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The paper proposes a Markovian approach to the performance evaluation of the ESSE (early slow start exit) modification of the TCP congestion control mechanism. ESSE takes advantage of estimating the optimal pipesize at the sender side to properly select initial slow start threshold. Previous simulative experiments have shown that ESSE allows one to speed-up TCP connections and significantly reduces the packet drop rate at the bottleneck. This work takes a step further in understanding the ESSE behavior by developing a model of TCP source to evaluate the influence of different settings of slow start threshold on a TCP performance. As confirmed by comparison with simulations, the model provides, significantly faster than simulations, accurate estimates of typical performance indicators such as the average completion time and average drop rate of short-lived TCP connections., 1. Introduction Recent studies (1), (2) have revealed that the majority of the Internet flows are short-lived (mice), while a smaller number of long-lived connections carry most of the Internet [...]

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.comnet.2010.02.014 Byline: Gustavo Marfia (c), Claudio E. Palazzi (b), Giovanni Pau (a), Mario Gerla (a), Marco Roccetti (c) Keywords: Fairness; RTT; TCP; Transport protocol Abstract: The Transmission Control Protocol (TCP), the most widely used transport protocol over the Internet, has been advertised to implement fairness between flows competing for the same narrow link. However, when session round-trip-times (RTTs) radically differ, the share may be anything but fair. This RTT-unfairness represents a problem that severely affects the performance of long-RTT flows and whose solution requires a revision of TCP's congestion control scheme. To this aim, we discuss TCP Libra, a new transport protocol able to ensure fairness and scalability regardless of the RTT, while remaining friendly towards legacy TCP. As main contributions of this paper: (i) we focus on the model derivation and show how it leads to the design of TCP Libra; (ii) we analyze the role of its parameters and suggest how they may be adjusted to lead to asymptotic stability and fast convergence; (iii) we perform model-based, simulative, and real testbed comparisons with other TCP versions that have been reported as RTT-fair in the literature. Results demonstrate the ability of TCP Libra in ensuring RTT-fairness while remaining throughput efficient and friendly towards legacy TCP. Author Affiliation: (a) Computer Science Department, University of California, Los Angeles, CA 90095, United States (b) Dipartimento di Matematica Pura e Applicata, Universita degli Studi di Padova, 35121 Padova, Italy (c) Dipartimento di Scienze dell'Informazione, Universita di Bologna, 40126 Bologna, Italy Article History: Received 10 March 2009; Revised 9 February 2010; Accepted 24 February 2010 Article Note: (miscellaneous) Responsible editor: S. Mascolo

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.jss.2010.03.080 Byline: Mohamed Alnas, Irfan Awan, R.D.W. Holton Keywords: Mobile IPv6; Fast Handover; Mobility Management; Link-Layer Information; Handover Latency Abstract: Handover latency is the primary cause of packet loss resulting in performance degradation of standard Mobile IPv6. Mobile IPv6 with fast Handover enables a Mobile Node (MN) to quickly detect at the IP layer that it has moved to a new subnet by receiving link-related information from the link-layer; furthermore it gathers anticipative information about the new Access Point (AP) and the associated subnet prefix when the MN is still connected to the previous Corresponding Node (CN). This paper proposes an enhancement to Fast Mobile IPv6 handover (FMIPv6), based on link layer information, we also present performance evaluations in terms of the packet loss and handover latency using evaluation models. Author Affiliation: Mobile Computing, Networks and Security Research Group, School of Informatics, University of Bradford, Bradford, West Yorkshire, BD7 1DP, UK Article History: Received 9 November 2009; Revised 18 January 2010; Accepted 30 March 2010

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.comnet.2010.01.013 Byline: P. Casas (a)(c), S. Vaton (a), L. Fillatre (b), I. Nikiforov (b) Keywords: Network Monitoring and Traffic Analysis; Traffic Matrix; Network Traffic Modeling; Optimal Volume Anomaly Detection and Isolation Abstract: Recent studies from major network technology vendors forecast the advent of the Exabyte era, a massive increase in network traffic driven by high-definition video and high-speed access technology penetration. One of the most formidable difficulties that this forthcoming scenario poses for the Internet is congestion problems due to traffic volume anomalies at the core network. In the light of this challenging near future, we develop in this work different network-wide anomaly detection and isolation algorithms to deal with volume anomalies in large-scale network traffic flows, using coarse-grained measurements as a practical constraint. These algorithms present well-established optimality properties in terms of false alarm and miss detection rate, or in terms of detection/isolation delay and false detection/isolation rate, a feature absent in previous works. This represents a paramount advantage with respect to current in-house methods, as it allows to generalize results independently of particular evaluations. The detection and isolation algorithms are based on a novel linear, parsimonious, and non-data-driven spatial model for a large-scale network traffic matrix. This model allows detecting and isolating anomalies in the Origin-Destination traffic flows from aggregated measurements, reducing the overhead and avoiding the challenges of direct flow measurement. Our proposals are analyzed and validated using real traffic and network topologies from three different large-scale IP backbone networks. Author Affiliation: (a) Telecom Bretagne, Brest, France (b) Universite de Technolgie de Troyes, Troyes, France (c) Universidad de la Republica, Montevideo, Uruguay Article History: Received 10 August 2009; Revised 13 January 2010; Accepted 23 January 2010 Article Note: (miscellaneous) Responsible Editor: A. Popescu

Partitioning of large networks is vital for decentralized management and control. This paper presents two algorithms called 'Hierarchical Recursive Progression-1' (HRP-1) and 'Hierarchical Recursive Progression-2' (HRP-2) for partitioning of large networks into subnetworks of limited size with very few interconnections between them. In other words, we are trying to maximize the internal nodes and minimize the external connections of the subnetworks. The restriction on the size and the external connections is obtained by comparison against a user-defined value for the size of the subnetwork and for external connections via a term called density. The density of a subnetwork is defined as the ratio of the number of external connections and the size of the subnetwork. The two algorithms presented in the paper are based on the principle of subnetwork clustering. At the start of the algorithms, the number of subnetworks is equal to the total number of nodes of the network with each subnetwork containing a single node. Later, subnetworks are merged at various runs of the algorithm to form new subnetworks using connectivity, density and size criteria. The algorithms terminate when all the subnetworks satisfy a user-defined size and density limit. The difference between the algorithms HRP-1 and HRP-2 lies in the definition of density of subnetworks and the way through which the subnetworks are grouped at consecutive iterations of the algorithm. Note that the number of nodes inside the subnetworks never violates the size limit, thereby ensuring even distribution of load on the partitions obtained. Finally, the two algorithms are compared and tested on randomly generated graphs and a part of Paris road Network. Journal of the Operational Research Society (2010) 61, 1159-1167. doi: 10.1057/jors.2009.52 Published online 24 June 2009 Keywords: graph partitioning; large networks; distributed computing; load balancing

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.comnet.2009.05.017 Byline: Carlos J. Bernardos, Maria Calderon, Ignacio Soto, Ana Beatriz Solana, Kilian Weniger Keywords: Community networks; Wireless mesh networks; Experimental evaluation; PACMAN Abstract: Wireless mesh networks are experiencing rapid progress and inspiring numerous applications in different scenarios, due to features such as autoconfiguration, self-healing, connectivity coverage extension and support for dynamic topologies. These particular characteristics make wireless mesh networks an appropriate architectural basis for the design of easy-to-deploy community or neighbourhood networks. One of the main challenges in building a community network using mesh networks is the minimisation of user intervention in the IP address configuration of the network nodes. In this paper we first consider the process of building an IP-based mesh network using typical residential routers, exploring the options for the configuration of their wireless interfaces. Then we focus on IP address autoconfiguration, identifying the specific requirements for community mesh networks and analysing the applicability of existing solutions. As a result of that analysis, we select PACMAN, an efficient distributed address autoconfiguration mechanism originally designed for ad-hoc networks, and we perform an experimental study - using off-the-shelf routers and assuming worst-case scenarios - analysing its behaviour as an IP address autoconfiguration mechanism for community wireless mesh networks. The results of the conducted assessment show that PACMAN meets all the identified requirements of the community scenario. Author Affiliation: Departamento de Ingenieria Telematica, Universidad Carlos III de Madrid, Avda. Universidad, 30, 28911 Leganes, Madrid, Spain Article Note: (footnote) [star] The research of UC3M authors leading to these results has received funding from the European Community's Seventh Framework Programme (FP7/2007-2013) under Grant Agreement No. 214994 (CARMEN Project)., [star][star] The work of UC3M authors was also partially supported by the Spanish Government under the POSEIDON (TSI2006-12507-C03-01) Project.

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.comnet.2009.06.002 Byline: Ian W.C. Lee, Abraham O. Fapojuwo Keywords: Wireless TCP; Long-range dependence; Multifractals; Cascades; IP; Heavy-tailed distribution Abstract: In this paper we analyzed and modeled wireless TCP/IP traffic. Specifically, we focused on the interarrival times of TCP flows and the number of packets within a flow. We show that the marginal distribution of the flow interarrival times is piecewise Weibull distributed. Second and higher order statistics show that the flow interarrival times are long-range dependent and exhibit multifractal scaling. Taking these higher order properties into consideration, we proposed a multinomial canonical cascade with 3 stages to model the flow interarrival times. Looking at the IP layer, we find that the number of packets in a flow is heavy-tailed distributed. Especially interesting is that in 2 of our data sets, the number of packets in a flow possesses infinite mean. The interarrival time of packets within a flow is highly correlated, bursty, and its statistical characteristics vary from flow to flow. Author Affiliation: Department of Electrical and Computer Engineering, University of Calgary, Calgary, AB, Canada T2N 1N4 Article History: Received 23 July 2007; Revised 10 March 2009; Accepted 8 June 2009 Article Note: (miscellaneous) Responsible Editor: R. Sivakumar

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.comnet.2009.05.010 Byline: Ben-Jye Chang (a), Shu-Yu Lin (b), Jun-Yu Jin (b) Keywords: TCP; Logarithmic Increase Adaptive Decrease; RTT-based; Shrink factor; Congestion control; Wireless error links; Heterogeneous wireless networks Abstract: For accessing plentiful resources in the Internet through wireless mobile hosts, diverse wireless network standards and technologies have been developed and progressed significantly. The most successful examples include IEEE 802.11 WiFi for wireless networks and 3G/HSDPA/HSUPA for cellular communications. All IP-based applications are the primary motivations to make these networks successful. In TCP/IP transmissions, the TCP congestion control operates well in the wired network, but it is difficult to determine an accurate congestion window in a heterogeneous wireless network that consists of the wired Internet and various types of wireless networks. The primary reason is that TCP connections are impacted by not only networks congestion but also error wireless links. This paper thus proposes a novel adaptive window congestion control (namely Logarithmic Increase Adaptive Decrease, LIAD) for TCP connections in heterogeneous wireless networks. The proposed RTT-based LIAD has the capability to increase throughput while achieving competitive fairness among connections with the same TCP congestion mechanism and supporting friendliness among connections with different TCP congestion control mechanisms. In the Congestion Avoidance (CA) phase, an optimal shrink factor is first proposed for Adaptive Decreasing cwnd rather than a static decreasing mechanism used by most approaches. Second, we adopt a Logarithmic Increase algorithm to increase cwnd while receiving each ACK after causing three duplicate ACKs. The analyses of congestion window and throughput under different packet loss rate are analyzed. Furthermore, the state transition diagram of LIAD is detailed. Numerical results demonstrate that the proposed LIAD outperforms other approaches in goodput, fairness, and friendliness under diverse heterogeneous wireless topologies. Especially, in the case of 10% packet loss rate in wireless links, the proposed approach increases goodput up to 156% and 1136% as compared with LogWestwood+ and NewReno, respectively. Author Affiliation: (a) Department of Computer Science and Information Engineering, National Yunlin University of Science and Technology, Taiwan, ROC (b) Department of Computer Science and Information Engineering, Chaoyang University of Technology, Taiwan, ROC Article History: Received 23 March 2008; Revised 21 March 2009; Accepted 13 May 2009 Article Note: (footnote) [star] This research was supported in part by the National Science Council of Taiwan, ROC, under Contracts NSC-96-2221-E-324-019 and NSC-97-2221-E-224-072-MY3.

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.comnet.2009.05.004 Byline: Hui Zang (a), Antonio Nucci (b) Keywords: NetFlow; CMON; Network monitor; Integer Linear Programming; Optimal placement Abstract: This paper investigates the problem of deploying network traffic monitors with optimized coverage and cost in an IP network. Deploying a network-wide monitoring infrastructure in operational networks is necessary for practical reasons. We investigate two representative solutions, a router-based solution called NetFlow and an interface-based solution called CMON. Several cost factors are associated with deploying either NetFlow or CMON in a network. We argue that enabling monitoring to cover a major portion of traffic instead of the entire traffic will achieve significant cost savings while at the same time give operators enough insight to their network. We use NetFlow as an example and develop a technique to achieve the optimal cost-coverage tradeoff. Specifically, we aim to solve the Optimal NetFlow Location Problem (ONLP) for a given coverage ratio. We analyze various cost factors to enabling NetFlow in such a network. We model the problem as an Integer Linear Program (ILP). We develop two greedy heuristics to cope with such problems of large scales given its NP-hard nature. The performance of the ILP and heuristics is demonstrated by numerical results and the LM heuristic is able to achieve sub-optimal solutions within 1-2% difference from the optimal solutions in a mixed router environment. It is observed that we can achieve 55% cost savings by covering 95% instead of 100% of the network traffic. We then extend our methodology to deploying CMON into such a network. The associated cost with deploying NetFlow and CMON is compared. The results demonstrate that CMON is more cost-effective when a small coverage ratio is desired because of its more modular nature. Author Affiliation: (a) Sprint, 1 Adrian Court, Burlingame, CA, United States (b) Narus, Inc., 500 Logue Avenue, Mountain View, CA, United States Article History: Received 21 October 2008; Revised 6 May 2009; Accepted 8 May 2009 Article Note: (footnote) [star] A partial version of the paper was published at ITC'19, 2005 and part of the work was done while Antonio Nucci was at Sprint.

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.comnet.2009.05.003 Byline: Alice Este, Francesco Gringoli, Luca Salgarelli Keywords: Traffic classification; Machine Learning Abstract: Support Vector Machines (SVM) represent one of the most promising Machine Learning (ML) tools that can be applied to the problem of traffic classification in IP networks. In the case of SVMs, there are still open questions that need to be addressed before they can be generally applied to traffic classifiers. Having being designed essentially as techniques for binary classification, their generalization to multi-class problems is still under research. Furthermore, their performance is highly susceptible to the correct optimization of their working parameters. In this paper we describe an approach to traffic classification based on SVM. We apply one of the approaches to solving multi-class problems with SVMs to the task of statistical traffic classification, and describe a simple optimization algorithm that allows the classifier to perform correctly with as little training as a few hundred samples. The accuracy of the proposed classifier is then evaluated over three sets of traffic traces, coming from different topological points in the Internet. Although the results are relatively preliminary, they confirm that SVM-based classifiers can be very effective at discriminating traffic generated by different applications, even with reduced training set sizes. Author Affiliation: DEA, Universita degli Studi di Brescia, via Branze, 38, 25123 Brescia, Italy Article History: Received 29 July 2008; Accepted 8 May 2009 Article Note: (footnote) [star] This work was supported in part by a grant from the Italian Ministry for University and Research (MIUR), under the PRIN project RECIPE.

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.comnet.2008.12.024 Byline: Mingyu Chen (a), Jinsong Zhang (b), Manohar N. Murthi (c), Kamal Premaratne (c) Keywords: Deterministic network calculus; Congestion control; Reverse path congestion; Transport protocols; High-speed networking Abstract: In delay-based TCP congestion avoidance mechanisms, a source adjusts its window size to adapt to changes in network conditions as measured through changing queueing delays. Although network calculus (NC) has been used to study window flow control and determine performance bounds, there is a lack of a bridge between NC theory and the practical issues of delay-based TCP congestion avoidance. In this paper, we use an NC-based approach to derive ideal congestion controllers for representative delay-based window flow control models with time-variant feedback delays. We show that the basic delay-based TCP congestion avoidance mechanisms in TCP Vegas, Enhanced TCP Vegas, and FAST TCP can be viewed as different approaches to approximating a certain NC controller. Moreover, we derive another NC controller that is explicitly constructed to address the throughput degradation of the current delay-based methods due to delayed acknowledgement (ACK) packets caused by network traffic in the reverse path (ACK packet) direction. The approximation of this F-model NC-based controller is shown to provide better throughput and fairness over the existing delay-based methods for a variety of network topologies in ns-2 simulations. Author Affiliation: (a) Skype, Stockholm, Sweden (b) Department of Electrical Engineering, University of Tulsa, OK, United States (c) Department of Electrical and Computer Engineering, University of Miami, McArthur Engineering Bldg. 406, 1251 Memorial Dr., Coral Gables, FL 33146, United States Article History: Received 12 April 2008; Revised 7 December 2008; Accepted 23 December 2008 Article Note: (miscellaneous) Responsible Editor: S. Mascolo

Large amounts of data and limited bandwidth are always at odds for digital video streaming over Internet Protocol (IP) networks. Packets could possibly be delayed or lost. Sender-based recovery techniques for packet loss generate redundant information that causes lower compression ratio and consumes additional bandwidth. In order to address this issue, a novel information hiding based recovery methodology has been developed. The basic idea is that the redundant information used for error recovery is embedded within the frames of the original video contents, by means of high bitrate information hiding techniques. In the receiver end, the hidden information is extracted for lost-packet recovery. Experimental results show that the damaged macroblocks can be recovered with a higher quality using the proposed methodology, as compared to spatial extrapolation used in H.264/AVC reference software. The main contributions of the proposed methodology are: 1) the delivery of redundant information does not increase bandwidth requirement significantly under the same media quality, bringing out advantages in practical applications; 2) the structures of encoder and decoder do not need to be changed in the case of intra-only coding mode. Index Terms--Embedding, recovery, redundancy, video.

We propose an efficient binary search algorithm for IP address lookup in the Internet routers. While most of the previous binary search algorithms do not provide a balanced search, the proposed algorithm provides a perfectly balanced search, and hence it provides excellent search performance and scalability toward large routing tables. Index Terms--Internet, router, IP address lookup, binary trie, binary search tree.

In this paper, we propose an analytical Markov chain model of the slotted carrier-sense multiple-access/collision-avoidance (CSMA/CA) protocol for IEEE 802.15.4 under unsaturated traffic conditions. Our proposed Markov chain model reflects the characteristics of the IEEE 802.15.4 medium-access control (MAC) protocol, such as a superframe structure, acknowledgements, and retransmissions with and without limit. We evaluate the throughput performance of the slotted CSMA/CA and verify the analytical model using simulation results. Index Terms--Carrier sense multiple access/collision avoidance (CSMA/CA), IEEE 802.15.4, medium-access control (MAC), throughput.

To secure a mobile ad hoc network (MANET) in adversarial environments, a particularly challenging problem is how to feasibly detect and defend possible attacks on routing protocols, particularly internal attacks, such as a Byzantine attack. In this paper, we propose a novel algorithm that detects internal attacks by using both message and route redundancy during route discovery. The route-discovery messages are protected by pairwise secret keys between a source and destination and some intermediate nodes along a route established by using public key cryptographic mechanisms. We also propose an optimal routing algorithm with routing metric combining both requirements on a node's trustworthiness and performance. A node builds up the trustworthiness on its neighboring nodes based on its observations on the behaviors of the neighbor nodes. Both of the proposed algorithms can be integrated into existing routing protocols for MANETs, such as ad hoc on-demand distance vector routing (AODV) and dynamic source routing (DSR). As an example, we present such an integrated protocol called secure routing against collusion (SRAC), in which a node makes a routing decision based on its trust of its neighboring nodes and the performance provided by them. The simulation results have demonstrated the significant advantages of the proposed attack detection and routing algorithm over some known protocols. Index Terms--Ad hoc network, mobile, routing protocol, security.

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.compeleceng.2008.05.006 Byline: Wei Zhang (a), Liansheng Tan (a)(b), Gang Peng (a) Keywords: Random early detection (RED); Active queue management (AQM); Stability; Hermite matrix; Congestion control Abstract: One main TCP congestion control objective is, by dynamically adjusting the source window size according to the router queue level, to stabilize the buffer queue length at a given target, thereby achieving predictable queueing delay, reducing packet loss and maximizing link utilization. One difficulty therein is the TCP acknowledging actions will experience a time delay from the router to the source in a TCP system. In this paper, a time-delay control theory is applied to analyze the mechanism of packet-dropping at router and the window-updating in TCP source in TCP congestion control for a TCP/RED dynamic model. We then derive explicit conditions under which the TCP/RED system is asymptotically stable in terms of the instantaneous queue. We discuss the convergence of the buffer queue lengths in the routers. Our results suggest that, if the network parameters satisfy certain conditions, the TCP/RED system is stable and its queue length can converge to any target. We illustrate the theoretical results using ns2 simulations and demonstrate that the network can achieve good performance and converge to the arbitrary target queues. Author Affiliation: (a) Department of Computer Science, Central China Normal University, Wuhan 430079, PR China (b) Research School of Information Sciences and Engineering, The Australian National University, Canberra ACT 0200, Australia Article History: Received 12 June 2007; Revised 27 April 2008; Accepted 8 May 2008

To link to full-text access for this article, visit this link: http://dx.doi.org/10.1016/j.compeleceng.2008.02.002 Byline: Hyun C. Cho (a), Sami M. Fadali (a), Hyunjeong Lee (b) Abstract: Active Queue Management (AQM) is a proven strategy to efficiently maintain queues and ensure high utilization of Transmission Control Protocol (TCP) network resources. The fundamental mechanism is to manage incoming packet rates at a router to prevent incipient network congestion. In this paper, we present an efficient neural network AQM system as a queue controller. The recurrent neural network has a Multi-layer Perceptron-Infinite Impulse Response (MLP-IIR) structure. Three distinct neural AQMs are trained under different network scenarios involving traffic levels. Selecting one of three neural AQMs is based on posterior probability history of traffic level. In addition, we investigate stochastic modeling of the network dynamics by a Dynamic Bayesian Network (DBN). This model allows implementation of a predictive AQM system in which queue dynamics are predicted and used for error prediction via online DBN estimation. Our AQM method is evaluated through simulation experiments both using an Ordinary Differential Equation (ODE) network model and using OPNET.sup.A[c]. The simulation results demonstrate that our adaptive neural AQM outperforms Random Early Detection (RED) and Proportional-Integral-Derivative (PID) based AQM. Author Affiliation: (a) Department of Electrical Engineering/260, University of Nevada, Reno, NV 89557, United States (b) Intel Corporation, Wireless Networking Laboratory, 2111 25th Avenue, Hillsboro, OR 97124, United States Article History: Received 9 August 2006; Accepted 21 February 2008

A timing model is described that has required timeliness on O(n) links in stable periods and allows unbounded periods of asynchrony. The results have shown that a protocol requiring fewer timely links can achieve better performance, even if it sends fewer messages.

A novel scheme is described for improving the practicality of log-based IP traceback by reducing its overhead on routers. The results have shown that this new method has maintained the ability to trace a single IP packet while reducing the storage overhead by half and the access time overhead by a factor of the number of neighboring routers.

Traditional operating system interfaces and network protocol implementations force some system state to be kept on both sides of a connection. This state ties the connection to its endpoints, impedes transparent failover, permits denial-of-service attacks, and limits scalability. This article introduces a novel TCP-like transport protocol and a new interface to replace sockets that together enable all state to be kept on one endpoint, allowing the other endpoint, typically the server, to operate without any per-connection state. Called Trickles, this approach enables servers to scale well with increasing numbers of clients, consume fewer resources, and better resist denial-of-service attacks. Measurements on a full implementation in Linux indicate that Trickles achieves performance comparable to TCP/IP, interacts well with other flows, and scales well. Trickles also enables qualitatively different kinds of networked services. Services can be geographically replicated and contacted through an anycast primitive for improved availability and performance. Widely-deployed practices that currently have client-observable side effects, such as periodic server reboots, connection redirection, and failover, can be made transparent, and perform well, under Trickles. The protocol is secure against tampering and replay attacks, and the client interface is backward-compatible, requiring no changes to sockets-based client applications. Categories and Subject Descriptors: C.2.0 [Computer-Communication Networks]: General--Security and protection; C.2.2 [Computer-Communication Networks]: Network Protocols--Protocol architecture; C.2.4 [Computer-Communication Networks]: Distributed Systems--Client/server; C.2.5 [Computer-Communication Networks]: Local and Wide-Area Networks--Internet; D.4.4 [Operating Systems]: Communications Management; D.4.7 [Operating Systems]: Organization and Design General Terms: Design, Performance Additional Key Words and Phrases: Stateless interfaces, stateless protocols ACM Reference Format: Shieh, A., Myers, A. C., and Sirer, E. G. 2008. A stateless approach to connection-oriented protocols. ACM Trans. Comput. Syst. 26, 3, Article 8 (September 2008), 50 pages. DOI=10.1145/ 1394441.1394444 http://doi.acm. org/10.1145/1394441.1394444