In multivariate long-term time series forecasting, the success of self-attention is commonly attributed to the attention matrix that encodes token interactions.In this paper, we provide evidence that challenges this view.Through extensive experiments on three classic and three latest Transformer models, we find that dotproduct attention can be replaced by elementwise operations without token interaction, such as the addition and Hadamard product, while maintaining or even improving accuracy.This motivates our central hypothesis: the effectiveness of self-attention in this task arises not from the dynamic attention matrix, but from the multi-branch feature extraction enabled by the parallel Query, Key, and Value projections and their fusion.To validate this hypothesis, we construct a minimalist multi-branch MLP that isolates the 'multi-branch mapping with element-wise operation' structure from the Transformer and show that it achieves competitive performance.Our findings indicate that the source of performance in self-attention is often misinterpreted, as its actual advantage stems from the architectural principle of multi-branch mapping and fusion, rather than the attention matrix.

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Blockchain faces significant challenges in scalability, user practicality, and regulatory compliance, especially ason-chain data grows. Stateless blockchains (SBCs) reduce node storage by moving stateoff-chain, but introduce critical issues: they require frequent witness updates that force users to remain online, and prevent regulatory nodes from independently verifying transaction histories. To address these limitations, we propose BMchain, an end-to-end SBC that innovates in data structure and collaboration model. BMchain maintains a lightweight main chain (under 2 MB per block) by recording only Bloom filters and Merkle roots of transactions. This design allows users to transact without being constantly online and enables regulators to verify data completeness and correctness solely fromon-chain information. Experimental results show BMchain achieves over 10 000 TPS throughput, reduces annual account storage to around 200 MB, and confirms transactions within 2 min at scale—significantly outperforming existing solutions. Unlike accumulator-based approaches, BMchain fundamentally decouples local state updates from global ledger changes, offering a practical and regulatory-friendly SBC architecture.

Detecting social bots is crucial for mitigating the spread of misinformation and preserving online conversation authenticity. State-of-the-art solutions typically leverage graph neural networks (GNNs) to model user representations from social relationships and metadata. However, these approaches overlook two key factors: the similarity of a user and her neighbors, as well as the coordinated behaviors of social bots, resulting in a suboptimal detection performance. To address these issues, we propose HyperScan, a novel representation learning method for social bot detection. Specifically, we introduce three effective learners to capture pair-wise, hop-wise, and group-wise relations. HyperScan learns pair-wise user representations based on social relations and user features. It then enhances user representations by building hop-wise interactions across the learned pair-wise user representations for capturing the structure-level proximity information. Subsequently, it models user representations by constructing higher-order (group-wise) relations derived from user profiles, tweets, and social relations to capture the feature-level proximity knowledge. By leveraging hop-wise interactions and higher-order relations, HyperScan significantly improves bot detection performance. Our extensive experiments demonstrate that HyperScan outperforms state-of-the-art methods on three benchmark datasets. Additional studies validate the robustness and effectiveness of each component of HyperScan.

Recently, decentralized online social networks (DOSNs) such as Mastodon have emerged quickly, bringing new opportunities for studies in user behavior modeling and multi-modal learning. However, their decentralized architecture presents two key challenges: 1) Distributed data and inconsistent access strategies across several individual instances make a unified collection difficult; 2) user-generated content (UGC) contains multiple modalities while lacking standard organization and high-quality annotation. To address these issues, we constructed FediData, a comprehensive multi-modal dataset from Mastodon. Our dataset integrates user profiles, text, images, and social interactions. To validate FediData's usefulness, we designed and analyzed several tasks and systematically evaluated the performance of existing state-of-the-art methods. Our analysis reveals the unique challenges of DOSNs and highlights the value of FediData in DOSN-related studies. We believe FediData could serve as a foundational dataset for advancing user behavior analytics, multi-modal learning, and future decentralized web research. All data and documentation are available in a Zenodo repository at https://zenodo.org/records/15621243 (DOI: 10.5281/zenodo.15621243).

Time series forecasting is always important, with multivariate long-term series forecasting being its most challenging task. Here, the existing methods typically learn only in a single domain and focus on optimizing model structures, leading to incomplete information mining and imprecise predictions. To address this, we propose a generalized Mixture of Multi-Domain Experts (MoME) for multivariate long-term series forecasting. Unlike most existing methods, MoME focuses on multi-perspective information mining and fusing. To this end, MoME transforms time series into the frequency and spatial domains to learn their respective representations. MoME regards variates information as embedded features and applies fast Fourier transform to the time dimension. Then it learns embedded features in the frequency domain. In spatial domain learning, MoME applies self-attention mechanism on the variates dimension to efficiently capture dependencies among multiple variates. Finally, MoME fuses the outputs from all domains, reinterprets and integrates information across multiple domains, and predicts future time series. Extensive experiments prove that MoME outperforms state-of-the-art (SOTA) methods. Code is available at: https://github.com/lxy-PhD2022/MoME

Spiking Neural Network (SNN) simulation involves emulating the activation and firing of spiking neurons on hardware platforms. This is a highly time-sensitive task, requiring the simulation of billions of neurons and their intercommunication within a few milliseconds. Each neuron performs a complex, interdependent multi-stage communication and computation task. We consider the task placement of SNN on supercomputers to accelerate SNN simulation. Existing task placement methods for SNN simulations have two major limitations. First, they lack the capability to handle large-scale SNNs with billions of neurons. Second, they focus primarily on optimizing communication delay, while neglecting multi-stage computation delays in SNN simulations. In this paper, we formalize the SNN Joint Multi-stage Communication and Computation Placement (SJCCP) problem. We demonstrate that SJCCP can be solved using an approximation algorithm with an approximation ratio of $O\left( {{k^2}\sqrt {\log n\log k} } \right)$, where n is the number of voxels in the SNN and k is the number of GPUs. To further reduce the time complexity of solving SJCCP in practice, we propose a novel efficient framework, FastSJP, tailored for large-scale SNN placement. Then we apply the FastSJP framework to a human brain simulation that runs a large-scale SNN model derived from authentic biological data on a supercomputer equipped with 1024 GPUs. Experimental results verify that our framework notably reduces time overhead, ranging from 17.31% to 28.45%, compared to state-of-the-art methods. Leveraging the computational power of the supercomputer, FastSJP maximizes the problem size and processing performance, significantly advancing the development of brain-inspired intelligence.

Multi-scale analysis, a classical and crucial methodology, is extensively employed in multivariate long-term time series forecasting. However, the existing methods struggle to model arbitrary scales, thus limiting their ability to delve into complex patterns, which in turn limits the forecasting precision. Therefore, we propose the HMSformer, a multi-scale Transformer that includes a shifted stacked embedding mechanism and a unified hierarchical framework, solving this problem in terms of the in-depth and comprehensiveness of multi-scale analysis. The mechanism conducts multi-scale modeling by sampling at different positions and sizes. This sampling reflects the arbitrariness of scales and allows for a thorough scan of any scale, supporting a deeper analysis of temporal relationships. Additionally, it converts input series into structured embeddings optimized for Transformer input, allowing the self-attention mechanism to effectively learn intricate cross-scale relationships. Moreover, the unified hierarchical framework integrates multi-scale modeling across different levels of abstraction by unifying global structures, local segments, and fine details into a consistent representation. Together, these two innovations ensure that HMSformer achieves a more comprehensive and in-depth analysis of temporal relationships. HMSformer demonstrates its effectiveness with a significant improvement, achieving an average mean squared error (MSE) that is 9.925% lower than the baseline across nine authoritative datasets. Source code is available at: https://github.com/lxy-PhD2022/HMSformer

Facing the key challenge in multivariate long-term series prediction, namely effectively modeling the long-term dependencies among high-dimensional variables, we propose an innovative forest network (ForeNet). Firstly, we construct a polytree, which takes variable as leaf node, convolution as edge, and progressive fusion as the root node. Polytree models the dependencies among variables from the bottom up, adopting a local-to-global progressive learning strategy. Moreover, polytree embeds the entire sequence as input channels for convolution, allowing interactions across arbitrary time steps between variables. Thereby, polytree could model long-term dependencies. Then, we construct multiple polytrees with varying branching factors, utilizing the self-attention mechanism to assign weights and combine polytrees into a forest. Forest adopts an ensemble learning strategy to capture complex patterns hidden under high-dimensional variables. Combining progressive and ensemble strategies, ForeNet could effectively address the key challenge mentioned at the beginning. Extensive experiments show that ForeNet could reduce the average MSE by up to 11.53% compared to the baseline, which verifies the effectiveness of ForeNet. Source code is available at: https://github.com/lxy-PhD2022/ForeNet

User interface (UI) display issues, such as widgets occlusion, missing elements, and screen overflow, are emerging as a non-negligible source of user complaints in commercial mobile apps. However, existing automated testing tools typically rely on a vast amount of high-quality training data, making them cost-ineffective for industrial practice. Given that display issues are intuitively recognizable by humans, their diverse appearances can be abstracted by the violation of human commonsense expectations of UI appearance. Therefore, this paper proposes to reduce data requirements in display issue detection through commonsense simulation. Although leveraging large vision-language models (VLMs) to replicate human visual ability looks straightforward, off-the-shelf VLMs lack task-specific knowledge of UI designs and display correctness. To address this, we fine-tune a VLM to learn what constitutes an expected display and to reason potential display issues. This approach is termed as Minuku, an industrial data-efficient UI display issue detector. We evaluate the design effectiveness of Minuku via a set of ablation experiments. Moreover, real-world deployments in one of the largest E-commerce app providers further demonstrate that Minuku can effectively detect 40 previously unknown UI display issues and significantly reduce manual effort in industrial settings.

Mobile technology is being increasingly adopted in teleconsultation for its convenience and mobility. Although widely accepted by physicians for informal online consultations, the effectiveness of mobile platforms in formal medical interventions, such as follow-up treatments, remains largely underexplored. This research presents a case study from a Chinese hospital, examining how physicians use mobile platforms to treat chronic patients, the challenges they encounter, and their overall experience. The study aims to determine whether mobile technology can enable physicians to fulfill their responsibilities in managing chronic disease follow-ups online. Through observations and interviews, we found that using mobile platforms introduces significant challenges. Physicians operate in complex and varying environments, such as workplaces, homes, and even public areas, making them struggle to access comprehensive information, communicate effectively, and maintain detailed medical records. As a result, physicians face poor working conditions, reduced efficiency, and struggle to make accurate treatment decisions. This study highlights the need for policy reforms and technological innovations to ensure sustainable teleconsultation practices on mobile platforms. It contributes to the HCI and CSCW communities by highlighting the feasibility of using mobile platforms in online chronic disease follow-ups.

Influential users in online social networks have been given a lot of attention, which could maximize information diffusion in a centralized dissemination process. However, there exist users that can significantly promote diversified information communication among users, which is a pluralistic interaction process. Structural hole (SH) spanners serve as a bridge between social groups, promoting interactions between users and allowing ordinary users to receive diverse information. Filling structural holes, SH spanners have more opportunities to obtain innovative content. Identifying SH spanners is challenging due to their inconspicuous characteristics and privacy policy restrictions. In this article, we are the first to study the SH spanner identification problem using the behavior data generated by users, releasing the dependency on the entire graph structure to find SH spanners. We propose DeepHole to design both the semantic and the sequential behavior analysis modules on users’ generated content, utilizing the TextCNN and Transformer models, respectively. The semantic and sequential embeddings are effective in distinguishing the SH spanners from ordinary users. We conduct comprehensive evaluations using the Yelp and Foursquare datasets simultaneously. Results show that DeepHole can achieve a satisfying performance in detecting SH spanners labeled by both constraint and effective size metrics, with AUC values of 0.923 and 0.933 for Yelp and 0.828 and 0.815 for Foursquare.

The rapid evolution of the Internet-of-Vehicles (IoV) has amplified the need for mobile computing resources, driving the shift toward offloading tasks to edge servers or vehicles with idle resources to optimize computational efficiency. To this end, an approach based on Deep Reinforcement Learning (DRL) is presented in this paper, termed DVTP, which integrates Variational Graph Attention Networks (VGAT) and Transformer models to optimize Directed Acyclic Graph (DAG) task scheduling in vehicular networks. DVTP effectively captures both the spatiotemporal information and task dependencies, enabling more accurate and efficient task offloading decisions. Extensive simulation experiments demonstrate that DVTP outperforms traditional methods in reducing task completion times across various multi-vehicle and multi-edge server scenarios, showcasing its potential for real-world IoV applications.

HTTP/3 (H3) has experienced significant growth and extensive adoption in various scenarios, especially in Content Delivery Networks (CDNs). Over the past few years, there have been numerous insightful studies on its deployment in industrial CDNs. However, these studies often separately analyze H3 and CDN, overlooking their synergistic integration. In this work, we explore the applicability of H3 in CDN from a holistic perspective. We analyze 325 websites hosted by seven CDN providers and identify three key characteristics where CDN align perfectly with H3's strengths. Firstly, CDN resources dominate the composition of webpages, where enabling H3 can amplify H3's benefits in connection acceleration. Secondly, CDN providers also exhibit a dominant characteristic, with the majority of CDN resources hosted by a few large providers. This phenomenon makes different webpages share the same provider. When browsing consecutively, H3 helps to skip the connection phase by resuming the connections to the same CDN provider across pages. Thirdly, H3 mitigates the congestion problem on webpages serving multiple CDN resources. This work provides a deeper insight into the applicability of H3 in large-scale distributed systems like CDNs, holding promise for informing the development and optimization of industrial H3.

IPv6, with its larger address space, is well-suited for IoT scenarios, allowing each device to have a unique address for efficient communication. However, its adoption in sensor networks is hindered by underdeveloped IPv6 infrastructure and compatibility issues with IPv4-based networks. To address these challenges, this paper proposes a dual-stack load balancing approach to enhance seamless and reliable connectivity between IPv6 and IPv4. By adaptively distributing traffic between IPv6 and IPv4 channels, our system alleviates performance bottlenecks and improves reliability. A real-world deployment in the campus network of Fudan University demonstrates a 1.9x increase in throughput and a 27% reduction in congestion-related failures with only CPU overhead of 2.44% and a translation delay of 0.58 ms. These results highlight the potential of intelligent load balancing and IPv4/IPv6 translation in supporting scalable and efficient sensor networks and facilitating the transition to IPv6.

Disaggregated memory (DM) is a widely discussed datacenter architecture in academia and industry. It decouples computing and memory resources from monolithic servers into two network-connected resource pools. Range indexes are widely adopted by storage systems on DM to efficiently locate and query remote data. However, existing range indexes on DM suffer from either high computing-side cache consumption or high memory-side read amplifications. In this paper, we propose CHIME, a hybrid index combining B+ trees with hopscotch hashing, to achieve low cache consumption and low read amplifications simultaneously. There are three challenges in constructing CHIME on DM, i.e., the complicated optimistic synchronization, the extra metadata access, and the read amplifications introduced by hopscotch hashing. CHIME leverages 1) a three-level optimistic synchronization scheme to synchronize read and write operations with various granularities, 2) an access-aggregated metadata management technique to eliminate extra metadata accesses by piggybacking and replicating metadata, and 3) an effective hotness-aware speculative read mechanism to mitigate the read amplifications of hopscotch hashing. Experimental results show that CHIME outperforms the state-of-the-art range indexes on DM by up to 5.1× with the same cache size and achieves similar performance with up to 8.7× lower cache consumption.

The temporal patterns of code submissions, denoted as work rhythms, provide valuable insight into the work habits and productivity in software development. In this paper, we investigate the work rhythms in software development and their effects on technical performance by analyzing the profiles of developers and projects from 110 international organizations and their commit activities on GitHub. Using clustering, we identify four work rhythms among individual developers and three work rhythms among software projects. Strong correlations are found between work rhythms and work regions, seniority, and collaboration roles. We then define practical measures for technical performance and examine the effects of different work rhythms on them. Our findings suggest that moderate overtime is related to good technical performance, whereas fixed office hours are associated with receiving less attention. Furthermore, we survey 92 developers to understand their experience with working overtime and the reasons behind it. The survey reveals that developers often work longer than required. A positive attitude towards extended working hours is associated with situations that require addressing unexpected issues or when clear incentives are provided. In addition to the insights from our quantitative and qualitative studies, this work sheds light on tangible measures for both software companies and individual developers to improve the recruitment process, project planning, and productivity assessment.

Phone number recycling (PNR) refers to the event wherein a mobile operator collects a disconnected number and reassigns it to a new owner. It has posed a threat to the reliability of the existing authentication solution for e-commerce platforms. Specifically, a new owner of a reassigned number can access the application account with which the number is associated, and may perform fraudulent activities. Existing solutions that employ a reassigned number database from mobile operators are costly for e-commerce platforms with large-scale users. Thus, alternative solutions that depend on only the information of the applications are imperative. In this work, we study the problem of detecting accounts that have been compromised owing to the reassignment of phone numbers. Our analysis on Meituan’s real-world dataset shows that compromised accounts have unique statistical features and temporal patterns. Based on the observations, we propose a novel model called temporal pattern and statistical feature fusion model (TSF) to tackle the problem, which integrates a temporal pattern encoder and a statistical feature encoder to capture behavioral evolutionary interaction and significant operation features. Extensive experiments on the Meituan and IEEE-CIS datasets show that TSF significantly outperforms the baselines, demonstrating its effectiveness in detecting compromised accounts due to reassigned numbers.

Human mobility trajectories are fundamental resources for analyzing mobile behaviors in urban computing applications. However, these trajectories, typically collected from location-based services, often suffer from sparsity and irregularity in time. To support the development of mobile applications, there is a need to recover or estimate missing locations of unobserved time slots in these trajectories at a fine-grained spatial-temporal resolution. Existing methods for trajectory recovery rely on either individual user trajectories or collective mobility patterns from all users. The potential to combine individual and collective patterns for precise trajectory recovery remains unexplored. Additionally, current methods are sensitive to the heterogeneous temporal distributions of the observable trajectory segments. In this paper, we propose CLMove (where CL stands for contrastive learning), a novel model designed to capture multilevel mobility patterns and enhance robustness in trajectory recovery. CLMove features a two-stage location encoder that captures collective and individual mobility patterns. The graph neural network based networks in CLMove explore location transition patterns within a single trajectory and across various user trajectories. We also design a trajectory-level contrastive learning task to improve the robustness of the model. Extensive experimental results on three representative real-world datasets demonstrate that our CLMove model consistently outperforms state-of-the-art methods in terms of trajectory recovery accuracy.

Camouflaged object detection has been considered a challenging task due to its inherent similarity and interference from background noise. It requires accurate identification of targets that blend seamlessly with the environment at the pixel level. Although existing methods have achieved considerable success, they still face two key problems. The first one is the difficulty in removing texture noise interference and thus obtaining accurate edge and frequency domain information, leading to poor performance when dealing with complex camouflage strategies. The latter is that the fusion of multiple information obtained from auxiliary subtasks is often insufficient, leading to the introduction of new noise. In order to solve the first problem, we propose a frequency domain reconstruction module based on contrast learning, through which we can obtain high-confidence frequency domain components, thus enhancing the model’s ability to discriminate target objects. In addition, we design a frequency domain representation decoupling module for solving the second problem to align and fuse features from theRGBdomain and the reconstructed frequency domain. This allows us to obtain accurate edge information while resisting noise interference. Experimental results show that our method outperforms 12 state-of-the-art methods in three benchmark camouflaged object detection datasets. In addition, our method shows excellent performance in other downstream tasks such as polyp segmentation, surface defect detection, and transparent object detection.

CDN is a crucial Internet infrastructure ensuring quick access to Internet content. With the expansion of CDN scenarios, beyond delay, resource types like bandwidth and CPU are also important for CDN performance. Our measurements highlight the distinct impacts of various resource types on different CDN requests. Unfortunately, mainstream CDN server selection schemes only consider a single resource type and are unable to choose the most suitable servers when faced with diverse resource types. To fill this gap, we propose Polygon, a QUIC-powered CDN server selection system that is aware of multiple resource demands. Being an advanced transport layer protocol, QUIC equips Polygon with customizable transport parameters to enable the seamless handling of resource requirements in requests. Its 0-RTT and connection migration mechanisms are also utilized to minimize delays in connection and forwarding. A set of collaborative measurement probes and dispatchers are designed to support Polygon, being responsible for capturing various resource information and forwarding requests to suitable CDN servers. Real-world evaluations on the Google Cloud Platform and extensive simulations demonstrate Polygon’s ability to enhance QoE and optimize resource utilization. The results show up to a 54.8% reduction in job completion time, and resource utilization improvements of 13% in bandwidth and 7% in CPU.

Disaggregated memory (DM) is an increasingly prevalent architecture with high resource utilization. It separates computing and memory resources into two pools and interconnects them with fast networks. Existing range indexes on DM are based on B+ trees, which suffer from large inherent read and write amplifications. The read and write amplifications rapidly saturate the network bandwidth, resulting in low request throughput and high access latency of B+ trees on DM. In this article, we propose that the radix tree is more suitable for DM than the B+ tree due to smaller read and write amplifications. However, constructing a radix tree on DM is challenging due to the costly lock-based concurrency control, the bounded memory-side IOPS, and the complicated computing-side cache validation. To address these challenges, we design SMART , the first radix tree for disaggregated memory with high performance. Specifically, we leverage (1) a hybrid concurrency control scheme including lock-free internal nodes and fine-grained lock-based leaf nodes to reduce lock overhead, (2) a computing-side read-delegation and write-combining technique to break through the IOPS upper bound by reducing redundant I/Os, and (3) a simple yet effective reverse check mechanism for computing-side cache validation. Experimental results show that SMART achieves 6.1× higher throughput under typical write-intensive workloads and 2.8× higher throughput under read-only workloads in YCSB benchmarks, compared with state-of-the-art B+ trees on DM.

Online Social Networks (OSNs) play a significant role in people's daily life. Increasing OSN traffic promotes the requirement for building self-operated Content Delivery Networks (CDNs) to deliver OSN media data efficiently and reduce traffic costs. OSN data in CDNs is heavily influenced by social connectivity, such as friendships. To reduce CDN network traffic by better using social connectivity information, we propose SocialCache, a pervasive social-aware caching strategy in self-operated CDNs. SocialCache supports several social connectivity metric options that reflect the importance of the OSN users and the popularity of the media files. Through the improvement of the cache replacement algorithm in the CDN node and the communication design between nodes, SocialCache realizes the optimization of network traffic. Worth mentioning, SocialCache can easily integrate into mainstream CDN architectures while protecting user privacy. We implement SocialCache on Mininet, using real-world network measurements for CDN hierarchy and a hill-climbing algorithm for parameter selection. SocialCache outperforms a range of state-of-the-art baselines on three real-world OSN datasets. On the Twitter dataset, SocialCache reduces the network traffic volume by 6.40% and improves the cache hit ratio by 14.22%.

Online developer communities like GitHub allow a massive number of developers to collaborate. However, the openness of the communities makes them vulnerable to different types of malicious attacks, since attackers can easily join these communities and interact with legitimate users. In this work, we propose GitSec, a deep learning-based solution for detecting malicious accounts in online developer communities. GitSec distinguishes malicious accounts from legitimate ones based on the account profiles, dynamic activity characteristics, as well as social interactions. First, GitSec introduces two user activity sequences and applies a parallel neural network design with an attention mechanism to process the sequences. Second, GitSec constructs two graphs to represent the interactions between users according to their repository operations. Especially, graph neural networks and structural hole theory are employed to deal with the two constructed graphs. Third, GitSec makes use of the descriptive features to enhance the detection performance. The final judgement is made by a decision maker implemented by a supervised machine learning-based classifier. Based on the real-world data of GitHub users, our comprehensive evaluations show that GitSec achieves a better performance than state-of-the-art solutions, with an AUC value of 0.916.

Outstanding users (OUs) denote the influential, "core" or "bridge" users in the online community. How to accurately detect and rank them is an important problem for third-party online service providers and researchers. Conventional efforts, ranging from early graph-based algorithms to recent machine learning-based approaches, typically rely on an entire network's information or at least ego networks. However, for privacy-conscious users or newly-registered users, such information is not easily accessible. To address this issue, we present DeepPick, a novel framework that considers both the generalization and specialization in the detection task of OUs. For generalization, we introduce deep neural networks to capture nonlinear features. For specialization, we leverage the traditional well-defined metrics to preserve common features. Extensive experiments based on real-world datasets demonstrate that our approach achieves a high efficacy in terms of detection performance against the state-of-the-art.

Human activity recognition (HAR) plays a critical role in a wide range of real-world applications, and it is traditionally achieved via wearable sensing. Recently, to avoid the burden and discomfort caused by wearable devices, device-free approaches exploiting radio-frequency (RF) signals arise as a promising alternative for HAR. Most of the latest device-free approaches require training a large deep neural network model in either time or frequency domain, entailing extensive storage to contain the model and intensive computations to infer human activities. Consequently, even with some major advances on device-free HAR, current device-free approaches are still far from practical in real-world scenarios where the computation and storage resources possessed by, for example, edge devices, are limited. To overcome these weaknesses, we introduce HAR-SAnet which is a novel RF-based HAR framework. It adopts an original signal adapted convolutional neural network architecture: instead of feeding the handcraft features of RF signals into a classifier, HAR-SAnet fuses them adaptively from both time and frequency domains to design an end-to-end neural network model. We apply point-wise grouped convolution and depth-wise separable convolutions to confine the model scale and to speed up the inference execution time. The experiment results show that the recognition accuracy of HAR-SAnet substantially outperforms the state-of-the-art algorithms and systems.

Despite continuous efforts to build and update mobile network infrastructure, mobile devices in developing regions continue to be constrained by limited bandwidth. Unfortunately, this coincides with a period of unprecedented growth in the sizes of mobile applications. Thus it is becoming prohibitively expensive for users in developing regions to download and update mobile apps critical to their economic and educational development. Unchecked, these trends can further contribute to a large and growing global digital divide. Our goal is to better understand the source of this rapid growth in mobile app code size, whether it is reflective of new functionality, and identify steps that can be taken to make existing mobile apps more friendly to bandwidth constrained mobile networks. We hypothesize that much of this growth in mobile apps is due to poor resource/code management, and do not reflect proportional increases in functionality. Our hypothesis is partially validated by mini-programs, apps with extremely small footprints gaining popularity in Chinese mobile platforms. Here, we use functionally equivalent pairs of mini-programs and Android apps to identify potential sources of “bloat,” i.e., inefficient uses of code or resources that contribute to large package sizes. We analyze a large sample of popular Android apps and quantify instances of code and resource bloat. We develop techniques for automated code and resource trimming, and successfully validate them on a large set of Android apps. We hope our results will lead to continued efforts to streamline mobile apps, making them easier to access and maintain for users in developing regions.

Deep learning (DL) techniques are proven effective in many challenging tasks, and become widely-adopted in practice. However, previous work has shown that DL libraries, the basis of building and executing DL models, contain bugs and can cause severe consequences. Unfortunately, existing testing approaches still cannot comprehensively exercise DL libraries. They utilize existing trained models and only detect bugs in model inference phase. In this work we propose Muffin to address these issues. To this end, Muffin applies a specifically-designed model fuzzing approach, which allows it to generate diverse DL models to explore the target library, instead of relying only on existing trained models. Muffin makes differential testing feasible in the model training phase by tailoring a set of metrics to measure the inconsistencies between different DL libraries. In this way, Muffin can best exercise the library code to detect more bugs. To evaluate the effectiveness of Muffin, we conduct experiments on three widely-used DL libraries. The results demonstrate that Muffin can detect 39 new bugs in the latest release versions of popular DL libraries, including Tensorflow, CNTK, and Theano.

Online-to-Offline (O2O) e-commerce service platforms and their users are faced with various fraud risks. Among them, financial identity theft is a widely existing challenge. However, existing methods are insufficient to detect this type of fraud. In this paper, we address the financial identity theft detection problem in e-commerce services by leveraging access environment and behavior sequence. To explore the fraud patterns, we first make a detailed analysis using real cases of identity theft from Meituan, a leading O2O e-commerce platform in China. Our findings are twofold. First, fraudulent accounts sharing the same personal ID would have different access environments, such as devices and IP addresses. Second, a group of fraudulent accounts may have aggregations of devices, IP addresses, and delivery addresses. Based on these observations, we propose a hybrid method termed EnvIT to detect financial identity theft based on the heterogeneous graph and the behavior sequence. EnvIT is able to characterize the access environment and the historical behavior of the accounts. Furthermore, an attentive module is adopted to assign weights to different features automatically. We further evaluate EnvIT via extensive experiments using a real-world dataset from Meituan. Our experimental results demonstrate that EnvIt outperforms several baseline methods in fraudulent account detection and achieves an AUC of 0.9210.

Programmable switches have been proposed in today’s network to enable flexible reconfiguration of devices and reduce time-to-deployment. Buffer sizing, an important factor for network performance, however, has not received enough attention in programmable network. The state-of-the-art buffer sizing solutions usually employ either fixed buffer size or adjust the buffer size heuristically. Without programmability, they suffer from either massive packet drops or large queueing delay in dynamic environment. In this paper, we propose Adaptive Buffer Sizing (ABS), a low-cost and deploy-friendly framework compatible with programmable network. By decoupling the data plane and control plane, ABS-capable switches only need to react to the actions from controller, optimizing network performance in run-time under dynamic traffic. Meanwhile, actions can be programmed by particular Machine Learning (ML) models in the controller to meet different network requirements. In this paper, we address two specific ML models for different scenarios, a reinforcement learning model for relatively stable network with user specific quality requirements, and a supervised learning model for highly dynamic network condition. We implement the ABS framework by integrating the prevalent network simulator NS-2 with ML module. The experiment shows that ABS outperforms state-of-the-art buffer sizing solutions by up to 38.23x under various network environments.

Although IPv6 has been proposed to solve the IP address exhaustion problem for decades, the transition process from IPv4 to IPv6 is rather slow due to the possible loss of users and increased costs for ISPs compared with the potential profits. In order to accelerate this process and make full use of the IPv6 network, in this paper, we propose a user-transparent solution named NetBoost by transferring IPv4 traffic through the IPv6 core network. We also implement a simulator called NetBoostSim to further verify the usefulness and prospective performance gain of NetBoost in different network environments. By deploying our system upon both real and simulated network environments, we showcase that better performance for IPv4 end-to-end connections can be acquired by utilizing the light-loaded IPv6 network to transfer traffic from heavy-loaded IPv4 core network, using stateless IPv4/IPv6 translation techniques. In this way, our system can serve as an incentive for ISPs to upgrade to pure IPv6 networks gradually without concerns for the user churn.

The past few years have witnessed a surge of interest towards deep reinforcement learning (DRL) in computer networks. With extraordinary ability of feature extraction, DRL has the potential to re-engineer the fundamental resource allocation problems in networking without relying on pre-programmed models or assumptions about dynamic environments. However, such black-box systems suffer from poor robustness, showing high performance variance and poor tail performance. In this work, we propose a unified Teacher-Student learning framework that harnesses rich domain knowledge to improve robustness. The domain-specific algorithms, less performant but more trustable than DRL, play the role of teachers providing advice at critical states; the student neural network is steered to maximize the expected reward as usual and mimic the teacher’s advice meanwhile. The Teacher-Student method comprises of three modules where the confidence check module locates wrong decisions and risky decisions, the reward shaping module designs a new updating function to stimulate the learning of student network, and the prioritized experience replay module to effectively utilize the advised actions. We further implement our Teacher-Student framework in existing video streaming (Pensieve), load balancing (DeepLB), and TCP congestion control (Aurora). Experimental results manifest that the proposed approach reduces the performance standard deviation of DeepLB by 37%; it improves the 90th, 95th, and 99th tail performance of Pensieve by 7.6%, 8.8%, and 10.7% respectively; and it accelerates the growth rate of Aurora by 2x at the initial stage, and achieves a more stable performance in dynamic environments.

In developing countries suffering from a severe shortage of physicians, the follow-up clinical consultation of long-term chronic care becomes a high-speed collaborative work, which has to be conducted in a few minutes. Although much existing work studies how time factors affect physicians' workflows and requirements for information systems, high-speed chronic care is yet to be well investigated. This work bridges the gap by presenting a case of a pediatric hospital in China. We focus on the processes of follow-up consultations, the factors enabling physicians to complete consultation in several minutes, as well as the challenges faced by physicians and patients. Through observations and interviews, we find that physicians conduct multiple tasks (information acquisition, patient-provider communication, and medical data documentation) simultaneously to reduce the consultation duration. Adopting an information summary alternative is the key to fast information acquisition. Templates and references in EMR contribute to rapid documentation and prescription. However, multitasking brings physicians a heavy cognitive load. It also severely compresses the duration of patient-provider communication. As a result, some of the patients' needs, especially emotional ones, are neglected. Based on these findings, we discuss the characteristics and requirements of high-speed chronic care and accordingly propose design suggestions.

Today, Internet service deployment is typically implemented with server replication at multiple locations. Domain name system (DNS), which translates human-readable domain names into network-routable IP addresses, is typically used for distributing users to different server replicas. However, DNS relies on several network-based queries and the queries delay the connection setup process between the client and the server replica. In this article, we propose Artemis, a practical low-latency naming and routing system that supports optimal server (replica) selection based on user-defined policies and provides lower query latencies than DNS. Artemis uses a DNS-like domain name-IP mapping for replica selection and achieves low query latency by combining the name resolution process with the transport layer handshake process. In Artemis, all server replicas at different locations share the same anycast IP address, called Service Address. Clients use the Service Address to establish a transport layer connection with the server. The client's initial handshake packet is routed over an overlay network to reach the optimal server. Then the server migrates the transport layer connection to its original unicast IP address after finishing the handshake process. After that, service discovery is completed, and the client communicates with the server directly via IP addresses. To validate the effectiveness of Artemis, we evaluate its performance via both real trace-driven simulation and real-world deployment. The result shows that Artemis can handle a large number of connections and reduce the connection setup latency compared with state-of-the-art solutions. More specifically, our deployment across 11 Google data centers shows that Artemis reduces the connection setup latency by 39.4% compared with DNS.

The past few years has witnessed a surge of interest towards deep reinforcement learning (Deep RL) in computer networks. With extraordinary ability of feature extraction, Deep RL has the potential to re-engineer the fundamental resource allocation problems in networking without relying on pre-programmed models or assumptions about dynamic environments. However, such black-box systems suffer from poor robustness, showing high performance variance and poor tail performance. In this work, we propose a unified Teacher-Student learning framework that harnesses rich domain knowledge to improve robustness. The domain-specific algorithms, less performant but more trustable than Deep RL, play the role of teachers providing advice at critical states; the student neural network is steered to maximize the expected reward as usual and mimic the teacher's advice meanwhile. The Teacher-Student method comprises of three modules where the confidence check module locates wrong decisions and risky decisions, the reward shaping module designs a new updating function to incentive the learning of student network, and the prioritized experience replay module to effectively utilize the advised actions. We further implement our Teacher-Student framework in existing video streaming (Pensieve), load balancing (DeepLB) and TCP congestion control (Aurora). Experimental results manifest that the proposed approach reduces the performance standard deviation of DeepLB by 37%; it improves the 90th, 95th and 99th tail performance of Pensieve by 7.6%, 8.8%, 10.7% respectively; and it accelerates the rate of growth of Aurora by 2x at the initial stage, and achieves a more stable performance in dynamic environments.

Symbolic execution is an essential approach for automated test case generation. However, the approach is generally not scalable to large programs. One critical reason is that the constraint solving problems in symbolic execution are generally hard. Consequently, the symbolic execution process may get stuck in solving such hard problems. To mitigate this issue, symbolic execution tools generally rely on a timeout threshold to terminate the solving. Such a timeout is generally set to a fixed, predefined value, e.g., five minutes in angr. Nevertheless, how to set a proper timeout is critical to the tool’s efficiency. This paper proposes an approach to tackle the problem by predicting the time required for solving a constraint model so that the symbolic execution engine could base on the information to determine whether to continue the current solving process. Due to the cost of the prediction itself, our approach triggers the predictor only when the solving time has exceeded a relatively small value. We have shown that such a predictor can achieve promising performance with several different machine learning models and datasets. By further employing an adaptive design, the predictor can achieve an F1-score ranging from 0.743 to 0.800 on these datasets. We then apply the predictor to eight programs and conduct simulation experiments. Results show that the efficiency of constraint solving for symbolic execution can be improved by 1.25x to 3x, depending on the distribution of the hardness of their constraint models.

Dating apps have gained tremendous popularity during the past decade. Compared with traditional offline dating means, dating apps ease the process of partner finding significantly. While bringing convenience to hundreds of millions of users, dating apps are vulnerable to become targets of adversaries. In this article, we focus on malicious user detection in dating apps. Existing methods overlooked the signals hidden in the textual information of user interactions, particularly the interplay of temporal-spatial behaviors and textual information, leading to limited detection performance. To tackle this, we propose DatingSec, a novel malicious user detection system for dating apps. Concretely, DatingSec leverages long short-term memory neural networks (LSTM) and an attentive module to capture the interplay of users' temporal-spatial behaviors and user-generated textual content. We evaluate DatingSec on a real-world dataset collected from Momo, a widely used dating app with more than 180 million users. Experimental results show that DatingSec outperforms state-of-the-art methods and achieves an F1-score of 0.857 and AUC of 0.940.

Owing to the ubiquitous penetration of Wi-Fi in our daily lives, Wi-Fi indoor localization has attracted intensive attentions in the last decade or so. Despite some significant progresses, the high accuracy of existing systems is still achieved at the cost of dense access point (AP) deployment. The more practical single AP localization is largely left as an open problem because the hardware-induced time delay “contaminates” the measurement of signal propagation time in the air. In this article, we design and implement M3to tackle this challenge with commodity Wi-Fi cards. M3exploits a multipath-assisted approach that turns the harmful multipath from foe to friend to enable single AP localization: a device can be pinpointed through the combination of azimuths and the relative time of flight (ToF) of Line-of-Sight (LoS) signal and reflection signals, eliminating the need for multiple APs along with their absolute ToF measurements. M3further utilizes frequency hopping to combine multiple channels to form a virtually wider-spectrum channel for higher ToF resolution. As a prominent feature of M3, the channels do not need to be adjacent. Comprehensive experiments demonstrate that M3outperforms the state-of-the-art systems and achieves a median localization accuracy of 71 cm in three environments with a single AP.

In theory, the capacity of a wireless network grows linearly with the number of users and antennas equipped at the communication devices, and hence large-scale MU-MIMO can scale up the network throughput. However, three main challenges are impeding the implementation of this promising technology in the state-of-the-art WLANs. Firstly, the current large-scale MU-MIMO technology demands a large number of high-priced RF chains. Secondly, the wireless access points (APs) are overwhelmed by channel state information (CSI) feedback for nulling multi-user and -antenna interference. Thirdly, the lack of scalable user selection scheme limits the capability of APs to serve a large user population. To address these problems, we design BUSH, a large-scale MU-MIMO prototype that performs scalable beam user selection with hybrid beamforming for phased-array antennas in legacy WLANs. We design a low complexity algorithm that assigns each pair of RF chain and analog beam to the users to effectively reduce channel correlation and cross-talk interference without instantaneous CSI feedbacks. As a prerequisite of user selection, BUSH presents a low-overhead probing scheme in multi-carrier WLANs and designs a highly accurate blind Power Azimuth Spectrum (PAS) estimation algorithm using a single RF chain. For reducing the number of RF-chains used, the phased-array antennas use analog beamforming to steer beams toward each selected downlink user, and multiple RF chains use beamforming to further mitigate the interference among users. We implement BUSH on a software-defined radio platform and evaluate its performance in more than 30 indoor scenarios. The experimental results reveal that for throughput, BUSH outperforms the legacy 802.11ac by 2.08×, and an alternative benchmark system by 1.22× on average.

Online social networks (OSNs) have become a commodity in our daily life. As an important concept in sociology and viral marketing, the study of social influence has received a lot of attentions in academia. Most of the existing proposals work well on dominant OSNs, such as Twitter, since these sites are mature and many users have generated a large amount of data for the calculation of social influence. Unfortunately, cold-start users on emerging OSNs generate much less activity data, which makes it challenging to identify potential influential users among them. In this work, we propose a practical solution to predict whether a cold-start user will become an influential user on an emerging OSN, by opportunistically leveraging the user’s information on dominant OSNs. A supervised machine learning-based approach is adopted, transferring the knowledge of both the descriptive information and dynamic activities on dominant OSNs. Descriptive features are extracted from the public data on a user’s homepage. In particular, to extract useful information from the fine-grained dynamic activities that cannot be represented by the statistical indices, we use deep learning technologies to deal with the sequential activity data. Using the real data of millions of users collected from Twitter (a dominant OSN) and Medium (an emerging OSN), we evaluate the performance of our proposed framework to predict prospective influential users. Our system achieves a high prediction performance based on different social influence definitions.

Accurate traffic speed prediction is an important and challenging topic for transportation planning. Previous studies on traffic speed prediction predominately used spatio-temporal and context features for prediction. However, they have not made good use of the impact of traffic incidents. In this work, we aim to make use of the information of incidents to achieve a better prediction of traffic speed. Our incident-driven prediction framework consists of three processes. First, we propose a critical incident discovery method to discover traffic incidents with high impact on traffic speed. Second, we design a binary classifier, which uses deep learning methods to extract the latent incident impact features. Combining above methods, we propose a Deep Incident-Aware Graph Convolutional Network (DIGC-Net) to effectively incorporate traffic incident, spatio-temporal, periodic and context features for traffic speed prediction. We conduct experiments using two real-world traffic datasets of San Francisco and New York City. The results demonstrate the superior performance of our model compared with the competing benchmarks.

The electronic medical record (EMR) systems face many challenges in supporting chronic disease treatment, especially in medical information summary. Many solutions have recently been proposed for hospitals in developed countries. However, these solutions maybe not suitable for hospitals in developing countries because their workflow and patterns may be quite different due to their resource limitations, especially in shortage of physicians. Investigating the information summary alternatives in treating chronic diseases in such hospitals can shed light on EMR system design, especially on that for developing countries. We study one of the best pediatric hospitals in China. It suffers from a severe shortage of physicians. We introduce its information summary alternative, \fs. In particular, we study how and why pediatricians treat chronic diseases with the sheet. Our work unveils the intense work patterns and their corresponding information requirements of hospitals in China. We also demonstrate the characteristics of the \fs\ and accordingly discuss how EMR systems can be optimized.

Online developer communities like GitHub provide services such as distributed version control and task management, which allow a massive number of developers to collaborate online. However, the openness of the communities makes themselves vulnerable to different types of malicious attacks, since the attackers can easily join and interact with legitimate users. In this work, we formulate the malicious account detection problem in online developer communities, and propose GitSec, a deep learning-based solution to detect malicious accounts. GitSec distinguishes malicious accounts from legitimate ones based on the account profiles as well as dynamic activity characteristics. On one hand, GitSec makes use of users' descriptive features from the profiles. On the other hand, GitSec processes users' dynamic behavioral data by constructing two user activity sequences and applying a parallel neural network design to deal with each of them, respectively. An attention mechanism is used to integrate the information generated by the parallel neural networks. The final judgement is made by a decision maker implemented by a supervised machine learning-based classifier. Based on the real-world data of GitHub users, our extensive evaluations show that GitSec is an accurate detection system, with an F1-score of 0.922 and an AUC value of 0.940.

Today, Internet service deployment is typically implemented with server replication at multiple locations for the purpose of load balancing, failure tolerance, and user experience optimization. Domain name system (DNS) is responsible for translating human-readable domain names into network-routable IP addresses. When multiple replicas exist, upon the arrival of a query, DNS selects one replica and responds with its IP address. Thus, the delay caused by the process of DNS query including the selection of replica is part of the connection setup latency.

Software Defined Networking (SDN) enables flexible flow control by deploying fine-grained rules in OpenFlow switches. Modern commodity switches usually use TCAM to store these rules and perform high-speed parallel lookups. Though efficient, the TCAM capacity is limited because TCAM is expensive in cost and power-hungry. The explosive growth in the number of rules has exacerbated the limitation of TCAM. There have been considerable efforts in implementing hybrid flow tables with both TCAM and RAM, where the high-speed TCAM is regarded as a cache to store the most popular rules and the cheap RAM is used to handle cache miss. The primary challenges for designing hybrid TCAM/RAM flow tables lie in how to improve cache hit rate and how to handle wildcard rule dependency when allocating rules between TCAM and RAM.

Layout bugs commonly exist in mobile apps. Due to the fragmentation issues of smartphones, a layout bug may occur only on particular versions of smartphones. It is quite challenging to detect such bugs for state-of-the-art commercial automated testing platforms, although they can test an app with thousands of different smartphones in parallel. The main reason is that typical layout bugs neither crash an app nor generate any error messages. In this paper, we present our work for detecting text-layout bugs, which account for a large portion of layout bugs. We model text-layout bug detection as a classification problem. This then allows us to address it with sophisticated image processing and machine learning techniques. To this end, we propose an approach which we call Textout. Textout takes screenshots as its input and adopts a specifically-tailored text detection method and a convolutional neural network (CNN) classifier to perform automatic text-layout bug detection. We collect 33,102 text-region images as our training dataset and verify the effectiveness of our tool with 1,481 text-region images collected from real-world apps. Textout achieves an AUC (area under the curve) of 0.956 on the test dataset and shows an acceptable overhead. The dataset is open-source released for follow-up research.

IPFS has surged into popularity in recent years. It organizes user data as multiple objects where users can obtain the objects according to their Content IDentifiers (CIDs). As a storage system, it is of great importance to understand its data I/O performance. But existing work still lacks such a comprehensive study. In this work, we deploy an IPFS storage system with geographically-distributed storage nodes on Amazon EC2. We then conduct extensive experiments to evaluate the performance of data I/O operations from a client's perspective. We find that the access patterns of I/O operations (e.g., request size) severely affect the I/O performance, since IPFS typically uses multiple I/O strategies to perform different I/O requests. Moreover, for the read operations, IPFS requires to resolve remote nodes and downloading objects via the internet. Our experimental study reveals that both resolving and downloading operations can become bottlenecks. Our results can shed light to optimizing IPFS in avoiding high-latency I/O operations.

With an increasing demand for flexible management in software-defined networks (SDNs), it becomes critical to minimize the network policy update time. Although major SDN controllers are now optimized for rapid network update at the control plane, there is still room for data plane optimization in terms of update time, when using TCAM-based physical SDN commodity-off-the-shelf switches. A slow update directly affects network performance and creates bottlenecks. To minimize the flow entry update time, a dependency graph, a kind of directed acyclic graph (DAG), can be used for the access management of flow entries at the switch. Thanks to the DAG, unnecessary entry movements, which are the main factor slowing down flow entry updates, can be avoided. However, existing algorithms show limitations when updates become very frequent. We propose a new flow entry update algorithm, called FastRule, that exploits a greedy strategy with an efficient data structure to accelerate flow entry update with a DAG approach. Moreover, we also adjust our algorithm for other flow table layouts to make it scalable. We elaborate on the correctness of FastRule and test our algorithm using a hardware switch. Compared with existing algorithms, the evaluation shows that our algorithm is about 100x faster than state-of-the-art solutions with a flow table of 1k size.

With an increasing demand for resilience in software-defined networks (SDN), it becomes critical to minimize service recovery delay upon route failures. Fast reroute (FRR) mechanisms are widely used in IP and MPLS networks by computing the recovery path before a failure occurs. The centralized control plane in SDN can potentially enhance path computation, so that FRR path computation can better scale in SDN than in traditional networks. However, the traditional FRR path computation algorithms could lead to a poor performance in large-scale SDN. The problem can become more severe for a highly dynamic network, which often sees dozens of failures or configuration changes in any single day. We propose a new algorithm that exploits pruned searching to quickly compute recovery paths for all-pair switches/hosts upon a link failure. For applications requiring stringent path robustness levels, we also extend this algorithm to quickly find the shortest guaranteed-cost path, which ensures that the recovery path used upon on-path link failures has the minimum cost. Compared with traditional solutions, our evaluations show that our algorithm is about 8 ~ 81 times faster than the practical implementation, 1.93 ~ 3.11 times faster than the state-of-the-art solution. Our results also show that the shortest guaranteed-cost path can reduce the cost of the recovery path significantly. Moreover, we design a prototype to show how to deploy our algorithm in an OpenFlow network.

Summary Most Android applications include third‐party libraries (3PLs) to make revenues, to facilitate their development, and to track user behaviors. 3PLs generally require specific permissions to realize their functionalities. Current Android systems manage permissions in app (process) granularity. As a result, the permission sets of apps with 3PLs (3PL‐apps) may be augmented, introducing overprivilege risks. In this paper, we firstly study how severe the problem is by analyzing the permission sets of 27 718 real‐world Android apps with and without 3PLs downloaded in both 2016 and 2017. We find that the usage of 3PLs and the permissions required by 3PL‐apps have increased over time. As a result, the possibility of overprivilege risks increases. We then propose Perman, a fine‐grained permission management mechanism for Android. Perman isolates the permissions of the host app and those of the 3PLs through dynamic code instrumentation. It allows users to manage permission requests of different modules of 3PL‐apps during app runtime. Unlike existing tools, Perman does not need to redesign Android apps and systems. Therefore, it can be applied to millions of existing apps and various Android devices. We conduct experiments to evaluate the effectiveness and efficiency of Perman. The experimental results verify that Perman is capable of managing permission requests of the host app and those of the 3PLs. We also confirm that the overhead introduced by Perman is comparable to that by existing commercial permission management tools.

Software-defined networking (SDN) provides flexible network control, which has enabled new, more complex network control mechanisms. These approaches impose high demands on flow updates in OpenFlow switches. Existing SDN controllers and firmware have been optimized for high-speed network updates, but specific switch implementations differ in their behavior. Previous optimization schemes for updates ignore the diversity of instruction types and switch behavior. In this paper, we present RuleTailor, an efficient, measurement-based optimization framework for SDN flow updates, to overcome these limitations. In contrast to other measurement-based optimization frameworks, RuleTailor uses new techniques, such as instruction type transformation, pseudo deletion, and match field distance, which can work in both the control plane and the data plane. Our evaluation shows that RuleTailor achieves a performance of fewer than 12 milliseconds per update in a content-addressable flow table with 1k entries, outperforming the state-of-the-art measurement-based framework, Tango, by a factor of 10.

The price-anticipating Kelly mechanism (PAKM) is one of the most extensively used strategies to allocate divisible resources for strategic users in communication networks and computing systems. The users are deemed as selfish and also benign, each of which maximizes his individual utility of the allocated resources minus his payment to the network operator. However, in many applications a user can use his payment to reduce the utilities of his opponents, thus playing a misbehaving role. It remains mysterious to what extent the misbehaving user can damage or influence the performance of benign users and the network operator. In this work, we formulate a non-cooperative game consisting of a finite amount of benign users and one misbehaving user. The maliciousness of this misbehaving user is captured by his willingness to pay to trade for unit degradation in the utilities of benign users. The network operator allocates resources to all the users via the price-anticipating Kelly mechanism. We present six important performance metrics with regard to the total utility and the total net utility of benign users, and the revenue of network operator under three different scenarios: with and without the misbehaving user, and the maximum. We quantify the robustness of PAKM against the misbehaving actions by deriving the upper and lower bounds of these metrics. With new approaches, all the theoretical bounds are applicable to an arbitrary population of benign users. Our study reveals two important insights: 1) the performance bounds are very sensitive to the misbehaving user's willingness to pay at certain ranges and 2) the network operator acquires more revenues in the presence of the misbehaving user which might disincentivize his countermeasures against the misbehaving actions.

Due to the rapid development of mobile devices and location-based services, location-based social networks (LBSNs) have become very popular in our daily-life. Malicious account detection is very helpful for different kinds of practical applications. In this paper, we explore the malicious account detection problem by introducing a deep learning-based framework. By using the long short-term memory (LSTM) neural network, we are able to build a classifier to achieve the binary classification. By using the real data collected from Momo, a widely used LBSN which has more than 180 million users around the world, we evaluate our framework and the result shows great promise for malicious account detection tasks.

Determining the shortest-path distance between vertices in the weighted graph is an important problem for a broad range of fields, such as context-aware search and route selection. While many efficient methods for querying shortest-path distance have been proposed, they are poorly suited for parallel architectures, such as multi-core CPUs or computer clusters, due to the strong task dependencies. In this paper, we propose ParaPLL, a new parallelism-friendly framework for fast shortest-path distance query on large-scale weighted graphs. ParaPLL exploits intra-node and inter-node parallelism by using shared memory and message passing paradigms respectively. We also design task assignment and synchronization policies, which allow ParaPLL to reach remarkable speedups compared to state-of-the-art solutions. Moreover, we also prove the correctness of ParaPLL. To the best of our knowledge, ParaPLL is the first parallel framework that utilizing pruned landmark labeling to accelerate shortest-path distance queries on large-scale weighted graphs. Our evaluation results show that ParaPLL is 9.46 times faster than the corresponding serial version on a weighted 0.3M-vertex graph using a 12-core computer. ParaPLL on a 6-node computer cluster can also achieve a speedup of up to 5.6 over the single-node implementation.

Cloud-of-clouds storage is a viable means to ensure security and reliability of distributed data storage, where data are encrypted, encoded, and stored in multiple clouds. However, it is a great challenge to adopt such a paradigm in mobile devices (e.g., smartphone). Mobile devices are generally incapable to perform the heavy-weight operations (i.e., data encryption, encoding, and transmission) required in such a paradigm, given the limited resources in such devices. This paper focuses on addressing this challenge, i.e., improving data storage performance in mobile cloud-of-clouds storage systems. The key of our proposal is to allow the low-capability mobile devices to offload the computational and transmission overhead to the clouds. In other words, we propose a Network Coding based Cloud-of-clouds Storage (NCCS) scheme, where the clouds can encode and exchange data collaboratively. We consider two state-of-the-art cloud-of-clouds storage approaches, i.e., AONT-RS and CAONT-RS, as example cases to deploy our scheme. Accordingly, we propose their network coding-based enhancements, namely NAONT-RS and NCAONT-RS. We implement a prototype cloud-of-clouds system to verify the efficiency of our proposal. We deploy the prototype on Microsoft Azure and conduct extensive experiments with real-world traces. The experimental results show that NAONT-RS and NCAONT-RS can reduce the time of data storage process by up to 50% and improve the throughput by up to 110% compared with their original versions, i.e., AONT-RS and CAONT-RS.

This paper considers the problem of how to reduce the I/O overhead of data update operations in erasure coding based storage systems. To this end, we first analyze the I/O overhead of update operations with current update approaches. We find the key to reduce such I/O overhead is designing a scheduling algorithm to construct the sequence of update operations. Such an algorithm needs to execute with a time limit, since update requests work under a stringent latency constraint. To quickly schedule the order of update operations, we propose an efficient algorithm, namely UCODR. Our theoretical analysis verifies that UCODR can effectively reduce the I/O overhead of update operations when multiple blocks are updated. To further confirm its effectiveness, we implement a prototype storage system to deploy UCODR with different erasure codes. Extensive experiments are conducted on the prototype storage system with real-world traces. The experimental results show that UCODR can reduce the time of update operations by up to 35 percent and improve the throughput of the storage system by up to 67 percent, compared with the state-of-the-art update approaches.

The multiple-unicast conjecture in network coding states that for multiple unicast sessions in an undirected network, network coding has no advantage over routing in improving the throughput or saving bandwidth. In this paper, we propose a reduction method to study the multiple-unicast conjecture, and prove the conjecture for a new class of networks that are characterized by relations between cut-sets and source-receiver paths. This class subsumes all the known types of networks with non-zero max-flow min-cut gaps but zero coding advantage. Combining this result with a computer-aided search, we derive as a corollary that network coding is unnecessary in networks with up to six coding nodes. We also prove the multiple-unicast conjecture for almost all unit-link-length networks with up to three sessions and seven nodes.

As a result of the blooming of online social networks (OSNs), a user often holds accounts on multiple sites. In this article, we study the emerging “cross-site linking” function available on mainstream OSN services including Foursquare, Quora, and Pinterest. We first conduct a data-driven analysis on crawled profiles and social connections of all 61.39 million Foursquare users to obtain a thorough understanding of this function. Our analysis has shown that the cross-site linking function is adopted by 57.10% of all Foursquare users, and the users who have enabled this function are more active than others. We also find that the enablement of cross-site linking might lead to privacy risks. Based on cross-site links between Foursquare and external OSN sites, we formulate cross-site information aggregation as a problem that uses cross-site links to stitch together site-local information fields for OSN users. Using large datasets collected from Foursquare, Facebook, and Twitter, we demonstrate the usefulness and the challenges of cross-site information aggregation. In addition to the measurements, we carry out a survey collecting detailed user feedback on cross-site linking. This survey studies why people choose to or not to enable cross-site linking, as well as the motivation and concerns of enabling this function.

Owing to great potential in smart home and human-computer interactive applications, WiFi indoor localization has attracted extensive attentions in the past several years. The state-of-the-art systems have successfully achieved decimeter-level accuracies. However, the high accuracy is acquired at the cost of dense access point (AP) deployment, employing large size of frequency bandwidths or special-purpose radar signals which are not compatible with existing WiFi protocol, limiting their practical deployments. This paper presents the design and implementation of AWL, an accurate indoor localization system that enables a single WiFi AP to achieve decimeter-level accuracy with only one channel hopping. The key enabler of the system is we novelly employ channel hopping to create virtual antennas, without the need of adding more antennas or physically move the antennas' positions for a larger antenna array. We successfully utilize the widely known "bad" spatial aliasing to improve the AoA estimation accuracy. A novel multipath suppression scheme is also proposed to combat the severe multipath issue indoors. We build a prototype of AWL on WARP software-defined radio platform. Comprehensive experiments manifest that AWL achieves a median localization accuracy of 38 cm in a rich multipath indoor environment with only a single AP equipped with 6 antennas. In a small scale area, AWL is able to accurately track a moving device's trajectory, enabling applications such as writing/drawing in the air.

Third-party libraries (3PLs) are widely introduced into Android apps and they typically request permissions for their own functionalities. Current Android systems manage permissions in process (app) granularity. Hence, the host app and the 3PLs share the same permission set. 3PL-apps may therefore introduce security risks. Separating the permission sets of the 3PLs and those of the host app are critical to alleviate such security risks. In this paper, we provide Perman, a tool that allows users to manage permissions of different modules (i.e., a 3PL or the host app) of an app at runtime. Perman relies on dynamic code instrumentation to intercept permission requests, and accordingly provide a policy-based permission control. Unlike existing tools that generally require to redesign 3PL-apps, it can thus be applied to the existing apps in market. We evaluate Perman on real-world apps. The experiment results verify its effectiveness in fine-grained permission management.

Multiview video is essential for various mobile three-dimensional (3D) and immersive applications that can capture scenes from multiple angles for better user experience. However, robust transmission of multiview video is very challenging in wireless networks due to high bandwidth requirement and time-varying channel quality. Though the up-to-date 802.11 system enables spatial multiplexing MIMO to enhance transmission capacity, it is still agnostic to 3D source coding structure in the transmission. In this paper, we study the optimal resource allocation problem in MIMO systems that deliver 3D content with multiview video coding. The basic idea is to exploit the channel diversity of multiple antennas and the source coding characteristics so as to achieve unequal error protection against channel errors. To achieve this goal, we develop a nonlinear mixed integer programming framework to perform antenna selection and power allocation, and propose low-complexity algorithms to assign these resources. We implement a proof-of-concept system, namely MuVi, on the software-defined-radio platform, WARP, to evaluate the proposed algorithms. MuVi is the practical system to tackle 3D multiview streaming in the latest Wi-Fi networks such as IEEE 802.11ac under realistic channel conditions. Extensive experimental results demonstrate that the peak signal-to-noise-ratio of MuVi significantly outperforms that of the conventional power allocation scheme in a variety of indoor environments.

The fundamental tasks of the control plane in software defined networking (SDN) are to customize forwarding policies for the data plane and to provide global network view for applications. The logically centralized control plane design brings benefits in terms of network programmability and can largely ease network management. However, it also increases efficiency concerns. One practical control plane challenge is path computation, because it can require a significant amount of computation load if the network scale is large and the path requests from applications are frequent. In this paper, our goal is to build a high-performance control plane for path computation using multiple controllers. Previous works attempt to improve control plane efficiency by balancing only the load for data plane behavior between multiple controllers. Going beyond conventional wisdom, we designed ParaCon, a solution we propose to speed up the control plane by distributing the load of path computation. We also address the consistency and synchronization overhead challenges related to ParaCon design. To the best of our knowledge, ParaCon is the first attempt that utilizes node parallelism in SDN path computation. We evaluated ParaCon using both Mininet and real-world clusters. Our results show that the path computing time of ParaCon can achieve a speedup of 10× over Floyd (used in POX) and Dijkstra (used in ONOS) baseline implementations for networks with hundreds of nodes.

Oil and gas well production prediction takes place in early stages of production to estimate future recovery. A data driven workflow is proposed in this paper to construct a symbolic tree model to predict new well production using historic time-series production data of analogous wells. Production data are firstly aggregated and symbolized for dimensionality reduction and data discretization of time-series data. A symbolic tree is constructed on time-series symbol sequences, and pre-pruning mechanisms - minimum node size and spatial information gain - are integrated to achieve a compact and informative tree. A coverage index is used to assess the tree size. A case study was conducted applying the proposed workflow to shale gas wells in Montney-A pool in Canada. It has proved the feasibility and accuracy of the proposed method.

In the past decade, Location-Based Social Networks (LBSNs) have attracted attentions from both the academia and industry. All LBSN sites are venue-centric, i.e., each review or check-in must be associated with a venue. Despite the importance of the venues, there still lacks a systematical study on LBSN sites from the venues' perspective. To fill this gap, we conduct a comprehensive study on Dianping, the largest online review site in China, with data of more than 506K venues. We first characterize the demographics of each venue. We then measure the venue popularity through real-time reviews. Finally, we propose the concept of “venue network” to study the linkage among the venues. Our paper not only provides a clear picture of venues on Dianping, but also sheds light on potential applications in LBSNs.

Distributed storage systems introduce redundancy to protect data from node failures. After a storage node fails, the lost data should be regenerated at a replacement storage node as soon as possible to maintain the same level of redundancy. Minimizing such a regeneration time is critical to the reliability of distributed storage systems. Existing work commits to reduce the regeneration time by either minimizing the regenerating traffic, or adjusting the regenerating traffic patterns, whereas nodes participating the regeneration are generally assumed to be given beforehand. However, real-world distributed storage systems usually exhibit heterogeneous link capacities, and the regeneration time is highly related to the selection of the participating nodes. In this paper, we consider the minimization of the regeneration time by selecting the participating nodes in heterogeneous networks. We propose optimal node selection algorithms respectively for two cases: 1) the newcomer is not given, 2) both the newcomer and the providers are not given. Analysis shows that the optimal regeneration time can be achieved in each case. We then consider the effect of flexible amount of data blocks from each provider on the regeneration time, and apply this observation to enhance our schemes. Experiment results show that our node selection schemes can significantly reduce the regeneration time, especially in practical networks with heterogeneous link capacities, compared with the scheme based on random node selection.

Data access is an important and challenging issue in Mobile Social Networks (MSNs), and cooperative caching is an effective technique to improve the access performance. Most of current research efforts in data access of MSNs focus on improving the access performance while neglecting the fair treatment of users. Because fairness is considered as a major incentive for peer-to-peer service especially in infrastructure-less wireless networks, in this paper we propose a novel approach to support fairness aware cooperative caching scheme in MSNs. Through capturing close friend set of each node, we cache data prior at nodes which are overlapped by most nodes' close friend sets. Then we derive the optimal cooperative scheme by using the minimum dominating set, which is an NP-Complete problem, and we design a heuristic algorithm to handle it. Experimental results show that our scheme can effectively improve data access fairness as well as maintain nearly the same access performance compared to existing cooperative caching schemes.

Distributed storage systems provide large-scale reliable data storage services by spreading redundancy across a large group of storage nodes. In such big systems, node failures take place on a regular basis. When a node fails or leaves the system, to maintain the same level of redundancy, it is expected to regenerate the redundant data at a replacement node as soon as possible. Previous studies aim to minimize the network traffic in the regeneration process, but in practical networks, where link capacities vary in a wide range, minimizing network traffic does not always mean minimizing regeneration time. Considering the heterogeneous link capacities, Li et al. proposed a tree-structured regeneration scheme, called RCTREE, to bypass the low-capacitated link encountered in direct transmissions. However, we find that RCTREE may rapidly lose data integrity after several regenerations. In this paper, we reconsider the problem of minimizing regeneration time in networks with heterogeneous link capacities. We derive the minimum amount of data to be transmitted through each link to preserve data integrity. We prove that building an optimal regeneration tree is NP-complete and propose a heuristic algorithm for a near-optimal solution. We further introduce a flexible regeneration scheme, which allows providers to generate different amount of coded data. Simulation results show that the flexible tree-structured regeneration scheme can reduce the regeneration time significantly.

Network coding encourages the mixing of information flows at intermediate nodes of a network for enhanced network capacity, especially for one-to-many multicast applications. A fundamental problem in multicast network coding is to construct a feasible solution such that encoding and decoding are performed over a finite field of size as small as possible. Coding operations over very small finite fields (e.g., F2) enable low computational complexity in theory and ease of implementation in practice. In this work, we propose a new approach based on matroid theory to study multicast network coding and its minimum field size requirements. Applying this new approach that translates multicast networks into matroids, we derive the first upper-bounds on the field size requirement based on the number of relay nodes in the network, and make new progresses along the direction of proving that coding over very small fields (F2and F3) suffices for multicast network coding in planar networks.

As storage systems grow in size, device failures happen more frequently than ever before. Given the commodity nature of hard drives employed, a storage system needs to tolerate a certain number of disk failures while maintaining data integrity, and to recover lost data with minimal interference to normal disk I/O operations. RAID-6, which can tolerate up to two disk failures with the minimum redundancy, is becoming widespread. However, traditional RAID-6 codes suffer from high disk I/O overhead during recovery. In this paper, we propose a new family of RAID-6 codes, the Minimum Disk I/O Repairable (MDR) codes, which achieve the optimal disk I/O overhead for single failure recoveries. Moreover, we show that MDR codes can be encoded with the minimum number of bit-wise XOR operations. Simulation results show that MDR codes help to save about half of disk read operations than traditional RAID-6 codes, and thus can reduce the recovery time by up to 40%.

Network coding encourages information flow mixing in a network. It helps increase the throughput and reduce the cost of data transmission, especially for one-to-many multicast applications. An interesting problem is to understand and quantify the coding advantage and cost advantage, i.e., the potential benefits of network coding, as compared to routing, in terms of increasing throughput and reducing transmission cost, respectively. Two classic network models were considered in previous studies: directed networks and undirected networks. This work further focuses on two types of parameterized networks, including bidirected networks and hyper-networks, generalizing the directed and the undirected network models, respectively. We prove upper- and lower-bounds on multicast coding advantage and cost advantage in these models.

Network coding encourages information coding across a communication network. While the necessity, benefit and complexity of network coding are sensitive to the underlying graph structure of a network, existing theory on network coding often treats the network topology as a black box, focusing on algebraic or information theoretic aspects of the problem. This paper aims at an in-depth examination of the relation between algebraic coding and network topologies. We mathematically establish a series of results along the direction of: if network coding is necessary/beneficial, or if a particular finite field is required for coding, then the network must have a corresponding hidden structure embedded in its underlying topology, and such embedding is computationally efficient to verify. Specifically, we first formulate a meta-conjecture, the NC-minor conjecture, that articulates such a connection between graph theory and network coding, in the language of graph minors. We next prove that the NC-minor conjecture for multicasting two information flows is almost equivalent to the Hadwiger conjecture, which connects graph minors with graph coloring. Such equivalence implies the existence of K4, K5, K6, and KO(q/log q) minors, for networks that require F3, F4, F5, and Fqto multicast two flows, respectively. We finally prove that, for the general case of multicasting arbitrary number of flows, network coding can make a difference from routing only if the network contains a K4minor, and this minor containment result is tight. Practical implications of the above results are discussed.

In distributed storage systems, a substantial volume of data are stored in a distributed fashion, across a large number of storage nodes. To maintain data integrity, when existing storage nodes fail, lost data are regenerated at replacement nodes. Regenerating multiple data losses in batches can reduce the consumption of bandwidth. However, existing schemes are only able to achieve lower bandwidth consumption by utilizing a large number of participating nodes. In this paper, we propose a cooperative pipelined regeneration process that regenerates multiple data losses cooperatively with much fewer participating nodes. We show that cooperative pipelined regeneration is not only able to maintain optimal data integrity, but also able to further reduce the consumption of bandwidth as well.

Network Coding encourages information coding across a communication network. While the necessity, benefit and complexity of network coding are sensitive to the underlying graph structure of a network, existing theory on network coding often treats the network topology as a black box, focusing on algebraic or information theoretic aspects of the problem. This work aims at an in-depth examination of the relation between algebraic coding and network topologies. We mathematically establish a series of results along the direction of: if network coding is necessary/beneficial, or if a particular finite field is required for coding, then the network must have a corresponding hidden structure embedded in its underlying topology, and such embedding is computationally efficient to verify. Specifically, we first formulate a meta-conjecture, the NC-Minor Conjecture, that articulates such a connection between graph theory and network coding, in the language of graph minors. We next prove that the NC-Minor Conjecture is almost equivalent to the Hadwiger Conjecture, which connects graph minors with graph coloring. Such equivalence implies the existence of K4, K5, K6, and KO(q/ log q)minors, for networks requiring F3, F4, F5and Fq, respectively. We finally prove that network coding can make a difference from routing only if the network contains a K4minor, and this minor containment result is tight. Practical implications of the above results are discussed.

IEEE 802.16j has introduced the concept of WiMAX mesh network model, which adds a special type of nodes called Relay Stations (RSs) for Subscriber Stations (SSs). With the help of RSs, a WiMAX mesh network is able to provide larger wireless coverage, higher network capacity and Non-Line-OfSight (NLOS) communications. In this paper, we propose a novel strategy which named Dual-Path Relay Placement (DPRP) to improve the efficiency of relay station placement in WiMAX mesh networks. DPRP is a more reliable model that takes users Spectrum Efficiency into account and thus can achieve efficient spatial reuse by two independent parallel transmission paths.If the distance between Subscriber Station and Base Station is constant, the SS's Spectrum Efficiency has a big influence on the number of relays required by the SS. For a given attenuation coefficient, DPRP needs less RSs than Single-Path Relay Placement (SPRP) when the SS's Spectrum Efficiency exceeds a certain threshold. The analysis proves that the higher the Spectrum Efficiency is, the greater the difference is between the relay's number of two models. If the new model is integrated with the SPRP model, the total number of RSs required per cell will decrease a lot.

Network coding is a technique that allows information flows to be encoded while routed across a data network. It was shown that network coding helps increase the throughput and reduce the cost of data transmission, especially for one-to-many multicast applications. An important direction in network coding research is to understand and quantify the coding advantage and cost advantage, i.e., the potential benefits of network coding, as compared to routing, in terms of increasing throughput and reducing transmission cost, respectively. Two classic network models were considered in previous studies of coding advantage: directed networks and undirected networks. The study of coding advantage in this work further focuses on two types of parameterized networks, including bidirected networks and hyper-networks, which generalizes the directed and the undirected network models, respectively. With proper parameter setting, more realistic modeling of networks in practice can be achieved. We prove upper-bounds and lower-bounds on the coding advantage for multicast in these models. Some of our bounds are new and unknown before, some improve upon previously proven bounds, and some answer open questions in the literature.

Space information flow is a new field of research recently proposed by Li and Wu [1], [2]. It studies the transmission of information in a geometric space, where information flows can be routed along any trajectories, and can be encoded wherever they meet. The goal is to satisfy given end-to-end unicast/multicast throughput demands, while minimizing a natural bandwidth-distance sum-product (network volume). Space information flow models the design of a blueprint for a minimum-cost network. We study the multicast version of the space information flow problem, in Euclidean spaces. We present a simple example that demonstrates the design of an information network is indeed different from that of a transportation network. We discuss properties of optimal multicast network embedding, prove that network coding does not make a difference in the basic case of 1-to-2 multicast, and prove upper-bounds on the number of relay nodes required in an optimal acyclic multicast network.

Small playback lag in live streaming is important for time-critical and interactive applications such as live stock, market updates, sports and remote education. In this paper, we present Elite addresses the playback lag problem in peer-assisted live streaming systems. Instead of deploying a large initial offset to all the users, Elite seeks the possibility of initializing users with layered proportional initial scheduling point, thus achieving differentiated playback lag service for the system. For saving server bandwidth and reducing lag time, Elite employs a novel strategy which arranges peers into a virtual tree structure and quantifies playback lag of each layer that finally converges to a constant value. This way, Elite can help users to achieve much shorter average playback lag and prioritized service within the same channel. As illustrated in our design, analysis, and simulation studies, Elite is able to fully exploit limited pool of server bandwidth to support peer-assisted live streaming with prioritized playback lag, and achieves shorter average playback lag compared with synchronized strategy, such as R2.

To alleviate the growing concern of energy waste in networked devices, we present PCube, a server-centric data center structure that conserves energy by varying bandwidth availability based on traffic demand. PCube not only supports a low-power mode for existing data centers offering full bisection bandwidth without hardware modification or re-wiring, but also provides an alternative for new data centers at a lower construction cost. The bandwidth demand could be dynamic and scalable in the optimal way when considering conserving the energy waste. To further reduce the cost, we take advantage of traffic locality, and propose Hybrid PCube, in order to offer different bandwidth to different servers. We use samples of traffic from a real-world production data center with full bisection bandwidth across thousands of servers, and evaluate our proposed algorithm using these collected samples. Our experimental results have shown convincing evidence that the proposed algorithm is able to substantially reduce energy costs incurred by networked devices in data centers.

As the physical link speeds grow and the size of routing table continues to increase, IP address lookup has been a challenging problem at routers. There have been growing demands in achieving high-performance IP lookup cost-effectively. Existing approaches typically resort to specialized hardwares, such as TCAM. While these approaches can take advantage of hardware parallelism to achieve high-performance IP lookup, they also have the disadvantage of high cost. This paper investigates a new way to build a cost-effective IP lookup scheme using graphics processor units (GPU). Our contribution here is to design a practical architecture for high-performance IP lookup engine with GPU, and to develop efficient algorithms for routing prefix update operations such as deletion, insertion, and modification. Leveraging GPU's many-core parallelism, the proposed schemes addressed the challenges in designing IP lookup at GPU-based software routers. Our experimental results on real-world route traces show promising gains in IP lookup and update operations.

Security plays an important role in the system design with the development of VANETs. Because of the unreliable communications in VANETs, security protocols need more considerations, such as privacy, authentication, and consistency of messages. However, efficiency has been overlooked before, because previous methods incur significant communication overhead. In this paper, we propose a novel security scheme, named Symmetric-Masquerade Security Scheme (SMSS), which can achieve the security requirements while keeping a low system overhead. In SMSS, we focus on the vehicle-to-vehicle communications, which is among the typical scenarios in VANETs. The symmetric encryption is employed in our scheme to ensure the consistency of the messages, the local pseudonyms are equipped to protect the privacy, and the pre-shared keys are introduced to implement the authentication. This paper makes a detailed description of the novel scheme, then analyzes its performance in comparison with that of the PKI security scheme.

Short playback lag is preferred in many urgent and interactive scenarios such as live sports and distance education. However, measurement studies have shown that many popular P2P live streaming systems still suffer from long playback lag, say, more than 100 seconds, which makes the live streaming le

Distributed storage systems provide reliable storage service by storing data, with a certain amount of redundancy, into a substantial number of storage nodes. In order to compensate the data loss incurred by node failures, the lost data should be regenerated. Tree-structured regeneration, during whi

The objective of many applications with the surveillance missions in wireless sensor networks is to provide long-term monitoring of the specific environments, such as disaster-hit regions. These applications usually perform continuous monitoring without any maintenance, even if some sensor nodes fai

Graphics processing unit (GPU) has evolved into a general-purpose computing platform. Inspired by the GPU technology advantage, this paper concerns the design and performance evaluation of practical GPU-accelerated server architecture for Video-on-Demand (VoD) services with network coding. Following

It has been well known that network coding can achieve better network throughput in certain topologies by allowing coding at intermediate nodes. However, the benefit of network coding in P2P content distribution is controversial in recent literatures. In this paper, we seek to investigate how network coding performs in a BitTorrent-like protocol from a practical perspective. To validate the benefit of network coding in BitTorrent, we first implement NCTorrent, a network coding enabled BitTorrent client. We then track about 10,000 real-world BitTorrent session's user behaviors and use the user behaviors to drive the experiments. Based on the results we establish, our conclusion with respective to the benefits of network coding is pessimistic. Network coding does not help that much in real world scenarios as claimed in previous work. In particular, the logged data also reveal that the last block problem, where network coding is believed to be helpful, only appears with very low probabilities.

Distributed storage systems provide large-scale reliable data storage by storing a certain degree of redundancy in a decentralized fashion on a group of storage nodes. To recover from data losses due to the instability of these nodes, whenever a node leaves the system, additional redundancy should be regenerated to compensate such losses. In this context, the general objective is to minimize the volume of actual network traffic caused by such regenerations. A class of codes, called regenerating codes, has been proposed to achieve an optimal trade-off curve between the amount of storage space required for storing redundancy and the network traffic during the regeneration. In this paper, we jointly consider the choices of regenerating codes and network topologies. We propose a new design, referred to as RCTREE, that combines the advantage of regenerating codes with a tree-structured regeneration topology. Our focus is the efficient utilization of network links, in addition to the reduction of the regeneration traffic. With the extensive analysis and quantitative evaluations, we show that RCTREE is able to achieve a both fast and stable regeneration, even with departures of storage nodes during the regeneration.

The use of network coding has been shown to improve throughput in input-queued multicast switches, but not without costs of computational complexity and delays. In this paper, we investigate the design of efficient online network coding algorithms in a switch with multicast traffic. We present Haste, an online opportunistic coding algorithm designed to streamline the computation when network coding is involved in a network switch with multicast traffic. Haste enjoys the advantage of incurring no decoding delays, which reduces packet delays compared with existing network coding algorithms on switches. We have conducted extensive simulations to show the efficiency of Haste, and implemented an emulation framework to emulate input-queued switches using asynchronous network sockets. Our emulation framework is able to process actual UDP traffic using Haste with online network coding, and to show convincing evidence that Haste is suitable for practical use, and is beneficial in multicast switches.

P2P live streaming systems suffer a lot from long playback lag in lag-sensitive scenarios. In this paper, we propose a new approach to reducing the playback lag in P2P live streaming systems. According to measurement studies, there exist a certain amount of active peers, who stay longer and contribute more bandwidth than other peers. Inspired by this, we propose a tiered overlay design, in which peers are organized into three tiers based on their degrees of activity. We develop a set of algorithms to evaluate the peers' degrees of activity. Specifically, the backbone of the overlay consists of the peers with high activity in tier-1. These active peers are responsible for diffusing the newly generated fresh chunks to peers located in all the involved Autonomous Systems (ASes). They contribute more bandwidth and thus enjoy shorter playback lag. Further more, adaptive biased neighbor selection algorithm is employed among non-backbone peers to keep traffic locality. Evaluated by extensive simulations, the proposed algorithms can reduce the average playback lag and cross-ISP traffic greatly.

IP address lookup is a challenging problem due to the increasing routing table size, and higher line rate. This paper investigates a new way to build an efficient IP lookup scheme using graphics processor units(GPU). Our contribution here is to design a basic architecture for high-performance IP lookup engine with GPU, and to develop efficient algorithms for routing prefix operations such as lookup, deletion, insertion, and modification. In particular, the IP lookup scheme can achieve O(1) time complexity. Our experimental results on real-world route traces show promising 6x gains in IP lookup throughput.

While it is a well known result that network coding achieves optimal flow rates in multicast sessions, its potential for practical use has remained to be a question, due to its high computational complexity. Our previous work has attempted to design a hardware-accelerated and multi-threaded implementation of network coding to fully utilize multi-core CPUs, as well as SSE2 and AltiVec SIMD vector instructions on X86 and PowerPC processors. This paper represents another step forward, and presents the first attempt in the literature to maximize the performance of network coding by taking advantage of not only multi-core CPUs, but also potentially hundreds of computing cores in commodity off-the-shelf graphics processing units (GPU). With GPU computing gaining momentum as a result of increased hardware capabilities and improved programmability, our work shows how the GPU, with a design involving thousands of lightweight threads, can boost network coding performance significantly. Many-core GPUs can be deployed as an attractive alternative and complementary solution to multi-core servers, by offering a better price/performance advantage. In fact, multi-core CPUs and many-core GPUs can be deployed and used to perform network coding simultaneously, potentially useful in media streaming servers where hundreds of peers are served concurrently by these dedicated servers. In this paper, we present Nuclei, the design and implementation of GPU-based network coding. With Nuclei, only one mainstream NVidia 8800 GT GPU outperforms an 8-core Intel Xeon server in most test cases. A combined CPU-GPU encoding scenario achieves coding rates of up to 116 MB/second for a variety of coding settings, which is sufficient to saturate a Gigabit Ethernet interface.

Distributed storage systems, built on peer-to-peer networks, can provide large-scale data storage and high data reliability by redundant schemes, such as replica, erasure codes and linear network coding. Redundant data may get lost due to the instability of distributed systems, such as permanent node departures, hardware failures, and accidental deletions. In order to maintain data availability, it is necessary to regenerate new redundant data in another node, referred to as a newcomer. Regeneration is expected to be finished as soon as possible, because the regeneration time can influence the data reliability and availability of distributed storage systems. It has been acknowledged that linear network coding can regenerate redundant data with less network traffic than replica and erasure codes. However, previous regeneration schemes are all star-structured regeneration schemes, in which data are transferred directly from existing storage nodes, referred to as providers, to the newcomer, so the regeneration time is always limited by the path with the narrowest bandwidth between newcomer and provider, due to bandwidth heterogeneity. In this paper, we exploit the bandwidth between providers and propose a tree-structured regeneration scheme using linear network coding. In our scheme, data can be transferred from providers to the newcomer through a regeneration tree, defined as a spanning tree covering the newcomer and all the providers. In a regeneration tree, a provider can receive data from other providers, then encode the received data with the data this provider stores, and finally send the encoded data to another provider or to the newcomer. We prove that a maximum spanning tree is an optimal regeneration tree and analyze its performance. In a trace-based simulation, the results show the tree-structured scheme can reduce the regeneration time by 75%-82% and improve data availability by 73%-124%.

In this paper we investigate the problems of searching for the capacity factors and determining the capacity ranks of edges in a network coding-based network, which were first proposed in (K. Cai and P.Y. Fan, 2007). For the former problem, we prove that it is computationally hard by reducing the well known NP-complete SUB-SUM problem to the current problem. For the latter problem, we devise efficient algorithms in a special case of networks and conjecture that in general case the problem is also hard.

The benefit of network coding with respect to simplifying scheduling overhead for content distribution has been extensively studied in previous literature. However, the complexity of network coding increases as the content size scales up. In this paper, we study the tradeoff between scheduling overhead and coding overhead. To this end, we propose Swifter, a P2P content distribution scheme, which employs local-rarest-first segment scheduling and chunked network coding algorithms. In Swifter, content is divided into segments, which are further divided into blocks. Each peer schedules a local-rarest segment request from its neighbors. Network coding is then used for generating a reply block within the requested segment. Leveraging our real-world implementation and experiments, we find that Swifter has low coding overhead and can reduce average download time by up to 40% compared to existing work.

Nowadays, the real practice of network coding in wireline networks is focused on the P2P overlay networks. Although it can help to utilize network resources more efficiently, P2P network coding does not exhibit benefits in terms of the maximum throughput. Our work aims to implement network coding at the IP layer, which is an idea not fundamentally new, but with little real practice because of the enormous difficulties involved. In this paper we propose CODED IP, a protocol framework that plugs network coding into the current IP stack. Experiments on a 22-node testbed show that CODED IP provides multicast traffic with not only a significantly higher throughput than overlay network coding and naive IP multicast, but also a more balanced load distribution as compared with overlay network coding.

Clustering is an effective topology maintenance technology to provide better scalability for WSN. When loop structures are deployed to clustering, it can provide better robustness and convenient routing, because the topology information can be disseminated within a loop instead of a single clusterhead and the nature of the loop that there are two paths between each pair of nodes within a loop provides backup routes. In this paper, we propose a loop-based approach to combine clustering and routing for cost saving, which consists of setup procedure, regular procedure and recovery procedure. For the setup procedure, we propose a loop discovery algorithm, which is a fully distributed, on-demand and configurable algorithm achieving route discovery and clustering at the same time. The algorithm prefers a low-mobility network which is typically suitable for wireless sensor networks.

Providing large-scale true video-on-demand (VoD) service is a challenging problem over the Internet. Some characteristics of the Internet, such as packet switching, best effort mechanism, make it difficult to provide VoD services with enough Quality of Service (QoS) assurance. In this paper, we propose a VoD service model that uses a content delivery network approach on backbone and a peer-to-peer (p2p) approach on access over the China’s high performance broadband information network (named 3Tnet). We discuss the feasibility of our approach and evaluate the performance by using simulation. The results show that our proposed service model can support true VoD service on the 3Tnet in a scalable way.

Automatic Switched Optical Network (ASON), standardized by ITU-T, is emerging as a major technology choice for inter-operational and intervendor optical backbone transportation. With the ability to flexibly and automatically establish and maintain connection, ASON promises an IP-traffic tolerant, end-to-end QoS guaranteed transportation mechanism within which both bandwidth-consuming stream media traffic and traditional web traffic can be relayed smoothly. In this paper, we look into the traffic aggregation effect of TCP traffic in an optical switched network environment modeled under the 3TNet network architecture. We provide a simplified simulation model, which shows how large-scale TCP traffic aggregation can be leveraged by ASON switching ability and meanwhile proves the superiority of this architecture.

MANETs are composed of mobile nodes without any infrastructure and nodes cooperate to set up routes for network communications. Because of these characters, MANETs operate in open medium, so they are particularly vulnerable to intrusions. In this paper, we present an efficient intrusion detection system called MAPIDS (Mobile Agent-based Peer-to-peer Intrusion Detection System). On detecting a suspicious activity, MAPIDS initiates a voting approach to make a collective decision and take further action. In contrast to other intrusion detection system based on collective decision, MAPIDS saves more bandwidth and energy and it is immune to sparse nodes problem.

Mobile Ad Hoc Networks (MANET) consists of mobile nodes interconnected by multihop communication paths can experience higher packets loss rate and lower bandwidth compare with wired network because of node mobility and characteristic of wireless channel. Multicast could prove to be an efficient way of providing necessary services for many kinds of applications. In this paper, we consider the performances of different scheduling algorithms in MANET which have both multicast and unicast applications. Here we discuss four scheduling models; that are one-queue model, priority-queue model, dynamic priority-queue model and adaptive queue-model. In these algorithms, we compare the difference between multicast and unicast application performance. In addition we analyze the impact of quality of service (QoS) parameters on hop-by-hop and end-to-end performance. These show that the scheduling algorithms, which we analyzed, have much better hop-by-hop and end-to-end traffic performance than simple scheduling.

We describe an improved agent-based multicast algorithm for mobile ad-hoc users. Mobile multicast agents (MMAs) form a virtual backbone of an ad-hoc network and they provide multicast tree discovery and multicast tree maintenance. However, they do not complete datagram delivery. In order to ensure reliable multicast, MMAs are also in charge of datagram retransmission. The improved mobile multicast algorithm can simplify the multicast tree discovery, reduce control overhead of the network, and increase the total network throughput, in comparison with general AODV multicast operation. At the same time, it can also save battery power and transmission capabilities of those MMAs since they are only used to transfer control information and retransmitted packets. Hence we avoid draining the resources of some mobile nodes (MMAs), which is unfair.

We propose an optimizing queue scheme for multimedia network communications with dynamic priority. Arriving users are classified into two types based on their transmission contents, and they are served by different priority degrees. The high priority user (e.g., control messages, video, voice) has higher access possibility than the low priority user (e.g., data). This is realized by adding a dynamic priority to the priority class, instead of a constant priority. After SOR (successive over-relaxation) simulation the results illustrate that the improved dynamic priority queue is efficient to improve the performance of the total system.