Transport Control Protocol Research Articles

An improved transport layer protocol for wireless sensor networks

Wireless sensor networks (WSNs) are a kind of communication networks having independent sensor nodes that form multi-hop ad hoc network to transfer data. In the past few years, various transport control protocols in wireless sensor networks (WSNs) have been developed and proposed in the literature. In this paper, we have analyzed pump slowly, fetch quickly (PSFQ) protocol and presented an improved transport layer protocol for wireless sensor networks. The improved protocol has been analyzed based on various factors such as average latency and average error tolerance and it is found that the proposed protocol is better than PSFQ in terms of these factors.

무선 메쉬 네트워크에서 TCP 성능 향상 기법

Wireless mesh network is similar to ad-hoc network, so when transferred to the data packet in the wireless environment, interfered factor arise. When TCP(Transport Control Protocol) was created, however as it was design based on wired link, wireless link made more transmission error than wired link. It is existent problem that TCP unfairness and congestion collapse over wireless mesh network. But packet losses due to transmission errors are more frequent. The cause of transmission error in wireless ad-hoc network may be inexactly regarded as indications of network congestion. And then, Congestion Control Algorithm was running by this situation causes the TCP performance degradation. In this paper, proposed TCP can adaptively regulate the congestion window through moving node in the Wireless Mesh Network. And it enhanced the performance.

TCP and web browsing performance in case of bi-directional packet loss

Performance modeling of the transport control protocol (TCP) has received a lot of attention over the past few years. The most commonly quoted results are approximate formulas for TCP throughput (Padhye et al. (2000) [1]) and document download times (Cardwell et al. (2000) [2]) which are used for dimensioning of IP networks. However, the existing modeling approaches unanimously assume that packet loss only occurs for packets from the server to the client, whereas in reality the packets in the direction from the client to the server may also be dropped. Our simulations with NS-2 show that this bi-directional packet loss indeed may have a strong impact on TCP performance. Motivated by this, we refine the models in Padhye et al. (2000) [1] and Cardwell et al. (2000) [2] by including bi-directional packet loss, also including correlations between packet loss occurrences. Simulations show that the proposed model leads to strong improvements of the accuracy of the TCP performance predictions. In addition we show how our model can be used to predict quality of experience for web browsing sessions.

TCP is Competitive with Resource Augmentation

The well-known Transport Control Protocol (TCP) is a crucial component of the TCP/IP architecture on which the Internet is built, and is a de facto standard for reliable communication on the Internet. At the heart of the TCP protocol is its congestion control algorithm. While most practitioners believe that the TCP congestion control algorithm performs very well, a complete analysis of the congestion control algorithm is yet to be done. A lot of effort has, therefore, gone into the evaluation of different performance metrics like throughput and average latency under TCP. In this paper, we approach the problem from a different perspective and use the competitive analysis framework to provide some answers to the question “how good is the TCP/IP congestion control algorithm?” We describe how the TCP congestion control algorithm can be viewed as an online, distributed scheduling algorithm. We observe that existing lower bounds for non-clairvoyant scheduling algorithms imply that no online, distributed, non-clairvoyant algorithm can be competitive with an optimal offline algorithm if both algorithms were given the same resources. Therefore, in order to evaluate TCP using competitive analysis, we must limit the power of the adversary, or equivalently, allow TCP to have extra resources compared to an optimal, offline algorithm for the same problem. In this paper, we show that TCP is competitive to an optimal, offline algorithm provided the former is given more resources. Specifically, we prove first that for networks with a single bottleneck (or point of congestion), TCP is ${\mathcal{O}}(1)$-competitive to an optimal centralized (global) algorithm in minimizing the user-perceived latency or flow time of the sessions, provided we allow TCP ${\mathcal{O}}(1)$times as much bandwidth and ${\mathcal{O}}(1)$extra time per session. Second, we show that TCP is fair by proving that the bandwidths allocated to sessions quickly converge to fair sharing of network bandwidth.

Rate Control Algorithm Considering Mobility in Mobile Ad Hoc Networks

For reducing the loss of data packet due to network congestion and mobility, rate control algorithm is effective means for ensuring the fair allocation of network resource among communication flows and alleviating network congestion. The Transport Control Protocol (TCP) was originally designed for wire-line networks, where the links are assumed to be reliable and with fixed capacities. However, the performance of TCP becomes very poor when it is directly used in ad hoc networks. Some optimization-based congestion control schemes have been proposed to improve TCP performance in ad hoc networks, but the mobility of hosts and route change frequently are not considered for designing rate control algorithm. In this paper, we propose rate control algorithms in a more practical way by taking into account link congestion and node movement. Numerical results show that the rate control algorithms can approach the globally optimal value and show the effect of the node mobility on the algorithms.

DETRENDED FLUCTUATION ANALYSIS OF THE TCP-RED ALGORITHM

It has been observed that Internet gateways employing Transport Control Protocol (TCP) and the Random Early Detection (RED) control algorithm may exhibit instability and oscillatory behavior. Most control methods proposed in the past have been based on analytical models that rely on statistical measurements of network parameters. In this paper, we apply the detrended fluctuation analysis (DFA) method to analyze stability of the TCP-RED system. The DFA is used to analyze time-series data and generate power-law scaling exponents, which indicate the long-range correlations of the time series. We quantify the stability of the TCP-RED system by examining the variation of the DFA power-law scaling exponent when the system parameters are varied. We also study the long-range power-law correlations of TCP window periods.

Cross-layer speculative architecture for end systems and gateways in computer networks with lossy links

The throughput degradation of Transport Control Protocol (TCP)/Internet Protocol (IP) networks over lossy links due to the coexistence of congestion losses and link corruption losses is very similar to the degradation of processor performance (i.e., cycle per instruction) due to control hazards in computer design. First, two types of loss events in networks with lossy links are analogous to two possibilities of a branching result in computers (taken vs. not taken). Secondly, both problems result in performance degradations in their applications, i.e., penalties (in clock cycles) in a processor, and throughput degradation (in bits per second) in a TCP/IP network. This has motivated us to apply speculative techniques (i.e., speculating on the outcome of branch predictions), used to overcome control dependencies in a processor, for throughput improvements when lossy links are involved in TCP/IP connections. The objective of this paper is to propose a cross-layer network architecture to improve the network throughput over lossy links. The system consists of protocol-level speculation based algorithms at transport layer, and protocol enhancements at middleware and network layers that provide control and performance parameters to transport layer functions. Simulation results show that, compared with prior research, our proposed system is effective in improving network throughput over lossy links, capable of handling incorrect speculations, fair for other competing flows, backward compatible with legacy networks, and relatively easy to implement.

Analytical modeling of TCP dynamics in infrastructure-based IEEE 802.11 WLANs

IEEE 802.11 wireless local area network (WLAN) has become the prevailing solution for wireless Internet access while transport control protocol (TCP) is the dominant transport-layer protocol in the Internet. It is known that, in an infrastructure-based WLAN with multiple stations carrying long-lived TCP flows, the number of TCP stations that are actively contending to access the wireless channel remains very small. Hence, the aggregate TCP throughput is basically independent of the total number of TCP stations. This phenomenon is due to the closed-loop nature of TCP flow control and the bottleneck downlink (i.e., access point-to-station) transmissions in infrastructure-based WLANs. In this paper, we develop a comprehensive analytical model to study TCP dynamics in infrastructure-based 802.11 WLANs. We calculate the average number of active TCP stations and the aggregate TCP throughput using our model for given total number of TCP stations and the maximum TCP receive window size. We find out that the default minimum contention window sizes specified in the standards (i.e., 31 and 15 for 802.11b and 802.11a, respectively) are not optimal in terms of TCP throughput maximization. Via ns-2 simulation, we verify the correctness of our analytical model and study the effects of some of the simplifying assumptions employed in the model. Simulation results show that our model is reasonably accurate, particularly when the wireline delay is small and/or the packet loss rate is low.

Superposition-Coding-Aided Multiplexed Hybrid ARQ Scheme for Improved End-to-End Transmission Efficiency

In this paper, we propose a novel superposition-coding-aided multiplexed hybrid automatic repeat request (HARQ) scheme for the sake of improving both the link layer and the transport control protocol (TCP) layer efficiency. The detailed system design is presented, and the transmission efficiency metrics of both layers are discussed. Furthermore, the achievable link-layer packet error rate and a range of other transmission efficiency metrics of both layers are quantified. It is demonstrated that our scheme substantially improves the attainable transmission efficiency of both layers and is particularly suitable for delay-sensitive services.

Analyzing Delivery, Latency and Power Consumption

This work presents Collaborative Transport Control Protocol (CTCP), a new transport protocol for sensor networks. Itaims at providing end-to-end reliability and adapts itself to different applications through a two level mechanism ofreliability variation. CTCP achieves these properties using hop-by-hop acknowledgments and a storage controlalgorithm that operates at each node along a flow. It was observed that distributed fault recovery increases theaverage delivery rate and that duplication of storage responsibility minimizes the message loss. Its congestion controldifferentiates communication losses from buffer overflow. CTCP is called collaborative because all nodes detect andact on congestion control and also because it includes distributed storage responsibility. It is scalable andindependent of the underlying network layer. The protocol energy consumption overhead was calculated anddiscussed for two reliability levels

Error Resistant Real-Time Transport Control Protocol

Problem statement: Real-time Transport Control Protocol (RTCP) protocol has been the subject of various criticisms due its problematic performance in large-scale networks. S-RTCP is a protocol with high potential as it has proved to be able to solve many problems of RTCP. It has numerous flaws on its own. This study aimed at dealing with flaws of S-RTCP and improving it in terms of stability and packet loss. Approach: A new proposed scheme was designed. Modifications included designing multi-manager scheme, improving parent-seeking procedures, reducing distribution of request packets, reforming the design to be independent from TTL, adding methods to check on sanity of manager nodes. This study considered packet loss ratio of below 2% as desirable. Results: ER-RTCP comparing to legacy RTCP in terms of packet loss using NS-2 in four different scenarios revealed improvements between 73 and 88% for various scenarios. It also kept packet loss rate below 2% for all scenarios. Comparison of ER-RTCP to S-RTCP showed that based on different α (stability of each single manager) values, ER-RTCP was more stable as it showed more resistance to entire scheme breakdown (β). ER-RTCP’s parent-seeking procedure, as modeled scenario revealed a packet generation reduction of 97%, compared to S-RTCP’s. In occurrence of parent AG leave or loss, ER-RTCP reduced request packet generation by 95%. Allowance of AG dismissing in ER-RTCP, avoided occurrence of packet loss, as sample scenario showed S-RTCP experiencing packet loss of 3.5% while ER-RTCP kept packet loss at zero in theory. Conclusion: Proposed design improved S-RTCP in terms of reduction of packet loss and stability.

TCP Window-Based Flow-Oriented Dynamic Assembly Algorithm for OBS Networks

In transport control protocol (TCP) over optical burst switching (OBS) networks, TCP window size and OBS parameters, including assembly period and burst dropping probability, will impact the network performance. In this paper, a parameter window data dropping probability(WDDP), is defined to analyze the impact of the assembly and the burst loss on the network performance in terms of the round trip time and the throughput. To reduce the WDDP without introducing the extra assembly delay penalty, we propose a novel TCP window based flow-oriented assembly algorithm dynamic assembly period (DAP). In the traditional OBS assembly algorithms, the packets with the same destination and class of service (CoS) are assembled into the same burst, i.e., the packets from different sources will be assembled into one burst. In that case, one burst loss will influence multiple TCP sources. In DAP, the packets from one TCP connection are assembled into bursts, which can avoid the above situation. Through comparing the two consecutive burst lengths, DAP can track the variation of TCP window dynamically and update the assembly period for the next assembly. In addition, the ingress node architecture for the flow-oriented assembly is designed. The performance of DAP is evaluated and compared with that of fixed assembly period (FAP) over a single TCP connection and multiple TCP connections. The results show that DAP performs better than FAP at almost the whole range of burst dropping probability.

Pathpair: a fast available bandwidth estimation tool with the asymptotic one way delay comparison model

Significant research has been dedicated to active probing methods used to estimate the end-to-end available bandwidth (avail-bw) of a network path. The measurement speed of an avail-bw estimation tool is vital to real-time network services such as transport control protocol (TCP) slow start, path selection for P2P applications and video streaming services and so on. In addition, a fast avail-bw estimation tool sends less probing traffic to a measured network. Pathload, a one way delay (OWD) based tool, has been widely regarded as one of the most accurate tools, but its estimation speed is often 10 or more seconds per measurement. A fast OWD-based avail-bw estimation scheme has been explored. This scheme is based on the proposed asymptotic OWD comparison model, and an estimation tool called Pathpair has been developed. Validated in a real Internet environment, experimental results show that with Pathpair the measurement speed has been significantly improved (1–3 s per measurement), while preserving accuracy compared with other major tools like Pathload and Spruce.

Adaptive Video Streaming Using Bandwidth Estimation for 3.5G Mobile Network

Currently deployed mobile networks including High Speed Downlink Packet Access (HSDPA) offer only best-effort Quality of Service (QoS). In wireless best effort networks, the bandwidth variation is a critical problem, especially, for mobile devices with small buffers. This is because the bandwidth variation leads to packet losses caused by buffer overflow as well as picture freezing due to high transmission delay or buffer underflow. In this paper, in order to provide seamless video streaming over HSDPA, we propose an efficient real-time video streaming method that consists of the available bandwidth (AB) estimation for the HSDPA network and the transmission rate control to prevent buffer over-flows/underflows. In the proposed method, the client estimates the AB and the estimated AB is fed back to the server through real-time transport control protocol (RTCP) packets. Then, the server adaptively adjusts the transmission rate according to the estimated AB and the buffer state obtained from the RTCP feedback information. Experimental results show that the proposed method achieves seamless video streaming over the HSDPA network providing higher video quality and lower transmission delay.

Dynamic load balancing through real-time transport feedback control for VoIP QoS

The Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP) can be utilised for dynamic load balancing over routing diversity. We implemented and tested an RTP-VoIP based application between two internet nodes located in Egypt and the USA. The collected real-time statistics were then used to emulate multiple paths between the source and destination nodes. In this study, the source node utilises RTCP to distribute the voice load between the two paths used for routing diversity. The results show that routing diversity along with load-balancing provide QoS that is much higher than if a single bad path is used and close to what is provided by a path with less traffic impairment.

Analytical Study of TCP Performance over IEEE 802.11e WLANs

IEEE 802.11 Wireless LAN (WLAN) has become a prevailing solution for broadband wireless Internet access while the Transport Control Protocol (TCP) is the dominant transport-layer protocol in the In...

Loss classification in optical burst switching networks using machine learning techniques: improving the performance of TCP

Optical burst switching (OBS) is considered as a contending technology for the core of the Internet in future. However, due to lack of the buffers, losses occur due to contention among simultaneously arriving bursts at the core nodes. Contention losses do not necessarily indicate a situation of congestion in the network. Thus differentiation (classification) of losses is essential in many applications to avoid false identification of congestion. In this paper, we propose a loss classification technique for the OBS networks based on machine learning techniques. We devise a new measure to differentiate between congestion and contention losses, which is derived from the observed losses, called the number of bursts between failures (NBBF). We observe that the NBBF follows a Gaussian distribution with different parameters for contention and congestion losses. This feature is used in differentiation. We use both a supervised learning technique (hidden Markov model (HMM)) and an unsupervised learning technique (expectation maximization (EM) clustering) on the observed losses and classify them into a set of states (clusters) after which an algorithm differentiates between the congestion and contention losses. We also demonstrate the use of loss differentiation in improving the performance of transport control protocol (TCP) over OBS networks. We modify congestion control mechanism of TCP suitably to arrive at two variants of TCP, HMM-TCP and EM-TCP. Their performance is compared with TCP NewReno, TCP SACK, and Burst TCP (X. Yu et al., Mar. 2004). Simulation results demonstrate the effectiveness and accuracy of the loss classification technique in different network scenarios.

Robust end-to-end loss differentiation scheme for transport control protocol over wired/wireless networks

The authors propose a robust end-to-end loss differentiation scheme to identify the packet losses because of congestion for transport control protocol (TCP) connections over wired/wireless networks. The authors use the measured round trip time (RTT) values to determine whether the cause of packet loss is because of the congestion over wired path or regular bit errors over wireless paths. The classification should be as accurate as possible to achieve high throughput and maximum fairness for the TCP connections sharing the wired/wireless paths. The accuracies of previous schemes in the literature depends on varying network parameters such as RTT, buffer size, amount of cross traffic, wireless loss rate and congestion loss rate. The proposed scheme is robust in that the accuracy remains rather stable under varying network parameters. The basic idea behind the scheme is to set the threshold for the classification to be a function of the minimum RTT and the current sample RTT, so that it may automatically adapt itself to the current congestion level. When the congestion level of the path is estimated to be low, the threshold for a packet loss to be classified as a congestion loss is increased. This avoids unnecessary halving of the congestion window on packet loss because of the regular bit errors over the wireless path and hence improves the TCP throughput. When the congestion level of the path is estimated to be high, the threshold for a packet loss to be classified as the congestion loss not to miss any congestion loss is decreased and hence improves the TCP fairness. In ns 2 simulations, the proposed scheme correctly classifies the congestion losses under varying network parameters whereas the previous schemes show some dependency on subsets of parameters.

Transport control protocol in optical burst switched networks: issues, solutions, and challenges

Since its advent in 1981, TCP has been subject to a tremendous amount of research effort and enhancements for achieving better performance over various network environments and application scenarios. Due to the transmission characteristics of optical burst switched networks, such as random burst dropping, retro-blocking (i.e., bursts proceeding or delayed from their actual reservation time slot), burstification delay, and burst signaling delay, TCP could be significantly affected if no corresponding countermeasure and enhancement are developed. In this review article we provide a comprehensive survey on reported studies for TCP enhancements over OBS networks in order to mitigate the numerous side effects due to the buffer- less characteristic of burst transmission. Furthermore, we closely analyze TCP behavior over OBS networks with various burst transmission characteristics while highlighting the open challenges that have not yet been extensively tackled or solved.

Mobile-Host-Centric Transport Protocol for<newline/>EAST Experiment

Some physics researchers retrieve EAST experiment data using the Transmission Control Protocol (TCP) in a wireless local area network (WLAN). TCP is the most commonly used transport control protocol. It assumes that every packet loss is caused by network congestion and invokes congestion control and avoidance. TCP's blind congestion control results in degraded performance in lossy wireless networks. In a wireless network, mobile hosts have first-hand knowledge of the lossy wireless links; therefore, mobile stations can make better transmission control based on the known status of wireless link. In this paper, we proposed a new mobile-host-centric transport protocol (MCP) that integrates the characteristics of sender-centric and receiver-centric transport control schemes. MCP shifts most control policies to the mobile host side. The general behavior of MCP is similar to the TCP, but by utilizing the local information collecting from the mobile node in WLAN, MCP allows for better congestion control and loss recovery. Specifically, we designed a cross-layer implementation of MCP and ran it on NS-2 network simulator. With valuable Medium Access Control (MAC) layer feedback information, MCP is able to distinguish packet loss caused by wireless random errors from network congestion more clearly and can perform a more accurate congestion control. We did extensive simulations of MCP on various WLAN scenarios, and the results show that the throughput of MCP with cross-layer feedback is higher than that of TCP Reno and Westwood.