End-to-End Simulation of 5G mmWave Networks

Due to its potential for multi-gigabit and low latency wireless links, millimeter wave (mmWave) technology is expected to play a central role in 5th generation cellular systems. While there has been considerable progress in understanding the mmWave physical layer, innovations will be required at all layers of the protocol stack, in both the access and the core network. Discrete-event network simulation is essential for end-to-end, cross-layer research and development. This paper provides a tutorial on a recently developed full-stack mmWave module integrated into the widely used open-source ns--3 simulator. The module includes a number of detailed statistical channel models as well as the ability to incorporate real measurements or ray-tracing data. The Physical (PHY) and Medium Access Control (MAC) layers are modular and highly customizable, making it easy to integrate algorithms or compare Orthogonal Frequency Division Multiplexing (OFDM) numerologies, for example. The module is interfaced with the core network of the ns--3 Long Term Evolution (LTE) module for full-stack simulations of end-to-end connectivity, and advanced architectural features, such as dual-connectivity, are also available. To facilitate the understanding of the module, and verify its correct functioning, we provide several examples that show the performance of the custom mmWave stack as well as custom congestion control algorithms designed specifically for efficient utilization of the mmWave channel.Comment: 25 pages, 16 figures, submitted to IEEE Communications Surveys and Tutorials (revised Jan. 2018

TCP with Network Coding Performance Under Packet Reordering

The adverse impact of packet reordering besides packet loss is significant on the goodput performance of TCP (Transmission Control Protocol), a dominant protocol for reliable and connection-oriented transmission. With the primary purpose of improving the TCP goodput in lossy networks, the Network Coding technique was introduced. TCP/NC (TCP with Network Coding) is a promising approach which can recover lost packets without retransmission. However, the packet reordering has not been considered, and no study on that issue is found for TCP/NC. Therefore, in this paper, we investigate the goodput performance degradation due to the out-of-order reception of data or acknowledgment packets and propose a new scheme for TCP/NC to estimate and adapt to the packet reordering. The results of our simulation on ns-3 (Network Simulation 3) suggest that the proposed scheme can maintain the TCP goodput well in a wide range of packet reordering environments compared to TCP NewReno as well as TCP/NC.International Conference on Emerging Internetworking, Data & Web Technologies (EIDWT 2019), 26-28 February, 2019, Fujairah Campus, United Arab Emirate

A control theoretic approach to achieve proportional fairness in 802.11e EDCA WLANs

This paper considers proportional fairness amongst ACs in an EDCA WLAN for provision of distinct QoS requirements and priority parameters. A detailed theoretical analysis is provided to derive the optimal station attempt probability which leads to a proportional fair allocation of station throughputs. The desirable fairness can be achieved using a centralised adaptive control approach. This approach is based on multivariable statespace control theory and uses the Linear Quadratic Integral (LQI) controller to periodically update CWmin till the optimal fair point of operation. Performance evaluation demonstrates that the control approach has high accuracy performance and fast convergence speed for general network scenarios. To our knowledge this might be the first time that a closed-loop control system is designed for EDCA WLANs to achieve proportional fairness

Building self-optimized communication systems based on applicative cross-layer information

This article proposes the Implicit Packet Meta Header(IPMH) as a standard method to compute and represent common QoS properties of the Application Data Units (ADU) of multimedia streams using legacy and proprietary streams’ headers (e.g. Real-time Transport Protocol headers). The use of IPMH by mechanisms located at different layers of the communication architecture will allow implementing fine per-packet selfoptimization of communication services regarding the actual application requirements. A case study showing how IPMH is used by error control mechanisms in the context of wireless networks is presented in order to demonstrate the feasibility and advantages of this approach

TCP network coding with adapting parameters for bursty and time-varying loss

The Transmission Control Protocol (TCP) with Network Coding (TCP/NC) was proposed to introduce packet loss recovery ability at the sink without TCP retransmission, which is realized by proactively sending redundant combination packets encoded at the source. Although TCP/NC is expected to mitigate the goodput degradation of TCP over lossy networks, the original TCP/NC does not work well in burst loss and time-varying channels. No apparent scheme was provided to decide and change the network coding-related parameters (NC parameters) to suit the diverse and changeable loss conditions. In this paper, a solution to support TCP/NC in adapting to mentioned conditions is proposed, called TCP/NC with Loss Rate and Loss Burstiness Estimation (TCP/NCwLRLBE). Both the packet loss rate and burstiness are estimated by observing transmitted packets to adapt to burst loss channels. Appropriate NC parameters are calculated from the estimated probability of successful recoverable transmission based on a mathematical model of packet losses. Moreover, a new mechanism for coding window handling is developed to update NC parameters in the coding system promptly. The proposed scheme is implemented and validated in Network Simulator 3 with two different types of burst loss model. The results suggest the potential of TCP/NCwLRLBE to mitigate the TCP goodput degradation in both the random loss and burst loss channels with the time-varying conditions

ARQ protocol for joint source and channel coding and its applications

Shannon\u27s separation theorem states that for transmission over noisy channels, approaching channel capacity is possible with the separation of source and channel coding. Practically, the situation is different. Infinite size blocks are needed to achieve this theoretical limit. Also, time-varying channels require a different approach. This leads to many approaches for source and channel coding. This dissertation will address a joint source and channel coding that suits Automatic Repeat Request (ARQ) application and applies it to packet switching networks. Following aspects of the proposed joint source and channel coding approach will be presented: The design of the proposed joint source and channel coding scheme. The approach is based on a variable length coding scheme which adapts the arithmetic coding process for joint source and channel coding. The protocol using this joint source and channel coding scheme in communication systems. The error recovery technique of the proposed scheme is presented. The application of the scheme and protocol. The design is applied to wireless TCP network and real-time video transmissions. The coding scheme embeds the redundancy needed for error detection in source coding stage. The self-synchronization property of lossless compression is utilized by decoder to detect channel errors. With this approach, error detection may be delayed. The delay in detection is referred to as error propagation distance. This work analyzes the distribution of error propagation distance. The error recovery technique of this joint source and channel coding for ARQ (JARQ) protocol is analyzed. Throughput is studied using signal flow graph for both independent channel and nonindependent channels. A packet combining technique is presented which utilizes the non-uniform distribution of error propagation distance to increase the throughput. The proposed scheme may be applied to many areas. In particular, two applications are discussed. A TCP/JARQ protocol stack is introduced and the coordination between TCP and JARQ layers is discussed to maximize system performance. By limiting the number of retransmission, the proposed scheme is applied to real-time transmission to meet timing requirement