A novel numerical algorithm based on self-tuning controller to support TCP flows

Active queue management (AQM) is an effective method used in Internet routers for congestion control, and to achieve a trade off between link utilization and delay. The de facto standard, the random early detection (RED) AQM scheme, and most of its variants use average queue length as a congestion indicator to trigger packet dropping.     

Reducing Queue Oscillation at a Congested Link

Queue length oscillation at a congested link causes many undesirable properties such as large delay jitter, underutilization of the link and packet drops in burst. The main reason of this oscillation is that most queue management schemes determine the drop probability based on the current traffic without consideration on the impact of that drop probability on the future traffic. In this paper, we propose a new active queue (AQM) scheme to reduce queue oscillation and realize stable queue length. The proposed scheme measures the current arrival and drop rates, and uses them to estimate the next arrival rate. Based on this estimation, the scheme calculates the drop probability which is expected to realize stable queue length. We present extensive simulation with various topologies and offered traffic to evaluate performance of the proposed scheme. The results show that the proposed scheme remarkably reduces queue length oscillation compared to other well-known AQMs. It is also shown that the proposed scheme improves fairness among TCP flows due to the stable drop probability, and maintains high utilization with small queue length.     

Incorporating packet semantics in scheduling of real-time multimedia streaming

In this work, we develop a novel packet scheduling algorithm that properly incorporates the semantics of a packet. We find that improvement in overall packet loss does not necessarily coincide with improvement in user perceivable QoS. The objective of this work is to develop a packet scheduling mechanism which can improve the user perceivable QoS. We do not focus on improving packet loss, delay, or burstiness. We develop a metric called, “Packet Significance,” that effectively quantifies the importance of a packet that properly incorporates the semantics of a packet from the perspective of compression. Packet significance elaborately incorporates inter-frame, intra-frame information dependency, and the transitive information dependency characteristics of modern compression schemes. We apply packet significance in scheduling the packet. In our context, packet scheduling consists of two technical ingredients: packet selection and interval selection. Under limited network bandwidth availability, it is desirable to transmit the subset of the packets rather than transmitting the entire set of packets. We use a greedy approach in selecting packets for transmission and use packet significance as the selection criteria. In determining the transmission interval of a packet, we incorporate the packet significance. Simulation based experiments with eight video clips were performed. We embed the decoding engine in our simulation software and examine the user perceivable QoS (PSNR). We compare the performance of the proposed algorithm with best effort scheduling scheme and one with simple QoS metric based scheduling scheme. Our Significance-Aware Scheduling scheme (SAPS) effectively incorporates the semantics of a packet and delivers best user perceivable QoS. SAPS can result in more packet loss or burstier traffic. Despite these limitations, SAPS successfully improves the overall user perceivable QoS.     

A traffic-class burst-polling based delta DBA scheme for QoS in distributed EPONs

Ethernet passive optical network (EPON) preserves the merits of traditional Ethernet network while reducing complexities and improving quality of service (QoS). In this paper, a traffic-class burst-polling based delta dynamic bandwidth allocation (TCBP-DDBA) scheme is presented to provide better QoS to expedited forwarding packet and maximize channel utilization service to assure forwarding and best effort packets. The network resources are efficiently utilized and adaptively allocated to the three traffic classes by guaranteeing the requested QoS. Simulation results using OPNET show that the TCBP-DDBA scheme performs well in comparison to the conventional allocation scheme for a set of given parameters such as: packet delay, queue size, packet delay variation and channel utilization. This work considers system-wide DBA development in contrast to unit-based approach. It is concluded that the algorithm can be used for many types of EPON-based practical distributed networks.     

Header Detection to Improve Multimedia Quality Over Wireless Networks

Wireless multimedia studies have revealed that forward error correction (FEC) on corrupted packets yields better bandwidth utilization and lower delay than retransmissions. To facilitate FEC-based recovery, corrupted packets should not be dropped so that maximum number of packets is relayed to a wireless receiver's FEC decoder. Previous studies proposed to mitigate wireless packet drops by a partial checksum that ignored payload errors. Such schemes require modifications to both transmitters and receivers, and incur packet-losses due to header errors. In this paper, we introduce a receiver-based scheme which uses the history of active multimedia sessions to detect transmitted values of corrupted packet headers, thereby improving wireless multimedia throughput. Header detection is posed as the decision-theoretic problem of multihypothesis detection of known parameters in noise. Performance of the proposed scheme is evaluated using trace-driven video simulations on an 802.11b local area network. We show that header detection with application layer FEC provides significant throughput and video quality improvements over the conventional UDP/IP/802.11 protocol stack     

General packet induced queueing schemes for reducing packet delays in ADSL routers with peer-to-peer file sharing applications

This paper proposes a class of queueing schemes named general packet induced queueing schemes (GPIQS) in ADSL routers to reduce the queueing delays of non-P2P packets. The objective of the proposed queueing schemes is to send out the general packets first as well as P2P packets are able to be sent in a bounded queueing delay. The proposed queueing schemes use the general packet to induce the transmission of P2P packets which are from the same client and arrived at the ADSL router before the general packet. The outbound order of the packets transmitted from a specific client is not altered in the proposed schemes. Two queueing schemes named general packet induced queueing scheme with single P2P queue (GPIQS-SQ) and general packet induced queueing scheme with multiple P2P queues (GPIQS-MQ) are proposed. The two proposed queueing schemes differ in the number of P2P queues. In order to prevent the unlimited waiting time of P2P packets, we introduced a variable called the largest number of preempting packets to send out the P2P packets in a bounded time. Simulation results show that the proposed queueing schemes may send out the packets from ADSL router efficiently and the average queueing delay is smaller than the common used first-come first-served algorithm. Specifically, the GPIQS-MQ performs better than the GPIQS-SQ method in terms of average queueing delay of non-P2P packets. We also found that the increased average queueing delay of P2P packets is small. Finally, the values of the largest number of preempting packets are discussed.     

Multi-indicator Active Queue Management Method

A considerable number of applications are running over IP networks. This increased the contention on the network resource, which ultimately results in congestion. Active queue management (AQM) aims to reduce the serious consequences of network congestion in the router buffer and its negative effects on network performance. AQM methods implement different techniques in accordance with congestion indicators, such as queue length and average queue length. The performance of the network is evaluated using delay, loss, and throughput. The gap between congestion indicators and network performance measurements leads to the decline in network performance. In this study, delay and loss predictions are used as congestion indicators in a novel stochastic approach for AQM. The proposed method estimates the congestion in the router buffer and then uses the indicators to calculate the dropping probability, which is responsible for managing the router buffer. The experimental results, based on two sets of experiments, have shown that the proposed method outperformed the existing benchmark algorithms including RED, ERED and BLUE algorithms. For instance, in the first experiment, the proposed method resides in the third-place in terms of delay when compared to the benchmark algorithms. In addition, the proposed method outperformed the benchmark algorithms in terms of packet loss, packet dropping, and packet retransmission. Overall, the proposed method outperformed the benchmark algorithms because it preserves packet loss while maintaining reasonable queuing delay.     

Design and analysis of an ONOFF variable structure controller for AQM routers supporting TCP flows

Active queue management （AQM） can maintain smaller queuing delay and higher throughput by purposefully dropping packets at intermediate nodes. Most of the existing AQM schemes follow the probability dropping mechanism originating from random early detection （RED）. This paper develops a novel packet dropping mechanism for AQM through designing an ONOFF controller applying the variable structure control theory. Because the binary ONOFF controller can considerably simplify the manipulation on the AQM router, it is helpful for implementing the high performance router. The design principles of ONOFF controller are discussed in detail. The guidelines towards parameter settings are presented. The performance is extensively evaluated and compared with other well-known controllers through simulations and theoretical analysis. The results demonstrate that the ONOFF controller is responsive and robust against external disturbances, and is insensitive to variances of the system parameters. Therefore, it is very suitable for the time- varying network system, and at the same time, it can also keep the instantaneous queue length at a desired level with rather small oscillations, which is conducive to achieving the technical objectives of AQM.     

Design and Analysis of a Multiscale Active Queue Management Scheme

Since Internet is dominated by TCP-based applications, active queue management （AQM） is considered as an effective way for congestion control. However, most AQM schemes suffer obvious performance degradation with dynamic traffic. Extensive measurements found that Internet traffic is extremely bursty and possibly self-similar. We propose in this paper a new AQM scheme called multiscale controller （MSC） based on the understanding of traffic burstiness in multiple time scale. Different from most of other AQM schemes, MSC combines rate-based and queue-based control in two time scales. While the rate-based dropping on burst level （large time scales） determines the packet drop aggressiveness and is responsible for low and stable queuing delay, good robustness and responsiveness, the queue-based modulation of the packet drop probability on packet level （small time scales） will bring low loss and high throughput. Stability analysis is performed based on a fluid-flow model of the TCP/MSC congestion control system and simulation results show that MSC outperforms many of the current AQM schemes.     

Adaptive playout scheduling and loss concealment for voice communication over IP networks

The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score.     

Priority and delay aware packet management framework for real-time video transport over 802.11e WLANs

The rigid delay constraint is one of the most challenging issues in real-time video delivery over wireless networks. The expired video packets will become useless for the decoding and display even if they are received correctly at the receiver. Because the significance of each video packet is different, the schedulers have to take into account not only the urgency of the packet but also its importance in the real-time video applications. However, the existing QoS-based IEEE 802.11e MAC protocol leaves the urgency and the importance of video packets out of consideration. This paper proposes a Priority and Delay Aware Packet Management Framework (PDA-PMF) to improve the transmission quality of real-time video streaming over IEEE 802.11e WLANs. In the MAC layer, this framework estimates the delay of each video packet. Subsequently, video packets are sent or dropped according to both the significance of the video packets and the estimation value of the delay. Simulation results show that the proposed scheme can not only reduce the packet losses, but also protect the more important video packets, so as to improve the received video quality effectively.     

AQM router design for TCP network via input constrained fuzzy control of time-delay affine Takagi–Sugeno fuzzy models

In this article, Takagi–Sugeno (T–S) fuzzy control theory is proposed as a key tool to design an effective active queue management (AQM) router for the transmission control protocol (TCP) networks. The probability control of packet marking in the TCP networks is characterised by an input constrained control problem in this article. By modelling the TCP network into a time-delay affine T–S fuzzy model, an input constrained fuzzy control methodology is developed in this article to serve the AQM router design. The proposed fuzzy control approach, which is developed based on the parallel distributed compensation technique, can provide smaller probability of dropping packets than previous AQM design schemes. Lastly, a numerical simulation is provided to illustrate the usefulness and effectiveness of the proposed design approach.     

Design and analysis of a self-tuning feedback controller for the Internet

Active queue management (AQM) is an effective method used in Internet routers for congestion avoidance, and to achieve a tradeoff between link utilization and delay. The de facto standard, the random early detection (RED) AQM scheme, and most of its variants use average queue length as a congestion indicator to trigger packet dropping. This paper proposes a novel packet dropping scheme, called self-tuning proportional and integral RED (SPI-RED), as an extension of RED. SPI-RED is based on a self-tuning proportional and Integral feedback controller, which considers not only the average queue length at the current time point, but also the past queue lengths during a round-trip time to smooth the impact caused by short-lived traffic dynamics. Furthermore, we give theoretical analysis of the system stability and give guidelines for selection of feedback gains for the TCP/RED system to stabilize the average queue length at a desirable level. The proposed method can also be applied to the other variants of RED. Extensive simulations have been conducted with ns2. The simulation results have demonstrated that the proposed SPI-RED algorithm outperforms the existing AQM schemes in terms of drop probability and stability.     

Enhanced bulk scheduling for supporting delay sensitive streaming applications

Providing end-to-end delay guarantees for delay sensitive applications is an important packet scheduling issue with routers. In this paper, to support end-to-end delay requirements, we propose a novel network scheduling scheme, called the bulk scheduling scheme (BSS), which is built on top of existing schedulers of intermediate nodes without modifying transmission protocols on either the sender or receiver sides. By inserting special control packets, which called TED (Traffic Specification with End-to-end Deadline) packets, into packet flows at the ingress router periodically, the BSS schedulers of the intermediate nodes can dynamically allocate the necessary bandwidth to each flow to enforce the end-to-end delay, according to the information in the TED packets. The introduction of TED packets incurs less overhead than the per-packet marking approaches. Three flow bandwidth estimation methods are presented, and their performance properties are analyzed. BSS also provides a dropping policy for discarding late packets and a feedback mechanism for discovering and resolving bottlenecks. The simulation results show that BSS performs efficiently as expected.     

A More Practical Approach for Single-Packet IP Traceback using Packet Logging and Marking

Tracing IP packets to their origins is an important step in defending Internet against denial-of-service attacks. Two kinds of IP traceback techniques have been proposed as packet marking and packet logging. In packet marking, routers probabilistically write their identification information into forwarded packets. This approach incurs little overhead but requires large flow of packets to collect the complete path information. In packet logging, routers record digests of the forwarded packets. This approach makes it possible to trace a single packet and is considered more powerful. At routers forwarding large volume of traffic, the high storage overhead and access time requirement for recording packet digests introduce practicality problems. In this paper, we present a novel scheme to improve the practicality of log-based IP traceback by reducing its overhead on routers. Our approach makes an intelligent use of packet marking to improve scalability of log-based IP traceback. We use mathematical analysis and simulations to evaluate our approach. Our evaluation results show that, compared to the state-of-the-art log-based approach called hash-based IP traceback, our approach maintains the ability to trace single IP packet while reducing the storage overhead by half and the access time overhead by a factor of the number of neighboring routers.     

Enhanced frameless slotted ALOHA protocol with Markov chains analysis

In this paper, we propose a novel approach to enhance the performance of frameless slotted ALOHA (SA) protocol. We employ signature codes to help the receiver identify the packets contained in collisions, and use successive interference cancellation (SIC) for packet recovery. We model the proposed scheme as a two-state Markov model represented by a uni-partite graph. We evaluate the throughput, expected delay and average memory size of the proposed scheme, and optimize the proposed scheme to maximize the throughput. We show that the theoretical analysis matches well with simulation results. The throughput and expected delay of the proposed protocol outperform the conventional slotted ALOHA protocol significantly.     

Real-time video streaming over multipath in multi-hop wireless networks

Given the limited wireless link throughput, high loss rate, and varying end-to-end delay, supporting video applications in multi-hop wireless networks becomes a challenging task. Path diversity exploits multiple routes for each session simultaneously, which achieves higher aggregated bandwidth and potentially decreases delay and packet loss. Unfortunately, for TCP-based video streaming, naive load splitting often results in inaccurate estimation of round trip time (RTT) and packet reordering. As a result, it can suffer from significant instability or even throughput reduction, which is also validated by our analysis and simulation in multi-hop wireless networks. To make real-time TCP-based streaming viable over multi-hop wireless networks, we propose a novel cross-layer design with a smart traffic split scheme, namely, multiple path retransmission (MPR). MPR differentiates the original data packets and the retransmitted packets and works with a novel QoS-aware multi-path routing protocol, QAOMDV, to distribute them separately. MPR does not suffer from the RTT underestimation and extra packet reordering, which ensures stable throughput improvement over single-path routing. Through extensive simulations, we further demonstrate that, as compared with state-of-the-art multi-path protocols, our MPR with QAOMDV noticeably enhances the TCP streaming throughput and reduces bandwidth fluctuation, with no obvious impact to fairness.     

地铁环境中双网卡软切换机制的研究与应用

高速移动的无线节点在接入点间切换时,切换延迟较大、丢包率较高.在单网卡切换环境中,不可避免地存在网络中断的现象.该文结合地铁无线通信环境对切换延迟和丢包问题进行研究,为移动节点配备两块无线网卡,控制两块网卡协同工作,共同完成无线切换和数据传输.测试结果表明,双网卡软切换机制无需修改网络层及上层协议栈,在特定环境中可以实现低延迟和零丢包.该切换机制已经在地铁信号系统国产化预研项目中得到应用.     

An underwater acoustic MAC protocol using reverse opportunistic packet appending

Underwater communication primarily utilizes propagation of acoustic waves in water. Its unique characteristics, including slow propagation speed and low data rates, pose many challenges to Media Access Control (MAC) protocol design. In most existing handshaking-based underwater MAC protocols, only an initiating sender can transmit data packets to its intended receiver after a channel reservation through a Request-to-Send (RTS)/Clear-to-Send (CTS) handshake. This conventional single-node transmission approach is particularly inefficient in underwater environments, as it does not account for long propagation delays. To improve channel utilization in high latency environments, we propose a novel approach that exploits the idle waiting time during a 2-way handshake to set up concurrent transmissions from multiple nodes. The sender can coordinate multiple first-hop neighbors (appenders) to use the current handshake opportunity to transmit (append) their data packets with partially overlapping transmission times. After the sender finishes transmitting its packets to its own receiver, it starts to receive incoming appended packets that arrive in a collision-free packet train. This not only reduces the amount of time spent on control signaling, but it also greatly improves packet exchange efficiency. Based on this idea, we propose an asynchronous, single-channel handshaking-based MAC protocol based on reverse opportunistic packet appending (ROPA). From extensive simulations (single- and multi-hop networks) and comparisons with several existing MAC protocols, including MACA-U, MACA-UPT, BiC-MAC, Slotted-FAMA, DACAP, unslotted Aloha, we show that ROPA significantly increases channel utilization and offers performance gains in throughput and delay while attaining a stable saturation throughput.     

Differentiated Bandwidth Allocation with TCP Protection in Core Routers

Differentiated Services (DiffServ) networks categorize routers into edge routers and core routers. In core routers, one of the technological challenges is how to implement differentiated bandwidth allocation and TCP protection together with low complexity. We present an Active Queue Management (AQM) scheme called CHOKeW. A method is borrowed from a previous scheme, CHOKe, which draws a packet at random from the buffer, compares it with the arriving packet, and drops both if they are from the same flow. CHOKeW enhances the drawing function by adjusting the maximum number of draws based on the priority of the new arrival and the current status of network congestion. With respect to the number of flows, both the memory-requirement complexity and the per-packet-processing complexity for CHOKeW is O(1). An analytical model and multiple simulations are used to explain and evaluate CHOKeW. We show that CHOKeW is able to 1) support differentiated bandwidth allocation; 2) provide the flows in the same priority with better fairness than other conventional stateless AQM schemes such as RED and BLUE; 3) maintain high link utilization as well as short queue length; and 4) protect TCP flows by restricting the bandwidth share of high-speed unresponsive flows.