Real-time communication (RTC) applications demand consistent low latency to ensure a smooth and interactive user experience. However, wireless networks, including WiFi and cellular, although they provide satisfactory median latency, often suffer from significant tail latency due to the highly variable network bandwidth. We observe that the control loop for managing the sending rate of RTC applications becomes inflated when congestion occurs at the wireless access point (AP), leading to untimely rate adaptation in response to wireless dynamics. Existing solutions fail to quickly adapt to bandwidth fluctuations due to the inflated control loop. In this paper, we propose Zhuge, a purely wireless AP-based solution that addresses these issues by separating congestion feedback from congested queues. Our approach involves the design of a Fortune Teller, which accurately estimates the wireless latency for each packet upon its arrival at the wireless AP. To ensure scalability, we also develop a Feedback Updater that translates the estimated latency into understandable feedback messages for various end-to-end protocols, delivering them back to the senders immediately for rate adaptation. Our evaluation, based on both trace-driven simulations and real-world scenarios, demonstrates that Zhuge significantly reduces the occurrence of large tail latency and alleviates RTC performance degradation by 22% to 95%.

Active queue management (AQM) algorithms, widely deployed in the internet, are designed to signal end hosts with network conditions in the format of packet losses. However, real-time communication (RTC) applications adopt delay-sensitive congestion control algorithms (CCAs), which are no longer responsive to losses or explicit notifications from AQMs. Moreover, packet losses introduced by different AQMs will further degrade the performance of RTC applications due to unexpected and unnecessary loss recovery. We are therefore motivated to understand the behaviors of AQMs and take necessary countermeasures for RTC applications proactively. For example, with the help of AQM inference, RTC applications will benefit by using loss recovery mechanisms that adapt to various kinds of AQMs to deal with packet losses. However, it is challenging to infer the AQM from end hosts since numerous AQMs have different configurations after decades of evolution. We analyze the temporal behaviors of loss series, extract the inherent invariant features of different AQMs, and categorize them into three types. Our simulation shows that AQM inference can classify AQMs with an accuracy of 96%. We also evaluate a use case on using the AQM inference to improve the loss recovery mechanism (forward error correction, FEC). Our FEC method based on AQM inference improves the recovery rate by at least 56%, and finally reduces the end-to-end delay by 13%.

Many video streaming applications will simultaneously transfer data in both directions, from the user to the Internet (uplink) and from the Internet to users (downlink). However, for wireless local area networks (WLANs), the dominant scenarios, the uplink and downlink flows share the same half-duplex physical channel and compete for bandwidth resources. Their bandwidths would be fairly apportioned under the existing link layer access method, but a fair share might be suboptimal for applications. For better application performance, we propose Plum, to coordinate the bitrate of uplink and downlink flows, and allocate the bandwidth in both directions to cater to the application's demands. To make the deployment of Plum practical, we aim at not modifying the link layer but optimizing the transport layers and above. We evaluate our mechanisms with simulations based on real-world traces and testbed experiments, and results show that Plum could improve the video bitrate of streaming applications by up to 48-59%.

Containers have become the de facto choice for major cloud services. Meanwhile, with demands for extremely high performance, data centers have widely adopted RDMA for their online services. RDMA virtualization is the critical technology that enables RDMA for containers. Hybrid RDMA virtualization leverages the software flexibility in the control path, and keeps the native performance in the data path. Thus, it is the best choice for RDMA virtualization. State-of-the-art hybrid RDMA virtualization cannot address containerspecific problems. This paper proposes DockRDMA, the first hybrid RDMA virtualization solution for containerized clouds. DockRDMA develops several mechanisms, including embedding physical addresses in virtual ones to provide efficient address translation, hybrid network policy enforcement at scale, a general virtual RDMA NIC initialization method to be compatible with all container platforms, and namespace checking to protect the RDMA NIC instances. Evaluation results show that DockRDMA provides bare-metal RDMA performance in the data path, and almost native communication setup time in the control path. Compared with the state-of-the-art hybrid virtualization technology, DockRDMA reduces Hadoop job completion time by 6%. It offers seamless integration with existing container platforms, protects critical information of RDMA NIC instances, and exhibits excellent scalability to meet diverse network policies required by different containers.

Datacenter applications increasingly demand microsecond-scale latency and tight tail latency. Despite recent advances in datacenter transport protocols, we notice that the timeout caused by packet loss is the killer of microsecond-scale latency. Moreover, refining the RTO setting is impractical due to the significant fluctuations in RTT. In this paper, we propose explicit dropping notification (EDN) to avoid timeouts. EDN rekindles ICMP Source Quench, where the switch notifies the source of precise packet loss information. Then the source can rapidly pinpoint dropped packets for fast retransmission instead of waiting for timeouts. More importantly, fast retransmission does not mean immediate retransmission which is prone to aggravate congestion and deteriorate latency. In light of this, we suggest finessing the timing and sending rate of retransmission. Specifically, as a reward of the paradigm shift to explicit notification, the source can pause for the queue draining time piggybacked on EDN messages and estimate connection capacity to figure out a proper sending rate, thus avoiding congestion aggravation. We implement EDN on the P4-programmable switching ASIC and Linux kernel. Evaluations show that, compared with state-of-the-art loss recovery schemes, EDN reduces the latency by up to 4.1× on average and 3.6× at the 99th-percentile.

To ensure high performance for HTTP adaptive streaming (HAS), it is critical to provide accurate prediction of end-to-end network bandwidth. Low Latency Live Streaming (LLLS), which has been gaining popularity, faces even greater challenges in this regard. Unlike Video-on-Demand (VOD) streaming, which only needs long-term bandwidth prediction and can tolerate some prediction errors, LLLS demands precise short-term bandwidth predictions. These challenges are amplified by the fact that short-term bandwidth experiences both large abrupt changes and uncertain fluctuations. Furthermore, obtaining valid bandwidth measurement samples in LLLS poses difficulties due to the on-off traffic pattern. In this work, we present DeeProphet, a system designed to enhance the performance of LLLS by achieving accurate bandwidth prediction. DeeProphet collects valid bandwidth samples by identifying intervals of packet continuous sending leveraging TCP state information, estimates the segment-level bandwidth robustly by filtering out noisy samples, and predicts both significant changes and uncertain fluctuations in future bandwidth by combining both time series and learning-based models. Experimental results demonstrate that DeeProphet effectively enhances the overall Quality of Experience (QoE) by 39.5% to 464.6% compared to state-of-the-art LLLS Adaptive Bitrate (ABR) algorithms.

The performance of HTTP adaptive streaming (HAS) depends heavily on the prediction of end-to-end network bandwidth. The increasingly popular low latency live streaming (LLLS) faces greater challenges since it requires accurate, short-term bandwidth prediction, compared with VOD streaming which needs long-term bandwidth prediction and has good tolerance against prediction error. Part of the challenges comes from the fact that short-term bandwidth experiences both large abrupt changes and uncertain fluctuations. Additionally, it is hard to obtain valid bandwidth measurement samples in LLLS due to its inter-chunk and intra-chunk sending idleness. In this work, we present DeeProphet, a system for accurate bandwidth prediction in LLLS to improve the performance of HAS. DeeProphet overcomes the above challenges by collecting valid measurement samples using fine-grained TCP state information to identify the packet bursting intervals, and by combining the time series model and learning-based model to predict both large change and uncertain fluctuations. Experiment results show that DeeProphet improves the overall QoE by 17.7%-359.2% compared with state-of-the-art LLLS ABR algorithms, and reduces the median bandwidth prediction error to 2.7%.

Real-time communication (RTC) applications like video conferencing or cloud gaming require consistent low latency to provide a seamless interactive experience. However, wireless networks including WiFi and cellular, albeit providing a satisfactory median latency, drastically degrade at the tail due to frequent and substantial wireless bandwidth fluctuations. We observe that the control loop for the sending rate of RTC applications is inflated when congestion happens at the wireless access point (AP), resulting in untimely rate adaption to wireless dynamics. Existing solutions, however, suffer from the inflated control loop and fail to quickly adapt to bandwidth fluctuations. In this paper, we propose Zhuge, a pure wireless AP based solution that reduces the control loop of RTC applications by separating congestion feedback from congested queues. We design a Fortune Teller to precisely estimate per-packet wireless latency upon its arrival at the wireless AP. To make Zhuge deployable at scale, we also design a Feedback Updater that translates the estimated latency to comprehensible feedback messages for various protocols and immediately delivers them back to senders for rate adaption. Trace-driven and real-world evaluation shows that Zhuge reduces the ratio of large tail latency and RTC performance degradation by 17% to 95%.

Live video applications are getting popular, and content providers widely use adaptive bitrate (ABR) streaming to improve QoE while maintaining low latency. However, users’ increasing preference to watch videos on mobile devices poses great challenges for ABR algorithm due to the dramatically varying cellular network. Existing learn-based ABR algorithms face difficulties to generalize to various network conditions because of their reliance on training traces, and model/rule-based ABR schemes suffer from rebuffering under low latency constraint since they cannot robustly control the buffer occupancy within a small range. To address it, this work proposes Cratus, a lightweight and robust ABR algorithm for mobile live streaming, which achieves high QoE and low latency by accurately regulating the buffer at a small level. To enhance the control ability, Cratus controls the buffer dynamic behavior rather than the buffer occupancy. By using sliding mode control approach, Cratus robustly controls the buffer dynamic and ensures that the buffer occupancy is bounded around the target level regardless of network uncertainties. Trace-driven experiments show that Cratus outperforms existing ABRs: average QoE is increased by 12.3 to 28.6 percent, and rebuffering time is limited within 0.8$s$on average, which is reduced by 53.5 to 92.3 percent.

Most video players use adaptive bitrate (ABR) algorithms to provide good quality-of-experience (QoE) in dynamic network conditions. To deal with the adaptation challenges, many ABR algorithms select bitrate by optimizing a defined QoE function. Within the framework, various algorithms mainly differ in how the optimization problem is solved, including prediction-based approaches and learn-based approaches. However, these algorithms suffer from limited performance in the current popular mobile streaming which has limited resources and rapidly changing link rates. Existing machine-learning approaches face deployment difficulties on mobile devices, and prediction-based approaches that rely on throughput prediction experience large buffer occupancy variations in cellular networks, resulting in rebuffering frequently. To provide a robust and lightweight ABR algorithm for mobile streaming, this work improves the robustness of prediction-based scheme against unpredictable network variations and develops RBC (Robust Bitrate Controller) algorithm. Rather than optimizing QoE over the entire buffer capacity, RBC creates buffer margins to absorb the impact of throughput jitters and solves QoE maximization on the narrowed buffer range. The amount of buffer margin is dynamically adjusted based on the real-time throughput fluctuation to ensure sufficient de-jitter space. For online lightweight deployment, RBC provides a closed-form solution of the desired bitrate with small computation complexity by using adaptive control approach. Trace-driven experiments and real-world tests show that RBC effectively reduces the playback freezing and gains an improvement in overall QoE.

With the development of cryptocurrencies’ market, the problem of cryptojacking, which is an unauthorized control of someone else’s computer to mine cryptocurrency, has been more and more serious. Existing cryptojacking detection methods require to install anti-virus software on the host or load plug-in in the browser, which are difficult to deploy on enterprise or campus networks with a large number of hosts and servers. To bridge the gap, we propose MineHunter, a practical cryptomining traffic detection algorithm based on time series tracking. Instead of being deployed at the hosts, MineHunter detects the cryptomining traffic at the entrance of enterprise or campus networks. Minehunter has taken into account the challenges faced by the actual deployment environment, including extremely unbalanced datasets, controllable alarms, traffic confusion, and efficiency. The accurate network-level detection is achieved by analyzing the network traffic characteristics of cryptomining and investigating the association between the network flow sequence of cryptomining and the block creation sequence of cryptocurrency. We evaluate our algorithm at the entrance of a large office building in a campus network for a month. The total volumes exceed 28 TeraBytes. Our experimental results show that MineHunter can achieve precision of 97.0% and recall of 99.7%.

The intranets in modern organizations are facing severe data breaches and critical resource misuses. By reusing user credentials from compromised systems, Advanced Persistent Threat (APT) attackers can move laterally within the internal network. A promising new approach called deception technology makes the network administrator (i.e., defender) able to deploy decoys to deceive the attacker in the intranet and trap him into a honeypot. Then the defender ought to reasonably allocate decoys to potentially insecure hosts. Unfortunately, existing APT-related defense resource allocation models are infeasible because of the neglect of many realistic factors.In this paper, we make the decoy deployment strategy feasible by proposing a game-theoretic model called the APT Deception Game to describe interactions between the defender and the attacker. More specifically, we decompose the decoy deployment problem into two subproblems and make the problem solvable. Considering the best response of the attacker who is aware of the defender’s deployment strategy, we provide an elitist reservation genetic algorithm to solve this game. Simulation results demonstrate the effectiveness of our deployment strategy compared with other heuristic strategies.

Video streaming over mobile is flourishing, and most commercial players use adaptive bitrate (ABR) streaming to deliver video in varying network conditions. Using network capacity and buffer occupancy as system states, ABR algorithms adjust bitrate based on the instantaneous system states, which is able to adapt to network changes in real-time and ensure high quality of experience (QoE). However, they are incapable of providing good QoE over mobile. Due to the high dynamic characteristics of cellular network, the system states change rapidly over time. The instantaneous state-based adaptation can induce significant video quality fluctuation which greatly degrades QoE. In this paper, we propose an online ABR algorithm called MSPC to provide good QoE in cellular network. To balance the conflict between rapid adaptation and smooth bitrate, MSPC utilizes the multi-step prediction of future system states to select bitrates instead of the instantaneous current states. At the same time, it controls the buffer occupancy to eliminate the impact of prediction error on performance. We implement MSPC on a reference video player with performance evaluated based on realistic cellular traces. Experimental results show that MSPC reduces the bitrate change of existing online algorithms by 62.4 percent on average while maintaining high bitrates and achieving zero rebuffering over 97.83 percent of all tested sessions.

Today, live streaming is gaining a rapid growth in use, which refers to streaming the media content recorded and broadcast in real time. In live streaming, latency is of utmost importance since smaller latency means higher user engagement. HTTP adaptive streaming (HAS) is now the most popular live streaming technology, where the video client sends HTTP requests to server to download video segments. The bitrate adaptation (ABR) algorithm inside the client determines bitrate level for every segment. It is of great help for ABR algorithm to quantify the influence of different HAS factors on streaming performance. However, existing work mainly focuses on video on demand (VoD) streaming rather than live streaming. In this paper, we theoretically analyze live streaming performance. We first establish a queuing model to describe playout buffer evolution. Based on the model, we respectively characterize rebuffering probability, rebuffering count and streaming latency, and analyze the effects of chunk arrival rate, arrival interval fluctuation, startup threshold and video skipping on them. From analysis results, we propose insights and recommendations for bitrate adaptation in live streaming and design a simple heuristic ABR algorithm leveraging them. Extensive simulations verify the insights as well as effectiveness of the designed algorithm.

Most video players use adaptive bitrate (ABR) algorithms to ensure good quality-of-experience (QoE) across diverse network conditions. To balance conflicting QoE factors, state-of-the-art ABR algorithms select bitrate by optimizing a defined QoE function. However, this scheme relies on throughput prediction that is sensitive to network conditions, so the achieved QoE can be poor in unstable networks. In this paper, we propose a robust ABR algorithm called RBC to avoid the impact of prediction error on QoE. RBC controls the buffer occupancy within a safe range while maximizing the QoE, and employs an adaptive bitrate controller to ensure a good control performance in various network condition. Trace-driven experiments show that RBC achieves much less playback freezing and an improvement of 13.5% on average QoE over the best approach RobustMPC, which confirms the effectiveness of buffer control and adaptive controller in improving system robustness against network uncertainty.

Video content providers are increasingly interested in interactive live streaming since user engagement increases their revenues. To provide high quality of experience (QoE), it is critical to design a low delay adaptive bitrate (ABR) algorithm, but which is lacked in existing studies. The low delay constraint poses much more challenges to achieve high bitrate and low rebuffering. For example, low delay requires the player to maintain a small playback buffer, which, however, increases the risk of rebuffering. This work designs a low delay ABR algorithm called HCA which provides good QoE by controlling the buffer occupancy at a low but non-empty level. To achieve accurate control, HCA uses the hybrid of feedback and feedforward control to regulate the buffer dynamic (buffer occupancy and its variation) based on predictions of future network condition (throughput and its variation). Trace-driven experiments show that HCA achieves zero rebuffering for 98% of all traces while ensuring high bitrate.

In recent years, the coflow abstraction has received significant attentions, for its prominent ability to capture application semantics. On this basis, multiple coflow scheduling mechanisms have been proposed to minimize the coflow completion time (CCT). Currently, existing coflow scheduling mechanisms mainly belong to two categories: information-omniscient and information-agnostic. However, in data center applications, there are still quite a few cases in between where incomplete coflow information is known, and such incomplete information makes great contributions to improving the CCT performance. To address such cases, we propose IICS, a coflow scheduling algorithm based on incomplete coflow information. IICS leverages information of a coflow's arrived parts to deduce the coflow's remaining transmission time, and uses it to approximate the Minimum Remaining Time First (MRTF) heuristic. Besides, IICS allocates bandwidth by monopolization and in a maximal manner, which achieves high bandwidth utilization. Extensive simulations under realistic settings show that IICS achieves the average CCT comparable to that of the information-omniscient algorithm and the 99th percentile CCT much smaller than both information-omniscient and information-agnostic algorithms. Furthermore, IICS holds observably higher throughput and is robust to algorithm parameters.

More and more commercial video players use bitrate adaptation to adjust video quality according to varying network conditions. Optimization-based approaches are widely used for bitrate adaptation in Dynamic Adaptive Streaming over HTTP (DASH). Essentially, the optimization problem is solved based on the prediction of buffer dynamics. However, stochastic chunk size deviates observably the buffer occupancy from the expected value, making the evolution hard to predict. In order to get rid of this effect and improve the prediction accuracy for buffer occupancy, we propose an algorithm based on markov decision process with incorporating chunk size information so that only the network capacity variation need to be considered in the decision-making process. Experiment results show that our solution can effectively eliminate performance oscillation induced by variable chunk size and achieve a good QoE.

Congestion control is an indispensable mechanism in the new trend of enhanced Ethernet as a unified fabric for traditional LAN, SAN, and high-performance computing networks. A congestion management framework for Converged Ethernet (CE) networks has been standardized by IEEE 802.1 Qau work group, and QCN is recommended as the congestion control scheme in the standard draft. QCN is heuristically designed for 1/10Gbps Ethernet without considering the impact of delays. Recent work find that QCN will encounter stability issues with feedback delays, and these issues will be more serious as Ethernet extends to 40/100Gbps and the delays become heterogeneous and time-varying. This work aims to mitigate the negative impact of delays on congestion control scheme in CE. Specially, considering the delays are heterogeneous and time-varying, we build a model for Converged Ethernet with the standard congestion management framework. The model provides a new congestion detector to estimate the real congestion status under the impact of delays and regards the heterogeneous and time-varying feature as disturbances. Leveraging the new congestion detector and tolerating the disturbance through the sliding mode control method, we design the Delay-tolerant Sliding Mode (DSM) congestion control scheme. Extensive simulations show that DSM outperforms other congestion control schemes when the Ethernet ranges from 1Gbps to 100Gbps and the delays are heterogeneous and time-varying.

Owing to Ethernet's low cost, high bandwidth and architecture openness, much attention has been paid to develop converged Ethernet to support both time-critical services and conventional communication services on a unified network infrastructure. The greatest challenge here is providing low and deterministic latency for time-critical packets. Recently, the IEEE time sensitive networking task group is launched to address it. However, their framework is complex and unsuitable for commodity switch architecture. In this paper, we propose a concise and efficient converged real-time Ethernet framework called XpressEth, which leverages Dual Preemption mechanism to minimize the delay of time-critical packets, and employs a lightweight Slot Assignment Scheduler to minimize the conflicts among time-critical packets at sources. XpressEth cuts off great burden from both forwarding and scheduling. The simulation results verify that XpressEth can provide ultra-low and deterministic latency for time-critical packets (1.024μ s per hop and zero jitter in 1Gbps network), which is 13× better than time sensitive networking solution, and the side-effect on conventional communication traffic is negligible.

The shuffle transfer pattern is widely adopted in today's cluster computing applications and the completion time of each group of transmissions directly affects application performance. Because of the restriction on the number of concurrent threads and the TCP Incast problem, the randomized data fetching strategy is widely employed in this kind of communication in practice. In this paper, to assess the performance of randomized data fetching, we build a general analytical model and define two metrics - link overload probability and K-deviation load balancing probability - to evaluate the degree of link overload and load balancing respectively, since they are closely related to the transfer completion time. Leveraging our model, we theoretically analyze the transfer performance in three typical scenarios and provide recommendations for setting the number of concurrent connections per receiver. Finally, we validate the theoretical analysis as well as the recommendations through extensive simulations.

Many commercial video players rely on bitrate adaptation algorithm to adapt video bitrate to dynamic network condition. To achieve a high quality of experience, bitrate adaptation algorithm is required to strike a balance between response agility and video quality stability. Existing online algorithms select bitrates according to instantaneous throughput and buffer occupancy, achieving an agile reaction to changes but inducing video quality fluctuations due to the high dynamic of reference signals. In this paper, the idea of multi-step prediction is proposed to guide a better tradeoff, and the bitrate selection is formulated as a predictive control problem. With it, a generalized predictive control based approach is developed to calculate the optimal bitrate by minimizing the cost function over a moving look-ahead horizon. Finally, the proposed algorithm is implemented on a reference video player with performance evaluations conducted using realistic bandwidth traces. Experimental results show that the multi-step predictive control adaptation algorithm can achieve zero rebuffer event and 63.3% of reduction in bitrate switch.