Transmission Control Protocol Receiver Research Articles

An Adaptive Delayed Acknowledgment Strategy to Improve TCP Performance in Multi-hop Wireless Networks

In multi-hop wireless networks, transmission control protocol (TCP) suffers from performance deterioration due to poor wireless channel characteristics. Earlier studies have shown that the small TCP acknowledgments consume as much wireless resources as the long TCP data packets. Moreover, generating an acknowledgment (ACK) for each incoming data packet reduces the performance of TCP. The main factor affecting TCP performance in multi-hop wireless networks is the contention and collision between ACK and data packets that share the same path. Thus, lowering the number of ACKs using the delayed acknowledgment option defined in IETF RFC 1122 will improve TCP performance. However, large cumulative ACKs will induce packet loss due to retransmission time-out at the sender side of TCP. Motivated by this understanding, we propose a new TCP receiver with an adaptive delayed ACK strategy to improve TCP performance in multi-hop wireless networks. Extensive simulations have been done to prove and evaluate our strategy over different topologies. The simulation results demonstrate that our strategy can improve TCP performance significantly.

Improving Performance of Remote TCP in Cognitive Radio Networks

Recent advances in cognitive radio technology have drawn immense attention to higher layer protocols above medium access control, such as transmission control protocol (TCP). Most proposals to improve the TCP performance in cognitive radio (CR) networks have assumed that either all nodes are in CR networks or the TCP sender side is in CR links. In those proposals, lower layer information such as the CR link status could be easily exploited to adjust the congestion window and improve throughput. In this paper, we consider a TCP network in which the TCP sender is located remotely over the Internet while the TCP receiver is connected by a CR link. This topology is more realistic than the earlier proposals, but the lower layer information cannot be exploited. Under this assumption, we propose an enhanced TCP protocol for CR networks called TCP for cognitive radio (TCP-CR) to improve the existing TCP by (1) detection of primary user (PU) interference by a remote sender without support from lower layers, (2) delayed congestion control (DCC) based on PU detection when the retransmission timeout (RTO) expires, and (3) exploitation of two separate scales of the congestion window adapted for PU activity. Performance evaluation demonstrated that the proposed TCP-CR achieves up to 255% improvement of the end-to-end throughput. Furthermore, we verified that the proposed TCP does not deteriorate the fairness of existing TCP flows and does not cause congestions.

Robust End-to-End Available Bandwidth Estimation forInteractive Satellite networks

This paper presents an efficient and accurate available bandwidth estimation algorithm for satellite links which generally have long propagation delay. Either the method using the rate of returning ACKs or the method using timestamp option is adopted for the bandwidth estimation algorithms to enhance TCP performance in the literature. The method using the rate of returning ACKs, which is the most popular for the available bandwidth estimation for a TCP connection, is known to have the influences from the reverse link (TCP receiver to TCP sender) traffic variation, especially in the asymmetric satellite links. The bandwidth estimation method using packet sending time information from the TCP timestamp option also has the drawback that the influence of the bottleneck link presented in the forward path is not properly considered. To overcome these limits, our algorithm performs in two steps. A forward path delay is calculated at the sender with the receiver sending time for an ACK packet in the first step. And in the second step, the forward path delay is updated more delicately through removing unrelated part from the forward path delay, and is applied for available bandwidth estimation. The simulation results show that our algorithm can avoid the influence of the traffic variation in the return path (e.g., ACK compression, DAMA delay), and estimate the available bandwidth over the forward path more accurately. Satellite networks are becoming the important candidate in information and communication infrastructures because they provide several advantages to the wireless Internet communications. They generally provide wider coverage, higher bandwidth, and relatively faster establishment than other terrestrial wireless networks. Nevertheless, satellite links pose some challenges to the congestion control operation of the TCP (Transmission Control Protocol) (10). Main challenges of the satellite networks to the TCP performance are the long RTT (round-trip time) which is generally around 600msec in the bi-directional networks over the geostationary satellite, and the presence of the high packet loss rate by wireless random error which causes congestion control of the TCP spuriously. To solve the problems while keeping end-to-end semantics of TCP, an accurate bandwidth estimation algorithm is

Cross-Layer-Based Adaptive Vertical Handoff With Predictive RSS in Heterogeneous Wireless Networks

A heterogeneous wireless network consists of various wireless networks [e.g., Worldwide Interoperability for Microwave Access (WiMAX) and Wireless Fidelity (WiFi)] and cellular communications [e.g., beyond the third generation (B3G) and the fourth generation (4G)]. Vertical handoff is an important mechanism for achieving continuous seamless transmissions in these networks. In contrast to horizontal handoff, vertical handoff considers not only the received signal strength (RSS) but also the service-class mapping between handoff-in and handoff-out networks. Most previous works have adopted the RSS-based mechanism to determine handoff thresholds, which causes a serious ping-pong effect that increases unnecessary handoff. Although integrating the RSS-based mechanism with a hysteresis method reduces the unnecessary handoff, it suffers from high dropping [i.e., high Sum of Weighted Grade of Service (SWGoS)] and low utilization. Therefore, this paper proposes a cross-layer-based adaptive vertical handoff algorithm with predictive RSS to reduce the unnecessary handoff while significantly increasing utilization and decreasing connection dropping. The proposed approach determines the optimal target network in two phases, i.e., polynomial regression RSS prediction and Markov decision process analysis. Furthermore, fast changes in bandwidth caused by vertical handoff result in inaccurate Transmission Control Protocol (TCP) congestion control and, thus, reduce the TCP goodput. The cross-layer scheme provides a TCP receiver to reply to the TCP sender with the wireless network's protocol type. By using the cross-layer information, the TCP sender can accurately predict the available bandwidth and increase the network goodput. Numerical results indicate that the proposed cross-layer-based approach outperforms the other approaches in the number of vertical handoffs and SWGoS while yielding competitive utilization. In addition, the cross-layer scheme cooperates with existing TCP algorithms to increase goodput by up to 18%.

Improving TCP performance over last-hop wireless networks for live video delivery

This paper discusses a link layer protocol that enhances performance of the Transmission Control Protocol (TCP) over a last-hop wireless network (LHWN) to make it more suitable for delivery of a live video with a short play out deadline. The presented protocol provides preference control of the transmission of video data within the network and it maximizes network goodput (the successfully transmitted amount of useful video data) by maximizing reliability of transmission for a given playout deadline and a varying digital bandwidth. The protocol makes use of the skip function in the TCP receiver to maximize end-to-end throughput of the connection at the expense of reducing full reliability. It can be implemented in a wireless access point to improve quality of the received live video over an Internet connection to an office or home. The main ideas of the protocol are: hide packet losses from the sender by performing local retransmissions based on the information from the video application and according to a certain scheduling policy, drop stale segments before they arrive at the receiver and prevent end-to-end retransmissions of segments received by the access point. The performance of the TCP Reno protocol in combination with both the presented protocol and the skip function in the receiver is compared against the performance of TCP Reno, the TCP-Real-Time Mode (RTM) protocol, TCP Reno in combination with the Snoop protocol and TCP-RTM in combination with Snoop. The results of simulations show improvement of network goodput across a wide range of packet-error rates.

Active Queue Management for Fair Resource Allocation in Wireless Networks

This paper investigates the interaction between end-to-end flow control and medium access control (MAC)-layer scheduling on wireless links. We consider a wireless network with multiple users receiving information from a common access point; each user suffers fading and a scheduler allocates the channel based on channel quality but is subject to fairness and latency considerations. We show that the fairness property of the scheduler is compromised by the transport-layer flow control of transmission control protocol (TCP) New Reno. We provide a receiver-side control algorithm, CLAMP, that remedies this situation. CLAMP works at a receiver to control a TCP sender by setting the TCP receiver's advertised window limit, and this allows the scheduler to allocate bandwidth fairly between the users.

Enhancing TCP performance to persistent packet reordering

In this paper, we propose a simple algorithm to adaptively adjust the value of dupthresh, the duplicate acknowledgement threshold that triggers the transmission control protocol (TCP) fast retransmission algorithm, to improve the TCP performance in a network environment with persistent packet reordering. Our algorithm uses an exponentially weighted moving average (EWMA) and the mean deviation of the lengths of the reordering events reported by a TCP receiver with the duplicate selective acknowledgement (DSACK) extension to estimate the value of dupthresh. We also apply an adaptive upper bound on dupthresh to avoid the retransmission timeout events. In addition, our algorithm includes a mechanism to exponentially reduce dupthresh when the retransmission timer expires. With these mechanisms, our algorithm is capable of converging to and staying at a near-optimal interval of dupthresh. The simulation results show that our algorithm improves the protocol performance significantly with minimal overheads, achieving a greater throughput and fewer false fast retransmissions.

패킷 손실률에 기반한 효율적인 TCP Buffer Tuning 알고리즘

기존 TCP 기술은 송<TEX>${\cdot}$</TEX>수신측에 각각 고정된 크기의 버퍼를 할당하기 때문에 높은 대역폭(High-Bandwidth) 및 큰 전송지연(High Delay)을 가진 통신에는 적합하지 못하다. 따라서 종단간의 TCP 처리량을 개선하기 위해 통신망 상황에 따라 자동으로 TCP 버퍼를 조절하려는 시도가 있어왔다. ATBT(Automatic TCP Buffer Tuning)에서 송신측은 현재의 혼잡 제어 윈도우(CWND)의 값에 따라 송신 버퍼 크기를 조절하고 수신측은 운영체제가 정해ens 최대 크기의 TCP 버퍼 값으로 수신 버퍼 크기를 고정한다. DRS(Dynamic Right Sizing) 에서는 이전에 수신한 TCP 데이터의 두 배를 현재 송신할 TCP 데이터라고 예측함으써, TCP 수신측은 단순히 이에 따라 수신 버퍼 크기를 동적으로 변화시킨다. 그렇지만 TCP 세그먼트의 손실 가능성으로 인해 정확히 두 배로 버퍼 크기를 변화시킬 필요는 없다. 따라서 우리가 제안한 패킷 손실률에 기반한 효율적인 TCP 버퍼 조절 알고리즘(TBT-PLR:TCP Buffer Tuning Algorithm based on Packet Loss Ratio)은 TCP 송신측에는 ATBT 방법을 TCP 수신측에는 TBT-PLR 방법을 적용하였다. 실제 TCP 성능을 테스트하기 위해서 리눅스 커널 2.4.18을 수정하여 구현하였으며 기존의 고정된 크기의 TCP 버퍼를 가진 경우와 버퍼 크기가 동적으로 변하는 TBT-PLR을 적용한 경우를 비교하였다. 결과적으로, TCP 연결들간의 균형있는 메모리 사용으로 인해 성능 향상을 얻을 수 있었다. Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

A multi-queue TCP window control scheme with dynamic buffer allocation

Abstract Explicit transmission control protocol (TCP) window control through the modification of the receiver’s advertised window in ACK packets is one of the ways intermediate network elements can contribute to the end-to-end TCP control. The TCP receiver’s advertised window, which indicates the level of the receive buffer of a TCP connection, limits the maximum window and consequently the throughput that can be achieved by the TCP sender. Thus, appropriate reduction of the advertised window by intermediate network elements can control the number of packets sent from a TCP sender. This paper describes a TCP window control scheme for a shared memory system with multiple queues. A dynamic buffer threshold, computed using a simple recursive algorithm, is used to dynamically allocate buffer space to the queues.