Study on the Performance of TCP over 10Gbps High Speed Networks

Internet traffic is expected to grow phenomenally over the next five to ten years. To cope with such large traffic volumes, high-speed networks are expected to scale to capacities of terabits-per-second and beyond. Increasing the role of optics for packet forwarding and transmission inside the high-speed networks seems to be the most promising way to accomplish this capacity scaling. Unfortunately, unlike electronic memory, it remains a formidable challenge to build even a few dozen packets of integrated all-optical buffers. On the other hand, many high-speed networks depend on the TCP/IP protocol for reliability which is typically implemented in software and is sensitive to buffer size. For example, TCP requires a buffer size of bandwidth delay product in switches/routers to maintain nearly 100\% link utilization. Otherwise, the performance will be much downgraded. But such large buffer will challenge hardware design and power consumption, and will generate queuing delay and jitter which again cause problems. Therefore, improve TCP performance over tiny buffered high-speed networks is a top priority. This dissertation studies the TCP performance in 10Gbps high-speed networks. First, a 10Gbps reconfigurable optical networking testbed is developed as a research environment. Second, a 10Gbps traffic sniffing tool is developed for measuring and analyzing TCP performance. New expressions for evaluating TCP loss synchronization are presented by carefully examining the congestion events of TCP. Based on observation, two basic reasons that cause performance problems are studied. We find that minimize TCP loss synchronization and reduce flow burstiness impact are critical keys to improve TCP performance in tiny buffered networks. Finally, we present a new TCP protocol called Multi-Channel TCP and a new congestion control algorithm called Desynchronized Multi-Channel TCP (DMCTCP). Our algorithm implementation takes advantage of a potential parallelism from the Multi-Path TCP in Linux. Over an emulated 10Gbps network ruled by routers with only a few dozen packets of buffers, our experimental results confirm that bottleneck link utilization can be much better improved by DMCTCP than by many other TCP variants. Our study is a new step towards the deployment of optical packet switching/routing networks

Controlo de congestionamento em redes sem fios

Doutoramento em Engenharia ElectrotécnicaCongestion control in wireless networks is an important and open issue. Previous research has proven the poor performance of the Transport Control Protocol (TCP) in such networks. The factors that contribute to the poor performance of TCP in wireless environments concern its unsuitability to identify/detect and react properly to network events, its TCP window based ow control algorithm that is not suitable for the wireless channel, and the congestion collapse due to mobility. New rate based mechanisms have been proposed to mitigate TCP performance in wired and wireless networks. However, these mechanisms also present poor performance, as they lack of suitable bandwidth estimation techniques for multi-hop wireless networks. It is thus important to improve congestion control performance in wireless networks, incorporating components that are suitable for wireless environments. A congestion control scheme which provides an e - cient and fair sharing of the underlying network capacity and available bandwidth among multiple competing applications is crucial to the definition of new e cient and fair congestion control schemes on wireless multi-hop networks. The Thesis is divided in three parts. First, we present a performance evaluation study of several congestion control protocols against TCP, in wireless mesh and ad-hoc networks. The obtained results show that rate based congestion control protocols need an eficient and accurate underlying available bandwidth estimation technique. The second part of the Thesis presents a new link capacity and available bandwidth estimation mechanism denoted as rt-Winf (real time wireless inference). The estimation is performed in real-time and without the need to intrusively inject packets in the network. Simulation results show that rt-Winf obtains the available bandwidth and capacity estimation with accuracy and without introducing overhead trafic in the network. The third part of the Thesis proposes the development of new congestion control mechanisms to address the congestion control problems of wireless networks. These congestion control mechanisms use cross layer information, obtained by rt-Winf, to accurately and eficiently estimate the available bandwidth and the path capacity over a wireless network path. Evaluation of these new proposed mechanisms, through ns-2 simulations, shows that the cooperation between rt-Winf and the congestion control algorithms is able to significantly increase congestion control eficiency and network performance.O controlo de congestionamento continua a ser extremamente importante quando se investiga o desempenho das redes sem fios. Trabalhos anteriores mostram o mau desempenho do Transport Control Proto- col (TCP) em redes sem fios. Os fatores que contribuem para um pior desempenho do TCP nesse tipo de redes s~ao: a sua falta de capacidade para identificar/detetar e reagir adequadamente a eventos da rede; a utilização de um algoritmo de controlo de uxo que não é adequado para o canal sem fios; e o colapso de congestionamento devido á mobilidade. Para colmatar este problemas foram propostos novos mecanismos de controlo de congestionamento baseados na taxa de transmissão. No entanto, estes mecanismos também apresentam um pior desempenho em redes sem fios, já que não utilizam mecanismos adequados para a avaliação da largura de banda disponível. Assim, é importante para melhorar o desempenho do controlo de congestionamento em redes sem fios, incluir componentes que são adequados para esse tipo de ambientes. Um esquema de controlo de congestionamento que permita uma partilha eficiente e justa da capacidade da rede e da largura de banda disponível entre múltiplas aplicações concorrentes é crucial para a definição de novos, eficientes e justos mecanismos de controlo congestionamento para as redes sem fios. A Tese está dividida em três partes. Primeiro, apresentamos um estudo sobre a avaliação de desempenho de vários protocolos de controlo de congestionamento relativamente ao TCP, em redes sem fios em malha e ad-hoc. Os resultados obtidos mostram que os protocolos baseados na taxa de transmissão precisam de uma técnica de avaliação da largura de banda disponível que seja eficiente e precisa . A segunda parte da Tese apresenta um novo mecanismo de avaliação da capacidade da ligação e da largura de banda disponível, designada por rt-Winf (real time wireless inference). A avaliação é realizada em tempo real e sem a necessidade de inserir tráfego na rede. Os resultados obtidos através de simulação e emulação mostram que o rt-Winf obtém com precisão a largura de banda disponível e a capacidade da ligação sem sobrecarregar a rede. A terceira parte da Tese propõe novos mecanismos de controlo de congestionamento em redes sem fios. Estes mecanismos de controlo de congestionamento apresentam um conjunto de caracter ísticas novas para melhorar o seu desempenho, de entre as quais se destaca a utilização da informação de largura de banda disponível obtida pelo rt-Winf. Os resultados da avaliação destes mecanismos, utilizando o simulador ns-2, permitem concluir que a cooperação entre o rt-Winf e os algoritmos de controlo de congestionamento aumenta significativamente o desempenho da rede

Evaluation of Active Queue Management (AQM) Models in Low Latency Networks

Abstract: Low latency networks require the modification of the actual queuing management in order to avoid large queuing delay. Nowadays, TCP’s congestion control maximizes the throughput of the link providing benefits to large flow packets. However, nodes’ buffers may get fully filled, which would produce large time delays and packet dropping situations, named as bufferbloat problem. For actual time-sensitive applications demand, such as VoIP, online gaming or financial trading, these queueing times cause bad quality of service being directly noticed in user’s utilization. This work studies the different alternatives for active queue management (AQM) in the nodes links, optimizing the latency of the small flow packets and, therefore, providing better quality for low latency networks in congestion scenarios. AQM models are simulated in a dumbbell topology with ns3 software, which shows the diverse latency values (measured in RTT) according to network situations and the algorithm that has been installed. In detail, RED, CoDel, PIE, and FQ_CoDel algorithms are studied, plus the modification of the TCP sender’s congestion control with Alternative Backoff with ECN (ABE) algorithm. The simulations will display the best queueing times for the implementation that mixes FQ_CoDel with ABE, the one which maximizes the throughput reducing the latency of the packets. Thus, the modification of queueing management with FQ_CoDel and the implementation of ABE in the sender will solve the bufferbloat problem offering the required quality for low latency networks.Resumen Las redes de baja latencia requieren la modificación de la actual gestión de las colas con el fin de eludir los extensos tiempos de retardo. Hoy en d´ıa, el control de congestión de TCP maximiza el rendimiento (throughput) del enlace otorgando beneficio a los grandes flujos de datos, sin embargo, los buffers son plenamente cargados generando altos tiempos de retardo y fases de retirada de paquetes, llamada a esta situación el problema de Bufferbloat. Par las aplicaciones contempor´aneas como las llamadas VoIP, los juegos on-line o los intercambios financieros; estos tiempos de cola generan una mala calidad de servicio detectada directamente por los usuarios finales. Este trabajo estudia las diferentes alternativas de la gestión activa de colas (AQM), optimizando la latencia de los peque˜nos flujos y, por lo tanto, brindando una mejor calidad para las redes de baja latencia en situaciones de congestión. Los modelos AQM han sido evaluados en una topolog´ıa ’dumbbell’ mediante el simulador ns3, entregando resultados de latencia (medidos en RTT) de acuerdo con la situación del enlace y el algoritmo instalado en la cola. Concretamente, los algoritmos estudiados han sido RED, CoDel, PIE y FQ_CoDel; adem´as de la modificación del control de congestión TCP del emisor denominada ABE (Alternative Backoff with ECN). Las simulaciones que mejor resultados ofrecen son las que implementan combinación de FQ_CoDel con el algoritmo ABE, maximizando el rendimiento y reduciendo la latencia de los paquetes. Por lo tanto, la modificación con FQ_CoDel en las colas y la de ABE en el emisor ofrecen una solución al problema del Bufferbloat altamente solicitada por las redes de baja latencia

Model based analysis of some high speed network issues

The study of complex problems in science and engineering today typically involves large scale data, huge number of large-scale scientific breakthroughs critically depends on large multi-disciplinary and geographically-dispersed research teams, where the high speed network becomes the integral part. To serve the ongoing bandwidth requirement and scalability of these networks, there has been a continuous evolution of different TCPs for high speed networks. Testing these protocols on a real network would be expensive, time consuming and more over not easily available to the researchers worldwide. Network simulation is well accepted and widely used method for performance evaluation, it is well known that packet-based simulators like NS2 and Opnet are not adequate in high speed also in large scale networks because of its inherent bottlenecks in terms of message overhead and execution time. In that case model based approach with the help of a set of coupled differential equations is preferred for simulations. This dissertation is focused on the key challenges on research and development of TCPs on high-speed network. To address these issues/challenges this thesis has three objectives: design an analytical simulation methodology; model behaviors of high speed networks and other components including TCP flows and queue using the analytical simulation method; analyze them and explore impacts and interrelationship among them. To decrease the simulation time and speed up the process of testing and development of high speed TCP, we present a scalable simulation methodology for high speed network. We present the fluid model equations for various high-speed TCP variants. With the help of these fluid model equations, the behavior of high-speed TCP variants under various scenarios and its effect on queue size variations are presented. High speed network is not feasible unless we understand effect of bottleneck buffer size on performance of these high-speed TCP variants. A fluid model is introduced to accommodate the new observations of synchronization and de-synchronization phenomena of packet losses at bottleneck link and a microscopic analysis is presented on different buffer sizes at drop-tail queuing scheme. The proposed model based methods promotes principal understanding of the future heterogeneous networks and accelerates protocol developments

Design, Implementation, and Evaluation of Join and Split Strategy for Transmission control protocol running on Software Defined Networks

Software Defined Networks (SDN)-enabled switches of today can be empowered to intelligently forward as well as elastically steer the network traffic. In this work, we focus on developing a SDN-based framework to provide improved delivery performance (of applications) in the network. This dissertation proposed a new TCP join and split proxy on SDN platform. The proposed framework allowed part of TCP (Transmission Control Protocol) optimization to migrate from the application server to the proxy. Therefore, with a control plane built between SDN controller and proxy, the SDN controller can further improve the TCP delivery performance. The proxy (join-proxy) joins all TCP flows at the beginning of the shared path into one long TCP flow. At the end of the shared path, the proxy (split-proxy) splits the long flow for each joined client with the same TCP session state. With the help of centralized controller of SDN and customized SDN switch, the new design simplifies the TCP session synchronization between proxies. Also, this dissertation developed Linked-ACK ((Acknowledgement) to maintain the end-to-end semantic and limit the buffer size in each proxy by coupling the ACK of three TCP flows separated by the join and split proxy. At the last, this dissertation shows that the proposed proxy can well integrate with wireless network and MPTCP (Multi-Path TCP) proxy [1] The extensions of the proposed TCP Join and Split platform are applied to Smart Grid network for improving fairness, WiFi network for reducing gaming traffic delay, and Data Center network for addressing Virtual Machine (VM) live migration problem. First, the proposed TCP Join and Split platform can be applied to Smart Grid network to provide better fairness on the application layer. The latest research in Smart Grid communications has advocated the aggregation of multiple traffic flows in order to achieve an improved throughput. While aggregation improves the overall throughput, the individual flows still suffer from unfair throughput performance. As a result, the enablers for time sensitive Smart Grid services, such as load-shedding which requires a timely report of data, are mostly affected. This dissertation proposed a novel SDN-based framework to provide fairness among smart-meters (SMs) through flow aggregation and scheduling. By exploring the SDN’s flow-level manageability features, for the first time in this paper, we present an implementation-based architecture to perform effective aggregation-and-scheduling of traffic flows. The proposed framework ensures fairness (among the smart-meters) as well as improve the throughput performance. Our extensive experimental results validate the efficacy of our proposed framework. Second, the proposed TCP Join and Split platform can be applied to WiFi network to reduce the gaming traffic delay. WiFi users typically expect different performance requirements for various types of applications. For instance, users expect 'better and consistent throughput' for Internet video consumption, and 'minimal delay' for local network gaming applications. The wireless access substrate (at the consumer-end), typically being the bottleneck in these networks, causes different users (in the same WiFi coverage) to experience unfair and fluctuating network performance. To combat such unfair situations, we need approaches to effectively control and steer the applications’ traffic in the shared WiFi medium. However, a network that deals with a crowd or private end-users (such as gaming multiplayers or the Internet content distributors), encounters a major challenge in controlling the traffic without involvement or modification at the end-host application devices. In this dissertation, we propose a SDN-based seamless traffic steering and control strategy in order to provide effective application-specific delivery services, such as reduced delay (for gaming traffic) and improved throughput (for video consumption). Unlike simulation-based solutions, our approach is production-ready, as we have implemented our framework on a real network testbed environment. With extensive performance study and sufficient mathematical insight, we demonstrate the prowess of our proposed framework. Last but not the least, the proposed TCP Join and Split platform can be applied to Data Center network to optimize the VM live migration. With the growth of data volumes and a variety of Internet applications, virtualization has become commonplace in modern data centers and an effective solution to provide better management flexibility, lower cost, scalability, better resources utilization, and energy efficiency. One of the powerful features provided by virtualization is Virtual Machine (VM) live migration, which facilitates moving workloads within the infrastructure with negligible downtime and minimal impact on workload. However, the performance of running applications is likely to be negatively affected during a live VM migration. The objective of this paper is to optimize the total performance degradation of concurrent VM live migration in the data center network by exploiting the SDN platform. The problem is modeled using mixed integer linear programming(MILP) for VM live migration with a fixed path and VM live migration with path selection. To provide a practical optimization, the greedy algorithm is proposed. Numerical study results show that a significant decrease occur in performance degradation in MILP model and greedy algorithm when the number of VMs increases. The proposed greedy algorithm cannot yield the optimum solution as the problem become harder, but it provides better solution than MILP model in terms of the time constrain exhibited in case of large problems

MANETs: Internet Connectivity and Transport Protocols

A Mobile Ad hoc Network (MANET) is a collection of mobile nodes connected together over a wireless medium, which self-organize into an autonomous multi-hop wireless network. This kind of networks allows people and devices to seamlessly internetwork in areas with no pre-existing communication infrastructure, e.g., disaster recovery environments. Ad hoc networking is not a new concept, having been around in various forms for over 20 years. However, in the past only tactical networks followed the ad hoc networking paradigm. Recently, the introduction of new technologies such as IEEE 802.11, are moved the application field of MANETs to a more commercial field. These evolutions have been generating a renewed and growing interest in the research and development of MANETs. It is widely recognized that a prerequisite for the commercial penetration of the ad hoc networking technologies is the integration with existing wired/wireless infrastructure-based networks to provide an easy and transparent access to the Internet and its services. However, most of the existing solutions for enabling the interconnection between MANETs and the Internet are based on complex and inefficient mechanisms, as Mobile-IP and IP tunnelling. This thesis describes an alternative approach to build multi-hop and heterogeneous proactive ad hoc networks, which can be used as flexible and low-cost extensions of traditional wired LANs. The proposed architecture provides transparent global Internet connectivity and address autocofiguration capabilities to mobile nodes without requiring configuration changes in the pre-existing wired LAN, and relying on basic layer-2 functionalities. This thesis also includes an experimental evaluation of the proposed architecture and a comparison between this architecture with a well-known alternative NAT-based solution. The experimental outcomes confirm that the proposed technique ensures higher per-connection throughputs than the NAT-based solution. This thesis also examines the problems encountered by TCP over multi-hop ad hoc networks. Research on efficient transport protocols for ad hoc networks is one of the most active topics in the MANET community. Such a great interest is basically motivated by numerous observations showing that, in general, TCP is not able to efficiently deal with the unstable and very dynamic environment provided by multi-hop ad hoc networks. This is because some assumptions, in TCP design, are clearly inspired by the characteristics of wired networks dominant at the time when it was conceived. More specifically, TCP implicitly assumes that packet loss is almost always due to congestion phenomena causing buffer overflows at intermediate routers. Furthermore, it also assumes that nodes are static (i.e., they do not change their position over time). Unfortunately, these assumptions do not hold in MANETs, since in this kind of networks packet losses due to interference and link-layer contentions are largely predominant, and nodes may be mobile. The typical approach to solve these problems is patching TCP to fix its inefficiencies while preserving compatibility with the original protocol. This thesis explores a different approach. Specifically, this thesis presents a new transport protocol (TPA) designed from scratch, and address TCP interoperability at a late design stage. In this way, TPA can include all desired features in a neat and coherent way. This thesis also includes an experimental, as well as, a simulative evaluation of TPA, and a comparison between TCP and TPA performance (in terms of throughput, number of unnecessary transmissions and fairness). The presented analysis considers several of possible configurations of the protocols parameters, different routing protocols, and various networking scenarios. In all the cases taken into consideration TPA significantly outperforms TCP

Evaluation of Active Queue Management (AQM) Models in Low Latency Networks

Abstract: Low latency networks require the modification of the actual queuing management in order to avoid large queuing delay. Nowadays, TCP’s congestion control maximizes the throughput of the link providing benefits to large flow packets. However, nodes’ buffers may get fully filled, which would produce large time delays and packet dropping situations, named as bufferbloat problem. For actual time-sensitive applications demand, such as VoIP, online gaming or financial trading, these queueing times cause bad quality of service being directly noticed in user’s utilization. This work studies the different alternatives for active queue management (AQM) in the nodes links, optimizing the latency of the small flow packets and, therefore, providing better quality for low latency networks in congestion scenarios. AQM models are simulated in a dumbbell topology with ns3 software, which shows the diverse latency values (measured in RTT) according to network situations and the algorithm that has been installed. In detail, RED, CoDel, PIE, and FQ_CoDel algorithms are studied, plus the modification of the TCP sender’s congestion control with Alternative Backoff with ECN (ABE) algorithm. The simulations will display the best queueing times for the implementation that mixes FQ_CoDel with ABE, the one which maximizes the throughput reducing the latency of the packets. Thus, the modification of queueing management with FQ_CoDel and the implementation of ABE in the sender will solve the bufferbloat problem offering the required quality for low latency networks.Resumen Las redes de baja latencia requieren la modificación de la actual gestión de las colas con el fin de eludir los extensos tiempos de retardo. Hoy en d´ıa, el control de congestión de TCP maximiza el rendimiento (throughput) del enlace otorgando beneficio a los grandes flujos de datos, sin embargo, los buffers son plenamente cargados generando altos tiempos de retardo y fases de retirada de paquetes, llamada a esta situación el problema de Bufferbloat. Par las aplicaciones contempor´aneas como las llamadas VoIP, los juegos on-line o los intercambios financieros; estos tiempos de cola generan una mala calidad de servicio detectada directamente por los usuarios finales. Este trabajo estudia las diferentes alternativas de la gestión activa de colas (AQM), optimizando la latencia de los peque˜nos flujos y, por lo tanto, brindando una mejor calidad para las redes de baja latencia en situaciones de congestión. Los modelos AQM han sido evaluados en una topolog´ıa ’dumbbell’ mediante el simulador ns3, entregando resultados de latencia (medidos en RTT) de acuerdo con la situación del enlace y el algoritmo instalado en la cola. Concretamente, los algoritmos estudiados han sido RED, CoDel, PIE y FQ_CoDel; adem´as de la modificación del control de congestión TCP del emisor denominada ABE (Alternative Backoff with ECN). Las simulaciones que mejor resultados ofrecen son las que implementan combinación de FQ_CoDel con el algoritmo ABE, maximizando el rendimiento y reduciendo la latencia de los paquetes. Por lo tanto, la modificación con FQ_CoDel en las colas y la de ABE en el emisor ofrecen una solución al problema del Bufferbloat altamente solicitada por las redes de baja latencia

Optimization and Performance Analysis of High Speed Mobile Access Networks

The end-to-end performance evaluation of high speed broadband mobile access networks is the main focus of this work. Novel transport network adaptive flow control and enhanced congestion control algorithms are proposed, implemented, tested and validated using a comprehensive High speed packet Access (HSPA) system simulator. The simulation analysis confirms that the aforementioned algorithms are able to provide reliable and guaranteed services for both network operators and end users cost-effectively. Further, two novel analytical models one for congestion control and the other for the combined flow control and congestion control which are based on Markov chains are designed and developed to perform the aforementioned analysis efficiently compared to time consuming detailed system simulations. In addition, the effects of the Long Term Evolution (LTE) transport network (S1and X2 interfaces) on the end user performance are investigated and analysed by introducing a novel comprehensive MAC scheduling scheme and a novel transport service differentiation model

Discrete-time queueing model for responsive network traffic and bottleneck queues

The Internet has been more and more intensively used in recent years. Although network infrastructure has been regularly upgraded, and the ability to manage heavy traffic greatly increased, especially on the core networks, congestion never ceases to appear, as the amount of traffic that flow on the Internet seems to be increasing at an even faster rate. Thus, congestion control mechanisms play a vital role in the functioning of the Internet. Active Queue Management (AQM) is a popular type of congestion control mechanism that is implemented on gateways (most notably routers), which can predict and avoid the congestion before it happens. When properly configured, AQMs can effectively reduce the congestion, and alleviate some of the problems such as global synchronisation and unfairness to bursty traffic. However, there are still many problems regarding AQMs. Most of the AQM schemes are quite sensitive to their parameters setting, and these parameters may be heavily dependent on the network traffic profile, which the administrator may not have intensive knowledge of, and is likely to change over time. When poorly configured, many AQMs perform no better than the basic drop-tail queue. There is currently no effective method to compare the performance of these AQM algorithms, caused by the parameter configuration problem. In this research, the aim is to propose a new analytical model, which mainly uses discrete-time queueing theory. A novel transient modification to the conventional equilibrium-based method is proposed, and it is utilised to further develop a dynamic interactive model of responsive traffic and bottleneck queues. Using step-by-step analysis, it represents the bursty traffic and oscillating queue length behaviour in practical network more accurately. It also provides an effective way of predicting the behaviour of a TCP-AQM system, allowing easier parameter optimisation for AQM schemes. Numerical solution using MATLAB and software simulation using NS-2 are used to extensively validate the proposed models, theories and conclusions

5GAuRA. D3.3: RAN Analytics Mechanisms and Performance Benchmarking of Video, Time Critical, and Social Applications

5GAuRA deliverable D3.3.This is the final deliverable of Work Package 3 (WP3) of the 5GAuRA project, providing a report on the project’s developments on the topics of Radio Access Network (RAN) analytics and application performance benchmarking. The focus of this deliverable is to extend and deepen the methods and results provided in the 5GAuRA deliverable D3.2 in the context of specific use scenarios of video, time critical, and social applications. In this respect, four major topics of WP3 of 5GAuRA – namely edge-cloud enhanced RAN architecture, machine learning assisted Random Access Channel (RACH) approach, Multi-access Edge Computing (MEC) content caching, and active queue management – are put forward. Specifically, this document provides a detailed discussion on the service level agreement between tenant and service provider in the context of network slicing in Fifth Generation (5G) communication networks. Network slicing is considered as a key enabler to 5G communication system. Legacy telecommunication networks have been providing various services to all kinds of customers through a single network infrastructure. In contrast, by deploying network slicing, operators are now able to partition one network into individual slices, each with its own configuration and Quality of Service (QoS) requirements. There are many applications across industry that open new business opportunities with new business models. Every application instance requires an independent slice with its own network functions and features, whereby every single slice needs an individual Service Level Agreement (SLA). In D3.3, we propose a comprehensive end-to-end structure of SLA between the tenant and the service provider of sliced 5G network, which balances the interests of both sides. The proposed SLA defines reliability, availability, and performance of delivered telecommunication services in order to ensure that right information is delivered to the right destination at right time, safely and securely. We also discuss the metrics of slicebased network SLA such as throughput, penalty, cost, revenue, profit, and QoS related metrics, which are, in the view of 5GAuRA, critical features of the agreement.Peer ReviewedPostprint (published version