IP Networks Research Articles

Design and simulation of a software defined networking-enabled smart switch for internet of things-based smart grid

Using sustainable energy is the future of our planet earth, this became not only economically efficient but also a necessity for the preservation of life on earth. Because of such necessity, smart grids became a very important issue to be researched. Many literatures discussed this topic and with the development of internet of things (IoT) and smart sensors, smart grids are developed even further. On the other hand, software defined networking is a technology that separates the cntrol plane from the data plan of the network. It centralizes the management and the orchestration of the network tasks by using a network controller. The network controller is the heart of the SDN-enabled network, and it can control other networking devices using software defined networking (SDN) protocols such as OpenFlow. A smart switching mechanism called (SDN-smgrid-sw) for the smart grid will be modeled and controlled using SDN. We modeled the environment that interact with the sensors, for the sun and the wind elements. The Algorithm is modeled and programmed for smart efficient power sharing that is managed centrally and monitored using SDN controller. Also, all if the smart grid elements (power sources) are connected to the IP network using IoT protocols.

Towards Double Defense Network Security Based on Multi-Identifier Network Architecture.

Recently, more and more mobile devices have been connected to the Internet. The Internet environment is complicated, and network security incidents emerge endlessly. Traditional blocking and killing passive defense measures cannot fundamentally meet the network security requirements. Inspired by the heuristic establishment of multiple lines of defense in immunology, we designed and prototyped a Double Defense strategy with Endogenous Safety and Security (DDESS) based on multi-identifier network (MIN) architecture. DDESS adopts the idea of a zero-trust network, with identity authentication as the core for access control, which solves security problems of traditional IP networks. In addition, DDESS achieves individual static security defense through encryption and decryption, consortium blockchain, trusted computing whitelist, and remote attestation strategies. At the same time, with the dynamic collection of data traffic and access logs, as well as the understanding and prediction of the situation, DDESS can realize the situation awareness of network security and the cultivation of immune vaccines against unknown network attacks, thus achieving the active herd defense of network security.

Design and implementation of NDN-based Ethereum blockchain

Blockchain technology allows public parties to agree on a common state without relying on a central authority. Despite it brings many innovative use cases, the technology is still in its early stage that needs improving on many aspects. One of the issues is to deliver blockchain data more efficiently. Named Data Networking (NDN), a new network paradigm, is designed to make content distribution with ease by enabling in-network caching and built-in multicasting, which blockchain technologies can take advantage. Moreover, blockchain may contribute to extending NDN application ecosystems including decentralized applications. Therefore, it is instrumental to have a working blockchain system that runs on NDN platform to supports its research and development.In this work, we design and implement an NDN-based Ethereum blockchain platform. We propose new protocols for propagating blockchain data making full use of NDN features for the delivery of transactions and blocks. Our experiments show that the distribution of blockchain data in NDN is more efficient than that of IP network. The latency of block delivery is also reduced, which in turn supports tuning blockchain parameter for better security. Our developed blockchain client is freely distributed as an open-source project. We hope that it can provide a platform to foster blockchain research on NDN in the future.

Research on the Method of Implementing Named Data Network Interconnection Based on IP Network

In order to extend the application scope of NDN and realize the transmission of different NDNs across IP networks, a method for interconnecting NDN networks distributed in different areas with IP networks is proposed. Firstly, the NDN data resource is located by means of the DNS mechanism, and the gateway IP address of the NDN network where the data resource is located is found. Then, the transmission between different NDNs across the IP network is implemented based on the tunnel technology. In addition, in order to achieve efficient and fast NDN data forwarding, we have added a small number of NDN service nodes in the IP network, and proposed an adaptive probabilistic forwarding strategy and a link cost function-based forwarding strategy to make NDN data obtaining the cache service provided by the NDN service node as much as possible. The results of analysis and simulation experiments show that, the interconnection method of NDN across IP network proposed is generally effective and feasible, and the link cost function forwarding strategy is better than the adaptive probability forwarding strategy.

Comparison in the Applicability of MPLS When Using Different Dynamic Routing Protocols

The purpose of this paper is to examine how the Multiprotocol Label Switching (MPLS) technology affects to the latency in an IP network when using different dynamic routing protocols. For the purposes of the study, a virtual IP network was created, in which was sequentially configured to work with Routing Information Protocol (RIP), Enhanced Interior Gateway Routing Protocol (EIGRP) and Open Shortest Path First (OSPF). The studies were performed without configured and then with configured MPLS. Techniques, methods and tools used in the monitoring of IP networks were used during the study of the network.

Deep Reinforcement Learning Based Routing in IP Media Broadcast Networks: Feasibility and Performance

The Media Broadcast industry has evolved from Serial Digital Interface (SDI) based infrastructures to IP networks. While IP based video broadcast is well established in the data plane, the use of IP networks to transport media flows still poses challenges in terms of resource management and orchestration. SDN based orchestration architectures have emerged in the industry that use SDN to route the media flows of a broadcast service across the provider IP network. Several approaches to multimedia flow routing in IP based SDN networks have been proposed in the context of streaming applications over the Internet. These range from model based linear optimization solutions that have high complexity to simple shortest path based routing heuristics with either static or dynamic link costs. More recently model-free optimization methods such as Deep Reinforcement Learning (DRL) have been proposed for routing and Traffic Engineering of multimedia flows in SDN networks. The media broadcast scenario however has specific requirements, with services like Master Control Room operation and Live broadcasting of events, and it has been rarely addressed in the literature. In this work we propose a DRL based routing method for this scenario and compare it to static and dynamic link cost algorithms based on Dijkstra shortest paths. This is to our knowledge the first work to follow this approach in the context of Media Broadcast services in IP infrastructures. The algorithm is designed considering the specifications and capabilities of one of the leading SDN orchestrators in the market and considers the more common Service Level Agreement requirements in the industry. Three different DRL algorithms are implemented and compared and we evaluate them using a real service provider network topology. The results indicate that DRL based routing is applicable in real production scenarios and that it achieves considerable performance gains when compared to the static and dynamic link cost shortest path algorithms commonly used today.

On Hybrid Network Covert Channel Capacity

Covert channel is a communication channel that was not designed for information transmission. Covert channels in IP networks can be implemented in a way that is difficult to detect. That is why the method of capacity limitation of a potential covert channel seems to be an effective counteraction that can be used when allowable level of covert channel capacity is set up. To effectively protect information from leakage, it is needed to estimate covert channel capacity accurately. In this paper authors investigated the way to estimate hybrid covert channel capacity taking into account cipher type for traffic encryption.

Consumer Mobility Awareness in Named Data Networks

Mobile data traffic has increased significantly due to the evolution of wireless communication technologies. The Information Centric Network paradigm is considered as an alternative to bypass the restrictions imposed by the traditional IP networks, especially those related with the mobility of its users. Despite the potential advantages of this paradigm regarding mobile wireless environments, several research challenges remain unaddressed, more specifically the ones related with the communication damage caused by handovers. This work presents a Named Data Network (NDN) based solution that supports Consumer mobility. The proposed scheme addresses a mobility manager entity that monitors and anticipates trajectories, while compelling the infrastructure to adjust to the new paths. This process results in an efficient way to manage the Consumers’ mobility, and therefore, in a better quality of service to its users. The implementation and evaluation of the proposed solution uses the ndnSIM, through functional and non-functional scenarios, and with real traces of urban vehicular mobility and connectivity. The results show that the proposed solution is superior to the native NDN workflow with respect to content delivery ratio and number of timeouts.

Application Areas of Information-Centric Networking: State-of-the-Art and Challenges

The Information-Centric Network (ICN) paradigm has gained popularity since its inception. The host-based IP networks were not primarily designed to handle scenarios that it is exposed to on the current Internet. In that direction lot of research has been happening to develop applications such as web applications, multimedia streaming, the Internet of Things, Wireless Sensor Networks and Vehicular networks. In addition, new ICN application areas, such as social networks, Industrial IoTs, etc., are emerging. This review investigates the possible application areas and their deficiencies evenly, broadly and at a certain level of depth with focus on security, scalability, IP interoperability, modularity and other application specific aspects. We discuss the current state-of-the-art in these ICN-based applications and the existing limitations. A comparative analysis of the literary works available is performed to understand the research gaps available, and a detailed discussion of the challenges in each area is provided.We conclude the review with future challenges in the application development with the ICN paradigm to reap its architectural benefits.

USING MPLS TECHNOLOGY TO SOLVE BGP “BLACKHOLE” PROBLEM

Traditional IP networks use a hop-by-pop principle for transmitting traffic. This leads to aggregation of heterogeneous traffic on links in different parts of the network, which causes considerable possible growth of congestion and leaves the network with both unbalanced use of resources and link failure in congested parts. To support a growing number of users and multiple classes of applications with different performance requirements and characteristics, service providers have been forced to adapt to new technologies. Researchers have found that conventional IP packet forwarding is not suitable for applications such as VOIP and video conferencing, which are currently in huge demand. In addition to offer a general provision of MPLS technology, architecture, operation method and features, we will consider the Border Gateway Protocol (BGP) “Black hole” issue that results in the inability of the network to transfer traffic between some end points, and how MPLS help us avoid this problem and even optimize network operation and resources utilization.

Saving the Bandwidth of IPv6 Networks Using the Fields of the Packet Header

IPv6 protocol is the future of IP networks due to its large IP address capacity. One of the key consequences of this large capacity is that the IP protocol header has enlarged from 20-byte in IPv4 to 40-byte in IPv6. This will consume a considerable share of the bandwidth (BW) for VoIP applications, which produce small packets in tens of bytes only. To handle this issue, we have introduced an efficient technique to use the superfluous fields in the VoIP packet header, including the IPv6 header, to hold the voice data of the packet. This introduced technique is called the Eliminator technique because it shortens or eliminates the packet payload. The Eliminator technique has been analyzed against the typical technique of transferring the VoIP data. The BW utilization of the two techniques has been compared with different codecs namely G.726, G.728, and G.723.1 codecs. The analysis showed that the BW wasting has alleviated by 16.4%, 19.1%, and 25% with these codecs respectively. This BW utilization improvement will indirectly boost the quality of the VoIP calls, especially in the wireless environments. Accordingly, the Eliminator technique is a viable solution for alleviating the wasted BW when running VoIP in IPv6 networks.

Performance Analysis and Functionality Comparison of First Hop Redundancy Protocol IPV6

First hop redundancy protocols (FHRP) are an essential tool for improving IP networks’ availability. FHRP are protocols used to manage and maintain the network default gateway router by using one or more redundant routers that will take over in case of default router failure. In this paper we present the first hop redundancy concept and the means for its realization in IPv6 network. We evaluate three FHRP protocols, namely, the Hot Standby Router Protocol (HSRPv6), Virtual Router Redundancy Protocol (VRRPv3), and Gateway Load Balancing (GLBPv6) using GNS3 tools. The First Hop Redundancy Protocols have been implemented, tested, optimized, and compared to one another in terms of convergence time, packet loss and convergence time. The comparison indicates which protocol is the best in each scenario and which is the best overall among the three protocols.

IMPACT OF TUNNELING ON NETWORK CAPACITY

For many companies, setting up a VPN for secure, encrypted communication is a cost-effective alternative to purchasing, operating, and managing a separate physical network. Many institutions, corporations, government agencies and nonprofit organizations want to have their own private IP network for secure and reliable connectivity between offices across multiple geographies. A virtual private network (VPN) is a secure, encrypted connection over a public public network. Creating a separate network requires the purchase of equipment, its installation and maintenance. A VPN-based solution using the public Internet is becoming a cost-effective solution for many corporations. As a research task, the authors define an assessment of the impact of the tunneling technology used to solve problems arising from the lack of network support from existing data transfer protocols. The possibility of using the IPSec protocol and MPLS technology to implement tunneling is considered and compared. The results of the comparison and evaluation of the impact of the choice of protocol on the required bandwidth are presented.

Analysis of the capabilities of MPLS technology for managing traffic in communication networks

To support a growing number of users and multiple classes of applications with different performance requirements and characteristics, service providers have been forced to adapt to new technologies. To improve traffic management and Internet service quality, the Internet Engineering Task Force (IETF) proposed MPLS technology to support several classes of latency-critical applications. Traditional IP networks use a hop-by-pop principle for transmitting traffic. This leads to aggregation of heterogeneous traffic on links in different parts of the network, which causes considerable possible growth of congestion and leaves the network with both unbalanced use of resources and link failure in congested parts. This raises the need for traffic engineering to ensure bandwidth guarantees and efficient use of network resources. To overcome these problems, the IETF has proposed a new data transmission mechanism, which is MPLS (Multi protocol label switching), in accordance with the current requirements. The application of MPLS (Multi protocol label switching) technology in modern communication networks is defined by the author as a research task. The report discusses the prospects of MPLS as a universal technology that supports several protocols. The features of construction of virtual private networks (VPN) on MPLS are considered, and how traffic engineering in MPLS takes into account the use of resources, which makes the development of routes based on separate streams or different streams between the same endpoints more effective.

Design and Implementation of SDN IP Based on Open Network Operating System and Border Gateway Protocol

The development of computer network technology in the form of Software Defined Networking (SDN), provides many facilities for users to be able to develop network control applications, which can separate the data plane function from the control plane. The existence of this separation on routers and switches makes it easy for developers to centrally develop software and devices according to what is needed by users. However, there were obstacles to implementing SDN on IP networks in a short time. For this reason, it is necessary to implement SDN in stages by adding SDN to the existing IP network in the form of SDN IP, so that SDN can be connected and exchange routing information autonomously. This study focuses on the design and implementation of SDN IP using the Open Network Operating System (ONOS) on the Border Gateway Protocol (BGP). The results show that the design and implementation of SDN IP based on ONOS and BGP can be done well, where SDN can connect and exchange routing information with the Autonomous System (AS) native BGP-based network.
 
 Key word(s):
 Autonomous System (AS)
 Border Gateway Protocol (BGP)
 Open Network Operating System (ONOS)
 Software-Defined Networking (SDN)
 SDN IP

A Survey of 802.15.4 TSCH Schedulers for a Standardized Industrial Internet of Things.

Concepts such as Industry 4.0 and Cyber-Physical Systems may bring forward a new industrial revolution. These concepts require extensive connectivity far beyond what is provided by traditional industrial networks. The Industrial Internet of Things (IIoT) bridges this gap by employing wireless connectivity and IP networking. In order for wireless networks to meet the strict requirements of the industrial domain, the Time Slotted Channel Hopping (TSCH) MAC is often employed. The properties of a TSCH network are defined by the schedule, which dictates transmission opportunities for all nodes. We survey the literature for these schedulers, describe and organize them according to their operation: Centralized, Collaborative, Autonomous, Hybrid, and Static. For each category and the field as a whole, we provide a holistic view and describe historical trends, highlight key developments, and identify trends, such as the attention towards autonomous mechanisms. Each of the 76 schedulers is analyzed into their common components to allow for comparison between schedulers and a deeper understanding of functionality and key properties. This reveals trends such as increasing complexity and the utilization of centralized principles in several collaborative schedulers. Further, each scheduler is evaluated qualitatively to identify its objectives. Altogether this allows us to point out challenges in existing work and identify areas for future research, including fault tolerance, scalability, non-convergecast traffic patterns, and hybrid scheduling strategies.

Residual packet loss rate analysis of 2-D parity forward error correction

To improve the robustness of video communications over IP networks against packet losses, numerous IPTV and broadcasting standards have adopted the 2D application-layer forward error correction (FEC) method presented in Pro-MPEG COP3. This method uses a two-dimensional encoding matrix in which redundant checksums for each row and each column of the data matrix are generated and transmitted with the data packets. The approach has been recently extended and applied to layered video streaming. But despite its wide adoption and importance, a formal analysis of the correction performance of the Pro-MPEG COP3’s 2D FEC method is still lacking. In this paper, we formulate and solve the problem using an abstract mathematical framework, which can be used also for some natural extensions of this method. We derive tight bounds on the residual packet loss rate as a function of the 2D FEC matrix dimensions, and the packet loss rate assuming that the losses are independently and randomly distributed. We then provide simple approximations on these bounds for low packet loss rates. The analysis leads to the observation that the addition of a single checksum packet to the Pro-MPEG COP3 method would lead to a significant reduction in residual packet loss rate. The expressions derived are validated experimentally using Monte Carlo simulations. We also apply the results to the case of packet losses occurring in bursts. Finally, we study the impact of the matrix dimensions on the performance of the 2D FEC method, leading to valuable recommendations to system designers.

Compensation for Delays and Losses of Packages in Dynamic Online Games

A couple of decades ago, data rates on the network were measured in kilobytes per second, and even then, online game developers had some problems with the packet loss and transmission delays. Now the transfer rate is hundreds of times higher, and the problem of delay compensation is even more relevant.For many dynamic online games, a transmission delay of as little as 20 ms can be quite noticeable, negatively affecting the gameplay and emotions of the game, which can repel players.The problem is exacerbated by the fact that along with the need to compensate for the time of delivery of packets, on the client side there are other non-network factors that are beyond the control of developers, which make the total delay 5-10 ms longer. Because of this, the desire to get rid of network delays as much and as well as possible becomes a necessity, and developers are forced to look for optimal ways to solve this problem.The problem statement is as follows: to review the causes of delays in online games and possible solu- tions, as well as the advantages and disadvantages of certain approaches. The problem is considered at the 4 levels of the TCP / IP network model, as well as at the application level. The approaches are given for the most commonly used protocols for each layer, but basic ideas can be easily transferred to other implementations.The main causes of delays under consideration: propagation delay, router queue delay, transmission delay, and processing delays.This article shows the impact of network delays on the online games and the ways to compensate for them, along with the theory of data transmission protocols in the network and the ways to solve the problems that arise in the development of algorithms.Recommendations for solving the compensation problem can be taken into account when designing and launching online shooters, strategies, etc. Thanks to the given receptions it is possible to minimize the general delay on the transfer of packets in a network, thanks to which the game on the client looks as if the player plays in the Single Player mode.

ОГРАНИЧЕНИЕ ПРОПУСКНОЙ СПОСОБНОСТИ СЕТЕВЫХ СКРЫТЫХ КАНАЛОВ ПО ВРЕМЕНИ ПУТЕМ ВВЕДЕНИЯ ДОПОЛНИТЕЛЬНЫХ СЛУЧАЙНЫХ ЗАДЕРЖЕК ПЕРЕД ОТПРАВКОЙ ПАКЕТА

Attention to covert channels has largely increased due to the documents published by E. Snowden, which described the software and hardware embedding that implement undeclared features of covert information transfer in the Huawei and Juniper network equipment, Apple mobile phones and computers with the Windows XP operating system. A covert channel can be built using any information technology, but often attackers build covert channels in IP networks, since they are widespread, have a high information transfer rate, and ubiquitous information security measures, such as traffic encryption, do not affect the possibility of covert transmission of information via some types of such channels. The limitation of their bandwidth is a promising direction for countering information leakage via network covert channels. This study considers network timing covert channels, provides a method for assessing their capacity, proposes and investigates a way to counter information leakage via such covert channels by introducing noise having added delays before sending packets. The values of the delays are distributed in two different ways: uniformly and according to distribution with a decreasing probability density function. As an experiment, the temporal characteristics of IP traffic from a host in the internal network to a public service were obtained, which were used to verify the obtained methods for assessing the covert channel capacity. A distinctive feature of the calculations is an illustration of the possibility to minimize the load on the communication channel in the context of introducing a countermeasure method, as well as taking into consideration the fact that the intruder's ability to observe the current conditions in the network allows him to adjust the parameters of information transfer to the load in the communication channel, thereby maintaining the maximum possible covert channel bandwidth.

An Adaptive Bitrate Switching Algorithm for Speech Applications in Context of WebRTC

Web Real-Time Communication (WebRTC) combines a set of standards and technologies to enable high-quality audio, video, and auxiliary data exchange in web browsers and mobile applications. It enables peer-to-peer multimedia sessions over IP networks without the need for additional plugins. The Opus codec, which is deployed as the default audio codec for speech and music streaming in WebRTC, supports a wide range of bitrates. This range of bitrates covers narrowband, wideband, and super-wideband up to fullband bandwidths. Users of IP-based telephony always demand high-quality audio. In addition to users’ expectation, their emotional state, content type, and many other psychological factors; network quality of service; and distortions introduced at the end terminals could determine their quality of experience. To measure the quality experienced by the end user for voice transmission service, the E-model standardized in the ITU-T Rec. G.107 (a narrowband version), ITU-T Rec. G.107.1 (a wideband version), and the most recent ITU-T Rec. G.107.2 extension for the super-wideband E-model can be used. In this work, we present a quality of experience model built on the E-model to measure the impact of coding and packet loss to assess the quality perceived by the end user in WebRTC speech applications. Based on the computed Mean Opinion Score, a real-time adaptive codec parameter switching mechanism is used to switch to the most optimum codec bitrate under the present network conditions. We present the evaluation results to show the effectiveness of the proposed approach when compared with the default codec configuration in WebRTC.