Bandwidth-delay Product Research Articles

RTTV-TCP: Adaptive congestion control algorithm based on RTT variations for mmWave networks

Internet applications such as video gaming virtual/ augmented reality necessitate efficient fifth-generation (5G) millimeter-wave (mmWave) cellular networks. The Transmission Control Protocol (TCP) is an essential protocol for network connectivity. However, TCP faces challenges in efficiently utilizing the available bandwidth of 5G mmWave cellular networks while maintaining low latency, mainly due to constraints like Non-Line of Sight (NLoS) conditions. This paper introduces Round-Trip-Time Variations-TCP (RTTV-TCP), enhancing TCP performance in 5G mmWave cellular networks. Simulation scenarios for a 5G mmWave cellular network have been conducted to evaluate RTTV-TCP’s performance, comparing it to legacy TCP variants such as NewReno, HighSpeed, CUBIC, Bottleneck Bandwidth and Round-trip propagation time (BBR), FB-TCP (Fuzzy Based-TCP). The results demonstrate that RTTV-TCP achieves higher average throughput than these TCP variants while maintaining the same level of delay in 5G mmWave cellular networks. RTTV-TCP outperforms NewReno and CUBIC by a very significant margin, demonstrating a 208% improvement compared to HighSpeed and a 6% increase compared to BBR protocol in the worst Packet Error Rate (PER) scenario and when the buffer size matches the Bandwidth Delay Product (BDP).

Experimental validation of group delay in a multi-window electromagnetically induced transparency metasurface

Metasurface analogs of electromagnetically induced transparency (EIT) are attracting sustaining attention due to their ability to maintain transparency windows accompanied by extreme dispersion of propagating waves, which are important for slow light devices and highly sensitive optical sensors. In this paper, we study theoretically, numerically, and experimentally the conditions for the existence of multi-band transparency windows in the metasurface supported by the interaction of dipole modes in an asymmetric unit cell. The unit cell is composed of a single bright resonator and several dark resonators made in the form of rectangular metal patches. The manifestation of EIT is studied for different metasurface configurations by varying the number and positions of resonators used within the unit cell. To validate the slow-down effect caused by EIT, a prototype of the metasurface is fabricated and tested, providing a measurement of the group delay and bandwidth-delay product features. The obtained results clearly confirm the presence of four EIT-like transparency windows in the metasurface transmission spectra originating from the coupling between either quasi-TE or quasi-TM modes of the resonators.

Bandwidth-Delay-Product-Based ACK Optimization Strategy for QUIC in Wi-Fi Networks

QUIC has drawn extensive attention in supporting low latency and secure Internet of Things (IoT) communications due to its efficient handshake and default end-to-end encryption. However, in Wi-Fi enabled IoT communications with contentions for shared media, QUIC’s inherent acknowledgment (ACK) policy may induce non-negligible control overhead and limited data throughput. To address the problem, this paper designs and implements an ACK frequency optimization scheme for QUIC by exploiting the tailored bandwidth-delay product (BDP) at the receiver, named QUIC-BDP. To accurately estimate real-time BDP, we design an “ACK-PING" strategy to compensate for the accuracy of round-trip timing estimation and utilize exponential averaging and sliding window filtering for stable bandwidth estimation. Experiments results show that our proposed QUIC-BDP balances between the robustness and throughput performance while maintaining stable performance in lossy cases, with a reduced energy cost. Particularly, QUIC-BDP achieves up to a 67% gain in goodput compared to the original QUIC, and it improves goodput by up to 38% and 28% compared to existing solutions MSQUIC and QUIC-1:10, respectively. In addition, QUIC-BDP reduces energy cost by up to 50% compared to the original QUIC.

Sustainable Cross-Regional Transmission Control for the Industrial Augmented Intelligence of Things

Sustainable connectivity between an Augmented Intelligence of Things (AIoT) system and terminal devices can provide a strong guarantee for an enterprise's continuity of production. To ensure sustainable data interaction in emergency situations such as damage to communication infrastructure facilities in a postdisaster period, a pyramidal air–space–ground network model is proposed in this paper, and its transmission performance is investigated. The space network, constructed of geostationary (GEO) satellites and medium and low Earth orbit (MEO and LEO) satellites, is heterogeneously fused with the terrestrial AIoT network to ensure around-the-clock network service for enterprise production. Then, bandwidth–delay product (BDP) analysis is performed for this heterogeneous network. The proposed scheme improves the data transmission efficiency of the network model by mitigating the impact of a long round-trip time (RTT) and improving the congestion window optimization in the case of data loss. The proposed scheme increases the transmission rate in the network model by a factor of 3.72 compared to the traditional transmission scheme in the case of a 5-hop terrestrial AIoT network and has a 3.94-fold advantage in download time.

Optimization of BBR Congestion Control Algorithm Based on Pacing Gain Model.

In 2016, Google proposed a congestion control algorithm based on bottleneck bandwidth and round-trip propagation time (BBR). The BBR congestion control algorithm measures the network bottleneck bandwidth and minimum delay in real-time to calculate the bandwidth delay product (BDP) and then adjusts the transmission rate to maximize throughput and minimize latency. However, relevant research reveals that BBR still has issues such as RTT unfairness, high packet loss rate, and deep buffer performance degradation. This article focuses on its most prominent RTT fairness issue as a starting point for optimization research. Using fluid models to describe the data transmission process in BBR congestion control, a fairness optimization strategy based on pacing gain is proposed. Triangular functions, inverse proportional functions, and gamma correction functions are analyzed and selected to construct the pacing gain model, forming three different adjustment functions for adaptive adjustment of the transmission rate. Simulation and real experiments show that the three optimization algorithms significantly improve the fairness and network transmission performance of the original BBR algorithm. In particular, the optimization algorithm that employs the gamma correction function as the gain model exhibits the best stability.

Variable optical true-time delay line breaking bandwidth-delay constraints.

Continuously variable true-time optical delay lines are typically subject to a constraint of the bandwidth-delay product, limiting their use in several applications. In this Letter, we propose an integrated topology that breaks the bandwidth-delay product limit. The device is based on multiple Mach-Zehnder Interferometers (MZIs) arranged in parallel, providing easier control and a larger bandwidth compared to ring resonator-based solutions. The functionality of this architecture is demonstrated with a 4-stage delay line by performing measurements in both the time and frequency domains. The delay line introduces a delay of 90 ps over a bandwidth of more than 22 GHz with a negligible group delay distortion, operates on a wavelength range of about 60 nm, and is scalable to a higher number of MZI stages.

Coupled CUBIC Congestion Control for MPTCP in Broadband Networks

Recently, multipath transmission control protocol (MPTCP) was standardized so that data can be transmitted through multiple paths to utilize all available path bandwidths. However, when high-speed long-distance networks are included in MPTCP paths, the traffic transmission performance of MPTCP is severely deteriorated, especially in case the multiple paths’ characteristics are heavily asymmetric. In order to alleviate this problem, we propose a “Coupled CUBIC congestion control” that adopts TCP CUBIC on a large bandwidth-delay product (BDP) path in a linked increase manner for maintaining fairness with an ordinary TCP traversing the same bottleneck path. To verify the performance excellence of the proposed algorithm, we implemented the Coupled CUBIC Congestion Control into Linux kernels by modifying the legacy MPTCP linked-increases algorithm (LIA) congestion control source code. We constructed asymmetric heterogeneous network testbeds mixed with large and small BDP paths and compared the performances of LIA and Coupled CUBIC by experiments. Experimental results show that the proposed Coupled CUBIC utilizes almost over 80% of the bandwidth resource in the high BDP path, while the LIA utilizes only less than 20% of the bandwidth for the same path. It was confirmed that the resource utilization and traffic transmission performance have been greatly improved by using the proposed Coupled CUBIC in high-speed multipath networks, as well as maintaining MPTCP fairness with competing single-path CUBIC or Reno TCP flows.

Learning-Based Adaptive Sliding-Window RLNC for High Bandwidth-Delay Product Networks

Learning-Based Adaptive Sliding-Window RLNC for High Bandwidth-Delay Product Networks

BBR-With Enhanced Fairness (BBR-EFRA): A new enhanced RTT fairness for BBR congestion control algorithm

Towards the end of 2016, the Google research team proposed and developed a new state-of-the-art TCP congestion control algorithm called Bottleneck Bandwidth and Round-trip propagation time (BBR). When deployed on various Google internal servers, BBR attained higher throughput and low latency performances on modern-day and sophisticated networks than traditional congestion control algorithms like Cubic, Reno, or Vegas. Unlike conventional congestion control algorithms, BBR controls data transmission by maximizing delivery rate and minimizing the round-trip time (RTT), therefore maximizing bandwidth utilization and improving throughput and latency delay performances. However, some experiments have reported persistent queue formation and massive packet retransmissions rate in the bottleneck link, which happens to be the main cause of unfairness between different RTT flows. BBR prefers and favors long RTT over short RTT flows; therefore, long RTT flows are allocated more bandwidth when competing with short RTT flows in the bottleneck link. It was also noted that even the minor difference between the two competing flows could be a source of throughput imbalance and unfairness. The dominance of long RTT flows is the central origin of the high queuing delay, packet loss and retransmission rates, and even severe BBR vulnerability that malicious users can exploit to obtain a larger share of bandwidth simply by increasing the RTT (delay). Therefore, we propose a BBR-With Enhanced Fairness (BBR-EFRA) to mitigate this major concern challenge. Our proposed algorithm adaptively controls the congestion window (CWND) by adjusting the bandwidth delay product (BDP) values of each BBR flow during data transmission based on buffer queue status computation. Our proposed approach guarantees different RTT flows to compete for the available bottleneck bandwidth more equally, hence improving the BBR fairness issue. We evaluated our algorithm on the NS3 simulating environment. BBR-EFRA shows improved RTT fairness by more than 16%, reduced retransmission rate by more than 20%, and queuing delay by more than 18%, and finally increased Jain’s fairness index by more than 1.3 times compared with other recently published BBR variants like an adaptive congestion window of BBR(BBR-ACW). Therefore, with BBR-EFRA, short RTT flows can fairly compete for the available bandwidth even with higher RTT differences under various network scenarios.

Yinker: A flexible BBR to achieve the high-throughput and low-latency data transmission over Wi-Fi and 5G networks

The popularization and development of Wi-Fi and 5G networks have introduced new applications requiring high data rates and low latency. However, the vast random packet loss caused by mobility and channel conditions in wireless networks can worsen the performance of traditional TCP congestion control algorithms. Therein, BBR [1] was proposed in 2016, and claims to operate at the optimum point. BBR strives to match the congestion window to the bandwidth-delay product, calculated from the measured bottleneck bandwidth and round trip times. Through simulations, we found that BBR suffers from different degrees of throughput degradation for different loss rates. The improved BBR, Yinker, is proposed to address this issue. Precisely, we dynamically adjust BBR’s pacing_gain based on the network conditions, including loss rate and congestion degree. We have evaluated Yinker in both real-world environments and trace-based emulations and compared its performance with different BBR variants and state-of-the-art schemes, including Cubic, Verus, and Copa. On average, TCP D*, BBR v2, Copa, and BBR have 4.24×, 2.34×, 2.01×, and 1.48× lower throughput compared to Yinker, respectively. This outstanding delay performance comes at little cost in latency. For instance, compared to TCP D* (which achieves the lowest latency), Yinker’s latency is only about 3% more.

Transmission Control Over Satellite Network for Marine Environmental Monitoring System

The transmission of huge amounts of data generated from various sensors is the basis and prerequisite for the analysis and prediction of the marine ecosystem. The special geographical conditions of the ocean determine that its data transmission is different from the common terrestrial data transmission. In this study, a marine environmental monitoring network architecture is proposed to collect environmental data from marine ecosystems using a heterogeneous network constructed by a multi-hop network and a satellite network. Considering the characteristics of satellite networks, such as high bandwidth delay product (BDP), long latency and high bit error rate, this work proposes a transmission scheme that improves the transmission performance of Transmission Control Protocol/Internet Protocol (TCP/IP)-based Internet networks over satellite networks. The improved scheme increases the monitoring data transmission rate, distinguishes between random data loss and congested data loss, rationalizes the threshold in fast recovery, and improves the overall transmission performance of the marine environmental monitoring network. The experimental results show that the proposed scheme can greatly improve the transmission performance of the monitoring network compared with traditional transmission protocols. The network model proposed in this study makes full use of current advanced network technologies, including multi-hop networks, wireless sensor networks, and satellite networks to expand the monitoring area at sea. This study provides a useful reference for the network model and data transmission for ocean monitoring.

Cross-Regional Transmission Control for Satellite Network-Assisted Vehicular Ad Hoc Networks

Vehicular ad hoc networks (VANETs) have attracted increasing attention in cross-regional communication. The interconnection between VANETs and satellite networks expands their communication range. To improve the transmission performance of heterogeneous networks composed of VANETs and satellite networks, this paper proposes an end-to-end transmission control scheme for satellite network-assisted VANETs, analyzes the bandwidth delay product of multihop VANETs through heterogeneous networks, and establishes a transmission control model applicable to cross-regional environments. To address the long delay and high bit error rate in heterogeneous networks, the model designs the congestion window in the slow start and congestion avoidance of the transmission scheme. The data transmission volume is increased in the slow start, and various packet losses are distinguished in congestion avoidance. The experimental results show that the scheme can achieve good transmission performance in heterogeneous networks composed of VANETs and satellite networks. The results show that VANETs data can be effectively transmitted in heterogeneous networks. Compared with the traditional transmission scheme, this simple and practical end-to-end transmission control scheme can improve speed by 83.49 kbps and reduce transmission duration time of FTP server by <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$4.72\times 10^{3}$ </tex-math></inline-formula> s. The results of this study provide a reference for massive data transmission in heterogeneous networks of VANETs and satellite networks.

End-to-End Transmission Control for Cross-Regional Industrial Internet of Things in Industry 5.0

Data transmission for the industrial Internet of Things (IoT) is crucial for industrial production, especially in the Industry 5.0 era, where human–machine collaboration is increasingly intensive. To ensure the continuity and robustness of industrial IoT communications in the case of damaged infrastructure communication facilities postdisaster, the industrial IoT can be connected with satellite networks in emergencies. This article presents a cross-regional, end-to-end, transmission control scheme for satellite-supported, multihop industrial IoT. The proposed scheme adjusts the window of data transmission from two phases, slow start and congestion avoidance, to accommodate the low-transmission performance caused by a long delay and high bit error rate in converged networks. The window of data transmission is also adjusted to increase the amount of data transmission for the slow start to fill the high bandwidth-delay product of the converged network, while adjusting the threshold of data transmission based on feedback information to distinguish different data losses during congestion avoidance. The feasibility of the heterogeneous network transmission model is experimentally verified. The results show that the scheme can achieve good performance in heterogeneous networks of industrial IoT and satellite networks. The scheme is effective in ensuring the continuity and stability of intelligent machine production in Industry 5.0 in emergency communication cases.

Lightweight network-coded ARQ: An approach for Ultra-Reliable Low Latency Communication

Sliding Window RLNC schemes represent a novel, promising and efficient class of algorithms for reliable data transmission over an unreliable link. They combine Random Linear Network Coding (RLNC) with the idea of sliding window from (Automatic Repeat reQuest) ARQ protocols in order to help the basic ARQ mechanism to quickly recover from losses through redundant coded packets. Although very successful, we make the observation that such schemes bear significant limitations stemming from the fact that both the ARQ mechanism and the coding process operate on the same group of packets, called the sliding window. To tackle the problem, we propose the use of two distinct windows; the sliding window, which can be used to optimize the data flow based on the link’s bandwidth-delay product, and the coding window that can be used exclusively in the coding process. We analyze the performance of the proposed strategy and show that it provides an improved coding operation while at the same time reduces the coding complexity. Then, we delineate rapidARQ, a sliding window RLNC protocol that adopts the proposed strategy, and experimentally confirm that it outperforms other state-of-the-art sliding window RLNC schemes. More specifically, it achieves superior throughput-delay performance that better fits in the context of Ultra-Reliable Low-Latency Communication (URLLC) while at the same time it significantly reduces the coding complexity. Moreover, we show that the superior performance of rapidARQ is more prominent in channels with large bandwidth-delay products, a fact that renders its utility to current and future networks more essential.

Updating the theory of buffer sizing

Routers have packet buffers to reduce packet drops during times of congestion. It is important to correctly size the buffer: make it too small, and packets are dropped unnecessarily and the link may be underutilized; make it too big, and packets may wait for a long time, and the router itself may be more expensive to build. Despite its importance, there are few guidelines for picking the buffer size. The two most well-known rules only apply to long-lived TCP Reno flows; either for a network carrying a single TCP Reno flow (the buffer size should equal the bandwidth-delay product, or BDP) or for a network carrying n TCP Reno flows (the buffer size should equal BDP/n). Since these rules were introduced, TCP Reno has been replaced by newer algorithms as the default congestion control algorithm in all major operating systems, yet little has been written about how the rules need to change. This paper revisits both rules. For the single flow case, we generalize the BDP rule to account for changes to TCP, such as Proportional Rate Reduction (PRR), and the introduction of new algorithms including Cubic and BBR. We find that buffers can be made 60%–75% smaller for newer algorithms. For the multiple flow case, we show that the square root of n rule holds under a broader set of assumptions than previously known, including for these new congestion control algorithms. We also demonstrate situations where the square root of n rule does not hold, including for unfair flows and certain settings with ECN. We validate our results by precisely measuring the time series of buffer occupancy in a real network, and comparing it to the per-packet window size.

Performance analysis of high-speed TCP protocols in LTE X2 handover under realistic operational conditions

Broadband mobile networks have rapidly evolved over the last years. The unique environment they operate, however, sets new challenges for the Transmission Control Protocol (TCP). In this paper, we study the impact of Long Term Evolution (LTE) handover on TCP performance with different TCP congestion control algorithms and in a wide range of networks settings, including different handover types, random channel errors and support of error correction at lower layers. Our results suggest that the congestion control algorithms employed by TCP variants during the LTE X2 handover have different impacts on TCP performance. In particular CUBIC, Scalable TCP and BIC-TCP achieve high goodput and fast window convergence in available channel capacity after LTE X2 handover. In environments involving packet drops that are not corrected at lower layers and single errors with high probability, all TCP variants exhibit very low performance levels. Finally, in terms of fairness, our results identify fairness issues among TCP flows with different Bandwidth Delay Products (BDP), with CUBIC offering the best results.

An online learning based path selection for multipath real‐time video transmission in overlay network

AbstractEven video telephony services have been pervasively applied, it is still a challenge to provide satisfactory quality of experience to users. Multipath transmission is a promising solution to improve video transmission quality by providing aggregated bandwidth. Due to varied traffic load, the default paths may fall into long lasting congestion. Sender has no choice but to reduce video bitrate to adapt such change, even in multipath transmission scenario. Insisting previous rate may lead the congestion in bottleneck deteriorate further. Deploying relay servers in current Internet infrastructure provides users with alternatives to bypass path failure and congestion. For such transmission scenario for video telephony service, an arising question is how to select paths dynamically to gain maximum profit. The path selection is mapped to the multi‐armed bandit problem. Tradeoff is made between exploration and exploitation to make decision under uncertain environment. The estimated bandwidth provided by congestion controller is used for decision‐making. Further, to make multiple flows share resource fairly and avoid congestion, an algorithm adapted from BBR is implemented for video telephony service. To maintain throughput stability and fairness, a smaller probe up gain value is used and the cycle length in bandwidth probe phase is randomized. To reduce delay, the inflight packets will be drained to match with bandwidth delay product in the probe down phase. The effectiveness of the proposed solution is verify on simulation platform.

Right buffer sizing matters: Some dynamical and statistical studies on Compound TCP

Large and unmanaged router buffers could lead to an increase in queuing delays in the Internet, which is a serious concern for network performance and quality of service. Our focus is to conduct a performance evaluation of Compound TCP (C-TCP), in a regime where the router buffer sizes are small (i.e., independent of the bandwidth-delay product), and the queue policy is Drop-Tail. In particular, we provide buffer sizing recommendations for high speed core routers fed by well multiplexed TCP controlled flows. For this, we consider two topologies: a single bottleneck and a multi-bottleneck topology, under different traffic scenarios. The first topology consists of a single bottleneck router, and the second consists of two distinct sets of TCP flows, regulated by two edge routers, feeding into a common core router. We focus on some key dynamical and statistical properties of the underlying system. From a dynamical perspective, we first develop fluid models. A local stability analysis for these models yields a key insight: buffer sizes need to be dimensioned carefully, and smaller buffers favour stability. We also highlight that larger Drop-Tail buffers, in addition to increasing latency, are prone to inducing limit cycles in the system dynamics. These limit cycles in turn induce synchronisation among the TCP flows, which then results in a loss of link utilisation. We then empirically analyse some statistical properties of the bottleneck queues. These statistical analyses serve to validate an important modelling assumption: that in the regime considered, each bottleneck queue may be reasonably well approximated as either an M∕M∕1∕B or an M∕D∕1∕B queue. We also highlight that smaller buffers, in addition to ensuring stability and low latency, would also yield reasonable system-wide performance, in terms of throughput and flow completion times.

NexGen D-TCP: Next Generation Dynamic TCP Congestion Control Algorithm

With the advancement of wireless access networks and mmWave New Radio (NR), new applications emerged, which requires a high data rate. The random packet loss due to mobility and channel conditions in a wireless network is not negligible, which degrades the significant performance of the Transmission Control Protocol (TCP). The TCP has been extensively deployed for congestion control in the communication network during the last two decades. Different variants are proposed to improve the performance of TCP in various scenarios, specifically in lossy and high bandwidth-delay product (high-BDP) networks. Implementing a new TCP congestion control algorithm whose performance is applicable over a broad range of network conditions is still a challenge. In this article, we introduce and analyze a Dynamic TCP (D-TCP) congestion control algorithm over mmWave NR and LTE-A networks. The proposed D-TCP algorithm copes up with the mmWave channel fluctuations by estimating the available channel bandwidth. The estimated bandwidth is used to derive the congestion control factor N. The congestion window is increased/decreased adaptively based on the calculated congestion control factor. We evaluated the performance of D-TCP in terms of congestion window growth, goodput, fairness and compared it with legacy and existing TCP algorithms. We performed simulations of mmWave NR during LOS NLOS transitions and showed that D-TCP curtails the impact of under-utilization during mobility. The simulation results and live air experiment points out that D-TCP achieves 32.9% gain in goodput as compared to TCP-Reno and attains 118.9% gain in throughput as compared to TCP-Cubic.

Sizing router buffers (redux)

The queueing delay faced by a packet is arguably the largest source of uncertainty during its journey. It therefore seems crucial that we understand how big the buffers should be in Internet routers. Our 2004 Sigcomm paper revisited the existing rule of thumb that a buffer should hold one bandwidth-delay product of packets. We claimed that for long-lived TCP flows, it could be reduced by √ N , where N is the number of active flows, potentially reducing the required buffers by well over 90% in Internet backbone routers. One might reasonably expect that such a result, which supports cheaper routers with smaller buffers, would be embraced by the ISP community. In this paper we revisit the result 15 years later, and explain where it has succeeded and failed to affect how buffers are sized.