Exploitation of DWDM for optical packet switching with quality ofservice guarantees

This paper addresses the problem of building optical packet switches that are able to effectively cope with variable length packet traffic and quality of service management, therefore able to support IP traffic. The paper aims at showing that the availability of dense wavelength division multiplexing is crucial. By suitably exploiting the wavelength dimension a multistage fiber delay line buffer can be implemented, with fine granularity and long delay with an architecture of limited complexity. This is necessary to fulfill the buffering requirements of variable length packets. Furthermore, the wavelength domain is proved to be more effective than the time domain to manage different levels of quality of service. Algorithms are presented that are peculiarly designed for this environment showing that they can effectively differentiate the packet loss probability between three priority classes     

Buffering management schemes for optical variable length packets under limited packet sorting

This paper studies several buffering strategies for optical packet switching (OPS) under limited packet sorting. Three schemes, which are able to sort newly arrived packets based on packet’s length as well as capability of finding the minimum buffer occupancy, are analyzed and compared. Results show that all three proposed schemes could improve OPS performance considerably in terms of probability of packet loss (PPL) and probability of information loss (PIL). In addition, the simulation results show that not all the newly arrived packets need to be sorted in order to obtain minimum packet loss probability. Since the amount of packets and thus the packet processing time is significant in OPS, it is possible that not all the packets can be processed using one of the buffering strategies. An important finding of this paper is that if only 10% of the packets are sorted, the PPL is comparable to the minimum packet loss value obtained when 100% of the packets are sorted.     

Traffic performance evaluation of an optical packet switch with multicast operation

We recently proposed a multicast-enabled optical packet switch architecture utilizing multicast modules. In this paper, we evaluate the traffic performance of our earlier proposed packet switch under a hybrid traffic model through simulations. The multicast packets are given higher priority than unicast packets so that only a small number of multicast modules are needed. The results show that the switch can achieve an acceptable packet loss probability in conjunction with a packet scheduling technique.     

一种新的用于IEEE 802.11e EDCA中提供QoS的方法

该文提出了一种新的应用于IEEE 802.11e EDCA (Enhanced Distributed Channel Access)中提供QoS(Quality of Service)的方法。这种方法是将几个时隙组合起来构成一个超时隙,每个超时隙的开始分配给不同的业务来进行发包。时隙的分配是根据各种业务的不同优先级来实现的。这种方法可以保证高优先级业务具有较大的吞吐量,较少的MAC延时和较低的丢包率。与802.11e EDCA草案中提出的不同冲突窗口大小的方法相比,这种方法具有可以提高吞吐量,降低丢包率,并能减小站点数目变化对高优先级业务吞吐量的影响等优点。这种新的提供QoS的方法优于不同冲突窗口大小的方法,在IEEE 802.11e EDCA中应用超时隙方法可以大大提高EDCA的性能。     

Optimal strategies for admitting voice and data traffic in networks of LEO satellites using CDMA

Efficient policies are derived for admitting voice and data traffic into networks of low-earth-orbit (LEO) satellites using code-division multiple-access (CDMA) with direct-sequence spread-spectrum (DS/SS) signaling. The satellites act as bent-pipes; no on-board processing or intersatellite links are present. Dual satellite diversity is used to mitigate the effects of shadowing. The policies assume a movable boundary form, allocate optimally the CDMA capacity (PN codes) to voice and data users, and can increase significantly the number of users served while satisfying their bit error rate (BER) requirements. In contrast to direct admission policies that do not take into consideration the statistical features of the traffic, the new policies do account for the different nature of voice and data traffic and the history of prior transmissions/admissions. Two priority schemes are considered: voice users have higher priority than data users, or voice and data users have the same priority. A modified version of our policies can handle two classes of data users: one with high priority which requires real-time delivery and another with low priority that can be queued; the BER requirements of the two data types may differ. Optimal policies have lower voice blocking rates and data packet error rates than direct admission policies.     

Performance analysis of handover delay and buffer capacity in mobile OpenFlow‐based networks

Software‐defined networking (SDN) is a network concept that brings significant benefits for the mobile cellular operators. In an SDN‐based core network, the average service time of an OpenFlow switch is highly influenced by the total capacity and type of the output buffer, which is used for temporary storage of the incoming packets. In this work, the main goal is to model the handover delay due to the exchange of OpenFlow‐related messages in mobile SDN networks. The handover delay is defined as the overall delay experienced by the mobile node within the handover procedure, when reestablishing an ongoing session from the switch in the source eNodeB to the switch in the destination eNodeB. We propose a new analytical model, and we compare two systems with different SDN switch designs that model a continuous time Markov process by using quasi‐birth–death processes: (1) single shared buffer without priority (model SFB), used for all output ports for both control and user traffic, and (2) two isolated buffers with priority (model priority finite buffering [PFB]), one for control and the other for user plane traffic, where the control traffic is always prioritized. The two proposed systems are compared in terms of total handover delay and minimal buffer capacity needed to satisfy a certain packet error ratio imposed by the link. The mathematical modeling is verified via extensive simulations. In terms of handover delay, the results show that the model PFB outperforms the model SFB, especially for networks with high number of users and high probability of packet‐in messages. As for the buffer dimensioning analysis, for lower arrival rates, low number of users, and low probability of packet‐in messages, the model SFB has the advantage of requiring a smaller buffer size.     

Analysis of storage and retrieval policies for ATM buffers

To satisfy the different performance (QoS) requirements of the different types of ATM traffic, several control strategies have been described in the literature. The control strategies that operate on cell level are combinations of storage and retrieval priority policies. In the comparisons, two classes of traffic are distinguished, one class (LL) requiring a low loss probability and one class (LD) requiring a low average delay (jitter). By using the delay-loss plane — plotting the loss of the LL cells against the delay of the LD cells — the performance of the various storage and (state-dependent) retrieval policies has been compared for both stationary and time-varying traffic. In a relative ranking of the different policies, the LDOLL threshold policies give the best results, in addition to creating the possibility of setting the threshold according to the desired delay-loss trade-off.     

Performance measures and scheduling policies in ring networks

A unidirectional ring network is considered. A node may transmit at most one packet per slot to its downstream neighbor. Potentially all nodes may transmit at the same slot. The achievable performance is studied and policies are proposed for both the evacuation mode and continual operation. In the evacuation mode each node has initially an amount of packets destined for every other node of the ring, and no more packets are generated later. It is shown that the furthest destination first (FDF) policy, that gives priority to the packet with the longest way to go at each node, minimizes the time until every packet reaches its destination. Furthermore it is shown that the closest destination first (CDF) policy, that gives priority to the packet with the shortest way to go at each node, minimizes the average packet delivery time. A formula for the optimal evacuation time is obtained. The continual operation of the ring is considered then where packets are generated according to some arrival process. For any arrival sample path, the PDF maximizes the fraction of the time at which the ring is empty. The performance analysis of individual origin-destination traffic streams under FDF is facilitated based on the following. For each traffic stream, a single server priority queue is identified such that the average sojourn time of the traffic stream in the ring is equal to the aggregate transmission time plus the queueing delay of the low priority stream in the queue. Formulas for the sojourn time are obtained for iid arrivals. The performance of CDF and FIFO in continual operation is studied by simulation. It turns out that the CDF, minimum delay policy for the evacuation, has the worst performance in continual operation     

Variable IP packets buffer management schemes for optical packet switching

Effective buffering of optical packets is essential to the efficient working of optical packet switches. In this paper three new schemes, which involve sorting and finding the least occupied buffer, are proposed. Their performance is compared with the common round-robin scheme. The results show that all these new schemes are able to enhance the optical packet switch performance significantly in terms of packet drop/loss probability. In addition, the results show that not all the newly arrived packets need to be sorted in order to obtain the minimum packet drop probability. As computation/processing time is significant in optical packet switching, partial sorting of the newly arrived packets with tolerable packet drop probability appears to be a viable proposition. Conversely, a complete sort of newly arrived packets wastes packet processing time unnecessarily while significantly increasing the packet drop probability.     

DRED‐MP: Queue management with multiple levels of drop precedence

There is current interest in differentiated service architectures where packets with different priorities can share the same queue. In the case of congestion, packets marked with higher drop probability are preferentially dropped in order to make buffer room for packets marked with lower drop probability. Active queue management (AQM) based on randomized packet dropping has become a key component of this packet forwarding model. This paper extends our previously developed AQM algorithm called DRED with multiple packet drop precedence to allow for priority treatment of traffic in a network. The main advantage of using the DRED algorithm is the lower parameter configuration complexity it offers and the ease of configuration for a wide range of network conditions. Copyright © 2004 John Wiley & Sons, Ltd.     

Assessing the accuracy of using aggregated traffic traces in network engineering

Aggregated traffic traces are commonly used in network engineering for QoS or performance parameters evaluation. Many performance models come from such aggregated traces. However, real traffic is a marked point process combining two processes: one for the arrival times of packets and the other for their size in bytes. This paper deals with assessing whether aggregated traces are a good representation of real traffic. Based on the analysis of many traffic traces, and focusing only on loss probability, it is shown that the packet drop probability obtained for the aggregated traffic traces can significantly differ from the real packet drop probability obtained for the real traffic traces. Then, a solution which enables one to obtain correct loss probability based on aggregated traffic traces is proposed by determining the correct aggregation scale and traffic parameters to be applied.     

Performance of fiber delay‐line buffers in asynchronous packet‐based optical switching networks with wavelength conversion

A major challenge in asynchronous packet‐based optical networks is packet contention, which occurs when two or more packets head to the same output at the same time. To resolve contention in the optical domain, two primary approaches are wavelength conversion and fiber delay line (FDL) buffering. In wavelength conversion, a contending packet can be converted from one wavelength to another in order to avoid conflict. In FDL buffering, contending packets can be delayed for a fixed amount of time. While the performance of wavelength conversion and FDL buffering has been evaluated extensively in synchronous networks with fixed‐sized packets, in this paper, we study the performance of FDL buffers in asynchronous packet‐based optical networks with wavelength conversion. An analytical model is proposed to evaluate the performance in terms of packet loss probability and average delay. Extensive simulation and analytical results show that, with appropriate settings, FDL buffers can perform much better in switches with wavelength conversion than in switches with no conversion. Copyright © 2014 John Wiley & Sons, Ltd.     

Output and delay of multi-channel slotted ALOHA systems for integrated voice and data transmission

Packet-switched technology has been developed to offer personal communication services not only for data but also for different types of user-end equipment such as phone-type audio. To satisfy the huge service demand and multi-traffic requirements with limited bandwidth, this paper proposes an efficient procedure of multi-channel slotted ALOHA for integrated voice and data transmission in wireless information networks and presents an exact analysis with which to numerically evaluate the performance of the systems. A channel reservation policy is applied, where a number of channels (called reserved channels) are used exclusively by voice packets, while the remaining channels are used by both voice and data packets, and voice packets select the reserved channels with a given probability (called selection probability). Probability distributions for the numbers of voice and data departures and for the data packet delay are derived. Numerical results compare some cases with different numbers of channels, different numbers of reserved channels and different selection probabilities to discuss what effects they may have on channel utilization, loss probability, average packet delay, coefficient of variation of data packet delay, and correlation coefficient of packet departures. This revised version was published online in June 2006 with corrections to the Cover Date.     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Advanced Subcarrier Multiplexed Label Swapping in Optical Packet Switching Nodes for Next Generation Internet Networks

This paper presents the recent achievements and main results of the optical packet switching node with label swapping capabilities that was developed in the framework of the IST-LABELS project. The implemented functionalities allows the routing and forwarding of optical packets at 10 Gb/s based on the information conveyed in the 18 GHz-subcarrier multiplexed (SCM) label at 155 Mb/s. Specific functions of the optical packet switching demonstrator such as label extraction and rewriting are based on high-performance fiber Bragg grating filters (FBG) whereas semiconductor optical amplifier (SOA) based wavelength conversion allows payload 2R regeneration. Advanced aspects implemented in the high speed electronic control unit such as traffic shaping, multicast, and for the first time to our knowledge, a contention resolution algorithm with priority processing are thoroughly described in this paper. This paper also presents simulations exploring the effect of these advanced features in the global performance of an optical packet switched network. In particular, we found that the contention resolution algorithm based on parallel converters with 16 wavelengths contributes to improve up to four orders of magnitude the packet loss rate for low traffic loads whereas the improvement from traffic shaping policies was of 1 order of magnitude for both low and medium traffic loads. Along with the experimental results, the feasibility of the approach as the basis for the next generation internet networks is demonstrated.     

A fast parallel-tree switch architecture for ATM networks

In this paper we present a novel fast packet switch architecture based on Banyan interconnection networks, called parallel-tree Banyan switch fabric (PTBSF). It consists of parallel Banyans (multiple outlets) arranged in a tree topology. The packets enter at the topmost Banyan. Internal conflicts are eliminated by using a conflict-free 3 × 4 switching element which distributes conflicting cells over different Banyans. Thus, cell loss may occur only at the lowest Banyan. Increasing the number of Banyans leads to a noticeable decrease in cell loss rate. The switch can be engineered to provide arbitrarily high throughput and low cell loss rate without the use of input buffering or cell pre-processing. The performance of the switch is evaluated analytically under uniform traffic load and by simulation, under a variety of asynchronous transfer mode (ATM) traffic loads. Compared to other proposed architectures, the switch exhibited stable and excellent performance with respect to cell loss and switching delay for all studied conditions as required by ATM traffic sources. The advantages of PTBSF are modularity, regularity, self-routing, low processing overhead, high throughput and robustness, under a variety of ATM traffic conditions. © 1998 John Wiley & Sons, Ltd.     

ABE: providing a low-delay service within best effort

We propose alternative best effort (ABE), a novel service for IP networks, which idea of providing low-delay at the expense of maybe less throughput. The objective is to retain the simplicity of the original Internet single-class best-effort service while providing low-delay to interactive adaptive applications. With ABE, every best effort packet is marked as either green or blue. Green packets are guaranteed a low bounded delay in every router. In exchange, green packets are more likely to be dropped (or marked using congestion notification) during periods of congestion than blue packets. For every packet, the choice of color is made by the application based on the nature of its traffic and on global traffic conditions. Typically, an interactive application with real-time deadlines, such as audio, will mark most at its packets as green, as long as the network conditions offer large enough throughput. In contrast, an application that transfers binary data such as bulk data transfer will seek to minimize overall transfer time and send blue traffic. We propose router requirements that aim at enforcing benefits for all types of traffic, namely that green traffic achieves low-delay and blue traffic receives at least as much throughput as it would in a flat (legacy) best effort network. ABE is different from differentiated or integrated services in that neither packet color can be said to receive better treatment; thus, flat rate pricing may be maintained, and there is no need for reservations or profiles. We define the ABE service, its requirements, properties, and usage. We discuss the implications of replacing the existing IP best effort service by the ABE service. We propose and analyze an implementation based on a new scheduling method called duplicate scheduling with deadlines. It supports any mixture of TCP, TCP-friendly, and non-TCP-friendly traffic     

QoS-oriented packet scheduling for wireless multimedia CDMA communications

In the third-generation (and beyond) wireless communication systems, there will be a mixture of different traffic classes, each having its own transmission rate characteristics and quality-of-service (QoS) requirements. In this paper, a QoS-oriented medium access control (MAC) protocol with fair packet loss sharing (FPLS) scheduling is proposed for wireless code-division multiple access (CDMA) communications. The QoS parameters under consideration are the transmission bit error rate (BER), packet loss, and delay requirements. The MAC protocol exploits both time-division and code-division statistical multiplexing. The BER requirements are guaranteed by properly arranging simultaneous packet transmissions and controlling there transmit power levels, whereas the packet loss and delay requirements are guaranteed by proper packet scheduling. The basic idea of FPLS is to schedule the transmission of multimedia packets in such a way that all the users have a fair share of packet loss according to their QoS requirements, which maximizes the number of the served users under the QoS constraints. Simulation results demonstrate effectiveness of the FPLS scheduler, in comparison with other previously proposed scheduling algorithms.     

Packet scheduling algorithms and performance of a bufferedshufflenet with deflection routing

In a multihop network, packets go through a number of hops before they are absorbed at their destinations. In routing to its destination using minimum path, a packet at a node may have a preferential output link (the so-called “care” packet) or may not (the so-called “don't care” packet). Since each node in an optical multihop network may have limited buffer, when such buffer runs out, contention among packets for the same output link can be resolved by deflection. In this paper, we study packet scheduling algorithms and their performance in a buffered regular network with deflection routing. Using shufflenet as an example, we show that high performance (in terms of throughput and delay) can he achieved if “care” packets can be scheduled with higher priority than “don't care” packets. We then analyze the performance of a shufflenet with this priority scheduling given the buffer size per node. Traditionally, the deflection probability of a packet at a node is solved from a transcendental equation by numerical methods which quickly becomes very cumbersome when the buffer size is greater than one packet per node. By exploiting the special topological properties of the shufflenet, we are able to simplify the analysis greatly and obtain a simple closed-form approximation of the deflection probability. The expression allows us to extract analytically the performance trend of the shufflenet with respect to its buffer and network sizes. We show that a shufflenet indeed performs very well with only one buffer, and can achieve performance close to the store-and-forward case using a buffer size as small as four packets per node     

Performance analysis of multihop lightwave networks with hot potatorouting and distance-age-priorities

Multihop lightwave networks with hot potato routing offer the potential for significantly higher aggregate capacity as compared against minimum distance routing since each interconnecting link can be operated at a rate much higher than the electro-optic constraint forced by store-and-forward buffering. If the links can be operated at a speed sufficiently high as to offset the hot potato misrouting inefficiency, then a net capacity gain will be enjoyed. In this paper, we study such a network in which contention for an outbound link is resolved by assigning to each packet a priority based on its distance to destination and its age (number of misroutes previously suffered). An approximate analysis based on a reasonable set of assumptions is presented, queueing delay, packet loss performance, and network aggregate capacity (saturation throughout) are compared for distance-age prioritization, age-distance prioritization, and no prioritization. Results suggest that the lowest delay, lowest packet loss performance, and highest saturation throughput are offered by distance-age prioritization (priority given first to a packet closest to its destination and second to the oldest of multiple packets of the same distance to destination). The tails of the probability distribution for the number of hops taken by a representative packet are, however, shorter for age-distance routing. The capacity gain with link speed-up and hot potato routing can be quite significant compared to link speed-constrained store and forward buffering, and grows with both link speed-up factor and network size