Transmission Control Protocol Retransmission Research Articles

The analytical framework for optimizing TCP retransmission algorithm based on adjustment of its EWMA parameters

In this paper we derive mathematical description of TCP (Transmission Control Protocol) retransmission probability based on Jacobson’s smoothing algorithm that belongs to EWMA (Exponentially Weighted Moving Average) category. This description is parametrized on the probability density function (pdf) of RTT (Round Trip Time) samples and α, β – two primary parameters of Jacobson’s algorithm. Although it is not a close form expression, it is formulated as an effective algorithm that let us to explicitly calculate the values of RTO (Retransmission Time Out) probability as a function of α, β and the pdf of RTT samples. We achieve the effectiveness of this approach by applying smart discretization of the state space and replacement of continuous functions with discrete approximate equivalents. In this way, we mitigate the cardinality of discrete distributions we deal with that results in linear (n+m) instead of multiplicative (n⋅m) growth of computational complexity. We provide the evaluation of RTO probability for a wide set of α, β parameter values and differently shaped Normal and Laplace pdfs the RTT samples are taken from. The obtained numerical results let us to draw some conclusions regarding the choice of optimal values of α, β parameters as well as the impact of pdf the RTT samples are taken from.

TCP NRT: a new TCP algorithm for differentiating non-congestion retransmission timeouts over multihop wireless networks

In multihop wireless networks, reliable data transfer is one of the most difficult tasks. When transmission control protocol (TCP) operates in multihop wireless networks, the performance of TCP reduces drastically. TCP retransmission timeouts (RTOs) related to non-congestion events such as spurious and random packet losses have been reported as one of the main problems in the performance degradation of TCP in multihop wireless networks. The RTOs triggered by random packet losses due to transmission errors lead to unnecessary reduction of TCP congestion window size, and the spurious RTOs due to sudden delay of packets on the network paths often cause unnecessary retransmissions as well as reduction of congestion window size. Existing solutions for detecting non-congestion RTOs have no mechanism to differentiate the spurious RTOs from RTOs caused by random packet loss. In this paper, we introduce an efficient algorithm called non-congestion retransmission timeouts (TCP NRT) which is capable of recovering packets after RTOs by reducing unnecessary retransmissions and needless reduction of congestion window size in order to improve the performance of TCP in multihop wireless networks. TCP NRT consists of three key components: NRT-detection, NRT-differentiation, and NRT-reaction. We implemented the algorithm in Qualnet network simulator and compared its performance to existing TCP versions. Results from the experiments show that our algorithm achieves significant performance improvement in terms of throughput and accuracy. Also, the results showed that our algorithm, TCP NRT, maintains a fair and friendly behavior compared to the most widely deployed TCP, NewReno.

Network Coding Meets TCP: Theory and Implementation

The theory of network coding promises significant benefits in network performance, especially in lossy networks and in multicast and multipath scenarios. To realize these benefits in practice, we need to understand how coding across packets interacts with the acknowledgment (ACK)-based flow control mechanism that forms a central part of today's Internet protocols such as transmission control protocol (TCP). Current approaches such as rateless codes and batch-based coding are not compatible with TCP's retransmission and sliding-window mechanisms. In this paper, we propose a new mechanism called TCP/NC that incorporates network coding into TCP with only minor changes to the protocol stack, thereby allowing incremental deployment. In our scheme, the source transmits random linear combinations of packets currently in the congestion window. At the heart of our scheme is a new interpretation of ACKs-the sink acknowledges every degree of freedom (i.e., a linear combination that reveals one unit of new information) even if it does not reveal an original packet immediately. Thus, our new TCP ACK rule takes into account the network coding operations in the lower layer and enables a TCP-compatible sliding-window approach to network coding. Coding essentially masks losses from the congestion control algorithm and allows TCP/NC to react smoothly to losses, resulting in a novel and effective approach for congestion control over lossy networks such as wireless networks. An important feature of our solution is that it allows intermediate nodes to perform re-encoding of packets, which is known to provide significant throughput gains in lossy networks and multicast scenarios. Simulations show that our scheme, with or without re-encoding inside the network, achieves much higher throughput compared to TCP over lossy wireless links. We present a real-world implementation of this protocol that addresses the practical aspects of incorporating network coding and decoding with TCP's window management mechanism. We work with TCP-Reno, which is a widespread and practical variant of TCP. Our implementation significantly advances the goal of designing a deployable, general, TCP-compatible protocol that provides the benefits of network coding.

Engineering TCP transmission and retransmission mechanisms for wireless networks

Transmission Control Protocol (TCP) provides mechanisms for reliable data communications. Although it works well in wired networks, it fails to offer satisfactory performance in lossy and wireless environments. And in the multi-hop wireless infrastructure, packet delivery suffers high cumulative loss rate if traveling over multiple wireless hops. The Selective acknowledgment (SACK) is one header option that might be used to combat segment corruptions in air channels. In this paper, an alternative set of flow control mechanisms is proposed to handle high packet loss rate in a wireless medium. Using a measurement-based mechanism, sustainable segment delivery is achievable through a novel size-reduction method. Multiple segment retransmission mechanisms are introduced to reduce successive timeout events. One single byte loss is sufficient to waste all other bytes in a file received at a destination. That is, a good TCP flow control mechanism should provide a high successful file transmission completion rate, and this is set as our design goal. Through thorough simulations, our proposed multi-segment retransmission designs perform with higher successful file transfer rate and fewer timeout events than NewReno and SACK under a wide range of packet loss probabilities.

A Comparative Study of Aggregate TCP Retransmission Rates

Segment retransmissions are an essential tool in assuring reliable end-to-end communication in the Internet. Their crucial role in Transmission Control Protocol (TCP) design and operation have been studied extensively, in particular with respect to identifying non-conformant, buggy, or underperforming behaviour. However, TCP segment retransmissions are often overlooked when examining and analysing large traffic traces. In fact, some have come to believe that retransmissions are a rare oddity, characteristically associated with faulty network paths, which, typically, tend to disappear as networking technology advances and link capacities grow. We find that this may be far from the reality experienced by TCP flows. We quantify aggregate TCP segment retransmission rates using publicly available network traces from six passive monitoring points attached to the egress gateways at large sites. In virtually half of the traces examined we observed aggregate TCP retransmission rates exceeding 1%, and of these, about half again had retransmission rates exceeding 2%. Even for sites with low utilization and high-capacity gateway links, retransmission rates of 1%, and sometimes higher, were not uncommon. Our results complement, extend, and bring up to date partial and incomplete results in previous work, and show that TCP retransmissions continue to constitute a non-negligible percentage of the overall traffic, despite significant advances across the board in telecommunications technologies and network protocols. The results presented are pertinent to end-to-end protocol designers and evaluators as they provide a range of “realistic” scenarios under which, and a “marker” against which, simulation studies can be configured and calibrated, and future protocols evaluated.

Retransmission steganography and its detection

The paper presents a new steganographic method called RSTEG (retransmission steganography), which is intended for a broad class of protocols that utilises retransmission mechanisms. The main innovation of RSTEG is to not acknowledge a successfully received packet in order to intentionally invoke retransmission. The retransmitted packet carries a steganogram instead of user data in the payload field. RSTEG is presented in the broad context of network steganography, and the utilisation of RSTEG for TCP (transmission control protocol) retransmission mechanisms is described in detail. Simulation results are also presented with the main aim of measuring and comparing the steganographic bandwidth of the proposed method for different TCP retransmission mechanisms, as well as to determine the influence of RSTEG on the network retransmission level.

Statistical analysis of TCP's retransmission timeout algorithm

The retransmission timeout (RTO) algorithm of Transmission Control Protocol (TCP), which sets a dynamic upper bound on the next round-trip time (RTT) based on past RTTs, plays an important role in reliable data transfer and congestion control of the Internet. A rigorous theoretical analysis of the RTO algorithm is important in that it provides insight into the algorithm and prompts optimal design strategies. Nevertheless, such an analysis has not been conducted to date. This paper presents such an analysis from a statistical approach. We construct an auto-regressive (AR) model for the RTT processes based on experimental results that indicate: 1) RTTs along a certain path in the Internet can be modeled by a shifted Gamma distribution and 2) the temporal correlation of RTTs decreases quickly with lag. This model is used to determine the average reaction time and premature timeout probability for the RTO algorithm. We derive a closed-form expression for the first measure and a formula for numerically calculating the second. Both measures are validated through tests on simulated and real RTT data. The theoretical analysis strengthens a number of observations reported in past experiment-oriented studies.