter while retaining minimal playout delay. We propose a neural-network-based intra-voice synchronization mechanism, called the intelligent voice smoother (IVoS). The IVoS is composed of three components: smoother buffer, neural network (NN) traffic predictor, and constant bit rate (CBR) enforcer. Newly arriving frames, being assumed to follow a generic Markov modulated Bernoulli process (MMBP), are queued in the smoother buffer. The NN traffic predictor employs an on-line-trained backpropagation neural network (BPNN) to predict three traffic characteristics of every newly encountered talkspurt period. Based on the predicted characteristics, the CBR enforcer derives an adaptive buffering delay. It then imposes such delay on the playout of the first frame in the talkspurt period. The CBR enforcer in turn regulates CBR-based departures for the remaining frames of the talkspurt, aiming at assuring minimal mean and variance of distortion of talkspurts (DOT) and mean playout delay (PD). Simulation results reveal that, compared to three other playout approaches, IVoS achieves superior playout yielding negligible DOT and PD irrespective of traffic variation.

We propose a versatile scheduling discipline, called precedence with partial push-out (PPP), in asynchronous transfer mode (ATM) switches supporting two delay and two loss priorities. By employing a threshold L, PPP provides delay guarantee by allowing a newly-arriving high-delay-priority cell to precede a maximum of L low-delay-priority cells. Through the use of another threshold R, the discipline offers loss guarantee by permitting a newly-arriving high-loss-priority cell to push out the last low-loss-priority cell located beyond the Rth location in a full queue. By setting L and R properly, PPP versatilely performs as any one of the four widely-accepted disciplines, namely the FCFS, head-of-line, push-out, or head-of-line with push-out disciplines. To determine L and R retaining demanded quality of services (QoSs), we provide an in-depth queueing analysis for the cell delay (CD) and cell loss ratio (CLR) of high-delay-priority low-loss-priority cells. We further propose a simple, algebra-based analysis for the CD and CLR for low-delay-priority, high-loss-priority cells. On the basis of these analyses, L and R can be dynamically and effectively adjusted to provide adequate delay and loss guarantees for high-priority cells while incurring only minimal performance degradation for other classes of cells. Finally, the paper presents simulation results confirming the accuracy of the analyses.

For distinctively transporting voice data with silence suppression over asynchronous transfer mode (ATM) networks via the variable bit rate (VBR) service, the problem of jitter introduced from the network often renders the speech unintelligible. It is thus indispensable to offer intramedia synchronization to remove jitter while retaining minimal playout delay. We propose a neural-network-based intra-voice synchronization mechanism, called the intelligent voice smoother (IVoS). The IVoS is composed of three components: smoother buffer, neural network (NN) traffic predictor, and constant bit rate (CBR) enforcer. Newly arriving frames, being assumed to follow a generic Markov modulated Bernoulli process (MMBP), are queued in the smoother buffer. The NN traffic predictor employs an on-line-trained backpropagation neural network (BPNN) to predict three traffic characteristics of every newly encountered talkspurt period. Based on the predicted characteristics, the CBR enforcer derives an adaptive buffering delay by means of a near-optimal, simple, closed-form formula. It then imposes such delay on the playout of the first frame in the talkspurt period. The CBR enforcer in turn regulates CBR-based departures for the remaining frames of the talkspurt, aimed at assuring minimal mean and variance of distortion of talkspurts (DOT) and mean playout delay (PD). Simulation results reveal that, compared to three other playout approaches, the IVoS achieves superior playout yielding negligible DOT and PD irrespective of traffic variation.

Multimedia communications often require intramedia synchronization for video data to prevent potential playout discontinuity resulting from network delay variation (jitter) while still achieving satisfactory playout throughput. In this paper, we propose a neural network (NN) based intravideo synchronization mechanism, called the intelligent video smoother (IVS), operating at the application layer of the receiving end system. The IVS is composed of an NN traffic predictor, an NN window determinator, and a window-based playout smoothing algorithm. The NN traffic predictor employs an on-line-trained back-propagation neural network (BPNN) to periodically predict the characteristics of traffic modeled by a generic interrupted Bernoulli process (IBP) over a future fixed time period. With the predicted traffic characteristics, the NN window determinator determines the corresponding optimal window by means of an off-line-trained BPNN in an effort to achieve a maximum of the playout quality (Q) value. The window-based playout smoothing algorithm then dynamically adopts various playout rates according to the window and the number of packets in the buffer. Finally, we show that via simulation results and live video scenes, compared to two other playout approaches, IVS achieves high-throughput and low-discontinuity playout under a mixture of IBP arrivals.

Call admission control (CAC) has been accepted as a potential solution for supporting diverse, heterogeneous traffic sources demanding different quality of services in asynchronous transfer mode (ATM) networks. Besides, CAC is required to consume a minimum of time and space to make call acceptance decisions. We present an efficient neural-network-based CAC (NNCAC) mechanism for ATM networks with heterogeneous arrivals. All heterogeneous traffic calls are initially categorized into various classes. Based on the number of calls in each class, the NNCAC efficiently and accurately estimates the cell delay and cell loss ratio of each class in real time by means of a pre-trained neural network. According to our decent study which exhibits the superiority of the employment of analysis-based training data over simulation-based data, we particularly construct the training data from heterogeneous-arrival dual-class queueing model M/sup [N1]/+I/sup [N2]//D/1/K, where M and I represent the Bernoulli process and interrupted Bernoulli process, and N/sub 1/ and N/sub 2/ represent the corresponding numbers of calls, respectively. Analytic results of the queueing model are confirmed by simulation results. Finally, we demonstrate the profound agreement of our neural-network-based estimated results with analytic results, justifying the viability of our NNCAC mechanism.

Multimedia communications often require intra-media synchronization for video data to prevent potential playout discontinuity resulting from network delay variation (jitter) while still achieving satisfactory playout throughput. We propose a neural-network-based intra-media synchronization mechanism, called neural network smoother (NNS). The NNS is composed of a neural network (NN) traffic predictor, an NN window determinator, and a window-based playout smoothing algorithm. The NN traffic predictor employs an on-line-trained backpropagation neural network (BPNN) to periodically predict future traffic characteristics. With the predicted traffic characteristics, the NN window determinator determines the corresponding optimal window by means of an off-line-trained BPNN in an effort to achieve a maximum of the playout quality (Q) value. The window-based playout smoothing algorithm then dynamically adopts various playout rates according to the window and the number of packets in the buffer. Compared to two other playout approaches, simulation results show that NNS achieves high-throughput and low-discontinuity playout under a variety of traffic arrivals.

This paper assesses two categories of partition techniques for computing terminal-pair reliability (path-based and cut-based algorithms) by experimenting on published benchmarks; the criteria are the number of subproblems and the computation time. The cut-based algorithm is superior to the path-based algorithm with respect to the computation time for most benchmarks. A refinement of the cut-based algorithm (using network reduction) profoundly outperforms the path-based algorithm (with reduction) for all benchmarks.

Terminal-pair reliability in network management determines the probabilistic reliability between the two nodes of a network, given failure probabilities of all links. The problem has been successfully solved by the path-based partition method, network reduction technique, and the combination of the two. The path-based partition algorithm, however, requires the repeated execution of a path-search procedure in an attempt to locally minimize the total number of subproblems generated. This paper proposes a cut-based partition algorithm based on the minimum cut separating the source from the remaining network nodes. This cut-based algorithm makes no attempt to locally minimize the number of subproblems, but greatly reduces the computation time for the partitioning of each subproblem. Experimental results show the superiority of the cut-based algorithm over the path-based algorithm especially for complex networks. By further employing the network reduction technique, this cut-based partition algorithm outperforms the path-based partition algorithm for all benchmarks and random sparse or dense networks.

Multimedia systems combine a variety of information sources, such as voice, graphics, animation, images, music, and full-motion video, into a wide range of applications. The paper initially categorizes existing multimedia applications into three classes: non-interactive-oriented, interactive-oriented client-server-based, and interactive-oriented peer-party-based applications. In particular, the paper examines interactive-oriented applications and provides an in-depth survey of the media synchronization problem for the design of these applications. The paper then presents our Distributed Multimedia Teleworking System (DMTS) prototype, which allows two or more remote systems in collaboration to access and modify multimedia data through a network in a fully synchronous fashion. The system was developed over TCP/IP on an FDDI network, using an XVideo D/A card. The media supported by DMTS include text, graphics, voice, and video. For text and graphics, to maintain the coherence of the data being simultaneously modified, DMTS employs a master-slave collaboration model between the two remote systems. Moreover, DMTS also adopts effective mechanisms to reduce skew (asynchrony) and jitter delays between video and voice data streams. Finally, the paper demonstrates that DMTS achieves a maximum throughput of 13 frames per second and the throughput bottleneck resides in the hardware capture and the D/A processing of video frames.

ATM networks are expected to efficiently provide multicast communication services such as video conferencing by means of feasible multicast routing algorithms. Several minimum-cost multicast routing algorithms with various degrees of computational complexity have been proposed. These algorithms, however, could be inappropriate for ATM networks due to their high complexities. In this paper, we initially present an optimal multicast routing algorithm, called LOPT (Load Optimal), which guarantees minimum load or a minimum number of cells to be generated throughout the network. LOPT simply corresponds to a minimum-cost multicast routing algorithm if the cost of each link is assumed to be unity. The algorithm efficiently determines the minimum-load multicast route by means of partition. Complexity analysis shows its superiority over the Balakrishnan's algorithm, one of promising multicast routing algorithms, especially when the number of destinations is much smaller than the size of the network. Experimental results further exhibit even better efficiency than its theoretical results. Moreover, based on LOPT, we further propose two variants of the algorithm for applications with different requirements. The first variant determines the multicast route satisfying the condition min[max(delay)] subject to the constraint of the minimum load. The second variant determines the minimum-load route subject to the constraint of a delay bound. The former can be applied for load-sensitive networks, whereas the latter can be applied for delay-sensitive applications.

Broadband applications require high-performance networking facilities and the guarantee of quality of service (QOS). Existing transport protocols are incapable of providing QOS. Should a transport entity result in insufficient processing bandwidth, the layer becomes a performance bottleneck and can no longer guarantee QOS. In addition, the problem of how transfer rates can be determined in the implementation of non-window-based flow control mechanisms has been left unexplored. This paper presents a transport system prototype, called the Broadband Application Transport System (BATS), conducted over the user datagram protocol platform. It supports four classes of applications with different time sensitivity, bit rates, and connection types. In an effort to guarantee QOS, BATS employs a resource management scheme and a rate-based flow control protocol. The resource manager may cause longer delays for time-insensitive applications and the blocking of connections for time-sensitive applications. Moreover, for rate-based flow control, the transfer rate for each application class can be accurately determined. Consequently, compared with TB, BATS achieves superior performance in terms of end-to-end transfer delay and delay jitter for all application classes under various system loads.>

Fiber distributed data interface-II (FDDI-II) is a hybrid high speed local area network (HSLAN) providing circuit switched and packet switched services. Although the packet switched service remains connectionless, the circuit switched service is connection-oriented. The weaknesses of the media access control (MAC) protocol of FDDI-II is discussed. A new access scheme, called FDDI-II*, aimed at providing distributed control and access of bandwidth using connectionless service for both packet and isochronous data transmissions is proposed. Simulation results that demonstrate the performance superiority of FDDI*-II to FDDI-II in terms of isochronous data blocking probability and packet data delay are presented.

The authors present a reliable double-ring metropolitan area network architecture, called DORFMAN, based on an enhanced Fiber Distributed Data Interface (FDDI) protocol. DORFMAN is composed of two local ring networks interconnected via two monitors. The two monitors are used to perform simple routing and error recovery logic as well as mode transitions of DORFMAN. There are three modes defined in DORFMAN: local, global, and transition modes. During the local mode, DORFMAN allows double rings to operate in parallel. During the global mode, DORFMAN behaves as an FDDI network. Modes are dynamically switched between local and global modes on a demand basis. The bandwidth waste resulting from the transition of modes has been reduced to a minimum. Performance analysis and simulation results show that with a reasonable degree of locality assumed, twice the throughput of an FDDI network can be achieved in DORFMAN. An error recovery scheme for DORFMAN is addressed.>

A prototype workstation, called communication protocol analyzer and simulator system (CPASS), aimed at the modeling, verification, analysis, and simulation for real-time networks by means of a formal and automated approach is presented. The system is based on an extended finite state machine model, called SDL/sup N/ (a variant of SDL). It is composed of six components: network topology editor, SDL/sup N/GR editor, protocol verifier and analyzer, SDL/sup N/-to-C compiler, simulator, and statistical generator and comparator. The functionality of each component of CPASS is described. An example of a token ring network is used to illustrate the complete processes of modeling, analysis, and simulation.>

Distributed queue dual bus (DQDB) was shown to exhibit throughput unfairness problems in which bandwidth allocation is dependent on the physical locations of nodes, as the size and load of the network increase. To eliminate the unfairness problems, various mechanisms have been proposed. This paper first analyzes the unfairness problems and presents a survey of these mechanisms, including an analysis of their strengths and weaknesses. The paper then proposes a fair and fast protocol, called multiple segment control (MSC). MSC grants node multiple slots on continuation basis for any single request issued in an attempt to minimize the propagation delay. The node keeps being granted until it has issued an 'end' request. Consequently, MSC offers fair and high throughput to all nodes for both light and heavy load conditions. Finally, the performance of MSC is evaluated and compared with that of DQDB via simulation results.>

A parallel protocol verification algorithm, called the two-phase algorithm, is proposed in an attempt to provide a maximum of verification with a minimum of state space. Rather than compose all communicating finite-state machines (FSMs) into one large global reachability tree, the two-phase algorithm constructs a local expanded tree for each FSM augmented with external information. The first phase of the algorithm performs the expanded tree construction and the second phase performs error detection based on the constructed expanded trees. By separating verification into two phases, the algorithm allows verification for all FSMs to be executed in parallel. The algorithm thus requires a shorter run-time. Moreover, by introducing a new method for the construction of the expanded trees, the algorithm requires fewer explored states. The algorithm can verify protocols with any number of processes. Verification for protocols with more than two processes is illustrated.>