QoS evaluation of different TCPs congestion control algorithm using NS2

The success of the current Internet relies to a large extent on cooperation between the users and network. The network signals its current state to the users by marking or dropping packets. The user then strives to maximize the sending rate without causing network congestion. To achieve this, the users implement a flow control algorithm that controls the rate at which data packets are sent into the Internet. More specifically, the Transmission Control Protocol (TCP) is used by the users to adjust the sending rate in response to changing network conditions. In this paper, we focus on the degree of fairness provided to TCP connections by comparing two packet-scheduling algorithms at the router. The first one is FIFO (First In First Out, or Drop-Tail), which is widely used in the current Internet routers because of its simplicity. The second is RED (Random Early Detection), which drops incoming packets at a certain probability

Differential Radio Link Protocol: An Improvement To Tcp Over Wireless Networks

New generations of wireless cellular networks, including 3G and 4G technologies, are envisaged to support more mobile users and a variety of wireless multimedia services. With an increasing demand for wireless multimedia services, the performance of TCP becomes a bottleneck as it cannot differentiate between the losses due to the nature of air as a medium and high data load on the network that leads to congestion. This misinterpretation by TCP leads to a reduction in the congestion window size thereby resulting in reduced throughput of the system. To overcome this scenario Radio Link Protocols are used at a lower layer which hides from TCP the channel related losses and effectively increases the throughput. This thesis proposes enhancements to the radio link protocol that works underneath TCP by identifying decisive frames and categorizing them as {\em crucial} and {\em non-crucial}. The fact that initial frames from the same upper layer segment can afford a few trials of retransmissions and the later frames cannot, motivates this work. The frames are treated differentially with respect to FEC coding and ARQ schemes. Specific cases of FEC and ARQ strategies are then considered and it is shown qualitatively as how the differential treatment of frames can improve the performance of the RLP and in effect that of TCP over wireless networks

STCP: A New Transport Protocol for High-Speed Networks

Transmission Control Protocol (TCP) is the dominant transport protocol today and likely to be adopted in future high‐speed and optical networks. A number of literature works have been done to modify or tune the Additive Increase Multiplicative Decrease (AIMD) principle in TCP to enhance the network performance. In this work, to efficiently take advantage of the available high bandwidth from the high‐speed and optical infrastructures, we propose a Stratified TCP (STCP) employing parallel virtual transmission layers in high‐speed networks. In this technique, the AIMD principle of TCP is modified to make more aggressive and efficient probing of the available link bandwidth, which in turn increases the performance. Simulation results show that STCP offers a considerable improvement in performance when compared with other TCP variants such as the conventional TCP protocol and Layered TCP (LTCP)

Quality of Service Control in Wireless Local Area Networks

Today wireless internet access, especially the IEEE 802.11 family WLANs, has become widely available. Public hotspots emerge everywhere, from hotels to airports. In addition, private usage also grows rapidly due to the offered convenience. The current WLAN standard 802.11b does not offer service differentiation. Therefore the quality of an offered service is often not guaranteed. However, a new standard 802.11e has been proposed to improve the quality of service provisioning. This new standard introduces different Traffic Classes for different services and can prioritize the traffic accordingly. This report covers both the 802.11b and the 802.11e WLANs. In the first place it provides investigation on file transfer times with TCP over 802.11b. This includes both modelling and dynamic simulations. These models are shown to give accurate results for large files. The second part focuses on 802.11e. The differentiation mechanism as well as its performance with realistic service settings is examined through simulation studies. The results provide some important characteristics of the 802.11e parameters and advices for tuning an 802.11e network

Delay aspects in Internet telephony

In this work, we address the transport of high quality voice over the Internet with a particular concern for delays. Transport of interactive audio over IP networks often suffers from packet loss and variations in the network delay (jitter). Forward Error Correction (FEC) mitigates the impact of packet loss at the expense of an increase of the end-to-end delay and the bit rate requirement of an audio source. Furthermore, adaptive playout buffer algorithms at the receiver compensate for jitter, but again this may come at the expense of additional delay. As a consequence, existing error control and playout adjustment schemes often have end-to-end delays exceeding 150 ms, which significantly impairs the perceived quality, while it would be more important to keep delay low and accept some small loss. We develop a joint playout buffer and FEC adjustment scheme for Internet Telephony that incorporates the impact of end-to-end delay on perceived audio quality. To this end, we take a utility function approach. We represent the perceived audio quality as a function of both the end-to-end delay and the distortion of the voice signal. We develop a joint rate/error/playout delay control algorithm which optimizes this measure of quality and is TCP-Friendly. It uses a channel model for both loss and delay. We validate our approach by simulation and show that (1) our scheme allows a source to increase its utility by avoiding increasing the playout delay when it is not really necessary and (2) it provides better quality than the adjustment schemes for playout and FEC that were previously published. We use this scheme in the framework of non-elevated services which allow applications to select a service class with reduced end-to-end delay at the expense of a higher loss rate. The tradeoff between delay and loss is not straightforward since audio sources may be forced to compensate the additional losses by more FEC and hence more delay. We show that the use of non-elevated services can lead to quality improvements, but that the choice of service depends on network conditions and on the importance that users attach to delay. Based on this observation, we propose an adaptive service choosing algorithm that allows audio sources to choose in real-time the service providing the highest audio quality. In addition, when used over the standard IP best effort service, an audio source should also control its rate in order to react to network congestion and to share the bandwidth in a fair way. Current congestion control mechanisms are based on packets (i.e., they aim to reduce or increase the number of packets sent per time interval to adjust to the current level of congestion in the network). However, voice is an inelastic traffic where packets are generated at regular intervals but packet size varies with the codec that is used. Therefore, standard congestion control is not directly applicable to this type of traffic. We present three alternative modifications to equation based congestion control protocols and evaluate them through mathematical analysis and network simulation

Fairness and Stability of Congestion Control Mechanisms of TCP

this paper, we focus on fairness and stability of the congestion control mechanisms adopted in several versions of TCP by investigating their time--transient behaviors through an analytic approach. In addition to TCP Tahoe and TCP Reno, we also consider TCP Vegas which has been recently proposed for higher throughput, and enhanced TCP Vegas, which is proposed in this paper for fairness enhancements. We consider homogeneous case, where two connections have the equivalent propagation delays, and heterogeneous case, where each connection has different propagation delay. We show that TCP Tahoe and TCP Reno can achieve fairness among connections in homogeneous case, but cannot in heterogeneous case. We also show that TCP Vegas can provide almost fair service among connection, but there is some unfairness caused by the essential nature of TCP Vegas. Finally, we explain the effectiveness of our enhanced TCP Vegas in terms of fairness and throughpu