IP address configuration algorithms for routerless and single-router zeroconf networks

In Wireless sensor networks (WSNs), group mobility is important in many practical application scenarios, and it is inconvenient for traditional static WSNs to collect information. In this paper, we propose a WSN model based on IPv6 over Low power Wireless Personal Area Networks (6LoWPANs) in mobile sensing application scenarios. Wireless mesh sensor network (WMSN) infrastructure and Network Mobility (NEMO) protocol are deployed to ensure the continuity of the communications. The routing optimization mechanism in the nested network is then discussed. By taking mobile network partition and IP address configuration in the nested network, the corresponding signaling flow is discussed. Simulation results indicate that our mechanism is able to minimize the transmission costing, handoff delay, throughput and energy consumption of sensor nodes. Energy of each mobile router (MR) saves around 0.2 J per 3 seconds.

In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, a number of issues have to be solved which have hampered it in the Internet. Therefore, in this paper the VoIP (Voice over Internet Protocol) network design problems are modeled as nonlinear non convex combinatorial integer mathematical programming problems. The optimization problems have great practice value for the network service providers. This problem is a constrained multicast QoS routing and is proved to be a NP complete problem. The total link capacity augmented cost should be minimized when we design a new VoIP network or when the original network could not serve all of the traffic demands. It is also known as a complicated problem. Introduction Originally, the Internet was designed to provide best-effort services for the data generated by computers. The timeliness of data communication is generally delay tolerant. Quality-of-Service (QoS) constraints are not as important as routing flexibility and connectivity. Hence, using IP to transport voice data is contradictory to the basic requirement of the voice service: a timely delivery of voice samples. Although IP was not initially designed to provide services for real-time traffic, recent technical progress has made IP have the capabilities to provide real-time services in near future. In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, we must deal with a number of issues that have hampered it in the Internet. Voice service requirements could be discussed from two perspectives: (1) application requirements such as end-to-end delay, jitter, packet loss and overdue probability; (2) user’s perspective such as reliability, availability, and supplementary services [1][2]. Telephony service providers must guarantee the quality of service they provide e.g. maximum one way delay does not exceed 150ms. Internet telephony requires a range of protocols, ranging from those needed for transporting real-time data across to the network e.g. Real-time Transport protocol (RTP) [3], to Quality-of-Service aware routing (QoS routing), signaling protocol, resource reservation, internetworking between IP networks and PSTN, QoS-aware network management and billing protocols. ITU-T defined H.323 to provide multimedia communication in packet networks. From the perspective of network service providers, they want to optimize the network performance such as minimizing the total bandwidth consumption, maximizing throughput or total revenue [4][5] subject to user and application constraints. In this paper, we want to develop the mathematical model for VoIP network. We minimize the total bandwidth consumption under users’ QoS requirements, the network topology and the network capacity. Performance Optimization Model It is generally accepted that Internet telephony and traditional circuit-switched telephony will coexist for quite some time. The VoIP architecture must deal with interworking between IP networks and PSTN, so we need gateways between the two worlds. There are four possible models of VoIP [6]. They are PC-to-PC, Gateway-to-Gateway, PC-to-Gateway, and Gateway-to-PC models. The architecture of VoIP is shown in Fig.1. The first model of VoIP is PC-to-PC architecture, which based 2nd International Conference on Electrical, Computer Engineering and Electronics (ICECEE 2015) © 2015. The authors Published by Atlantis Press 1144 on the assumption that two or more users have access to multimedia computers that are connected to the Internet. Fig.1 VoIP Architecture The VoIP system is modeled as a graph, where the hosts, VoIP gateways, and switches are represented by nodes and communication link sets are represented by links. Let N = {1, 2,..., n} be the set of nodes and L be the set of links in the graph (network). Let G be the set of all user groups. An user group g is a voice communication session requesting for transmission in the network. An user group g may be an unicast from the source to a destination or a multicast from the source to multiple destinations. For each user group g, the traffic is transmitted exactly over one tree. Dg represents the set of destinations of user group g. g λ is the required bandwidth of the voice transmission for each user group g. Below is a verbal description of the VoIP system design problem we considered. Given : Network topology Capacities of network links Equivalent bandwidth of each multicast/unicast group Time threshold and tolerable overdue probability for each multicast/unicast group To determine : The minimum overall bandwidth consumption Routing tree for each multicast/unicast group Overdue probability of each multicast/unicast group Objective : To minimize the total bandwidth consumption Subject to : End-to-end QoS (overdue probability) constraint Tree constraint Multi-commodity flow constraint Capacity constraint Integrality constraint Hop constraint Table 1 and 2 are the notations we use in this paper . Internet VoIP Gateway VoIPGatewa Central office switch or local hub Central office switch or local hub Multimedia PC IP Router Multimedia PC IP Router Gatekeepe r

We have been developing a network visualization technique: a network architecture technology to overlay IP networks over an optical backbone network to improve flexibility and resiliency. Virtual network topology created in the optical backbone network is used as a substrate for a IP network. In order to improve flexibility and resiliency, we need to implement carrier-grade mechanisms for operation, administration, and maintenance. For the IP network, the following management and controls are addressed: traffic measurement, route trace, topology and route optimization, and failure analysis. For the optical backbone network, the impact of the GMPLS control plane is addressed. We also discuss the integrated operation and management of the IP and optical backbone networks.

IP hosts and network infrastructure have historically been difficult to configure, but emerging networking protocols promise to enable hosts to establish IP networks without prior configuration or network services. Even very simple devices with few computing resources will be able to communicate via standard protocols wherever they are attached. Current IETF (Internet Engineering Task Force) standardization efforts, such as those in the Zeroconf Working Group, aim to make this form of networking simple and inexpensive. In this tutorial, I examine the background, current status and future prospects for zero-configuration networking.

Knowledge of the up-to-date physical (i.e., layer-2) topology of an Ethernet network is crucial to a number of critical network management tasks, including reactive and proactive resource management, event correlation, and root-cause analysis. Given the dynamic nature of today's IP networks, keeping track of topology information manually is a daunting (if not impossible) task. Thus, effective algorithms for automatically discovering physical network topology are necessary. In this paper, we propose the first complete algorithmic solution for discovering the physical topology of a large, heterogeneous Ethernet network comprising multiple subnets as well as (possibly) dumb or uncooperative network elements. Our algorithms rely on standard SNMP MIB information that is widely supported in modern IP networks and require no modifications to the operating system software running on elements or hosts. Furthermore, we formally demonstrate that our solution is complete for the given MIB data; that is, if the MIB information is sufficient to uniquely identify the network topology then our algorithm is guaranteed to recover it. To the best of our knowledge, ours is the first solution to provide such a strong completeness guarantee.

Nowadays, with the increasing scale of the IP network, the traffic flow demand also increases. And the situation of load imbalance is becoming more obvious. In the IP network, the failure of a link not only causes the interruption of the traffic flow on this link, but also makes the network operator pay the corresponding penalty to the user. So how to select an appropriate path to reduce the operator's payment while achieving a balanced distribution of the traffic flow at the same time is a big challenge. Usually, traffic equilibrium is used as a routing optimization objective in traffic engineering in order to avoid network traffic flow congestion. In practical problems, the economic losses of the network operators also need to be taken into consideration. Existing link allocation methods often take the failure of links as connection problems and ignore the restrictions of the traffic flow, or only consider the traffic interruption and ignore the network operator's economic losses caused by the failure of links. In this paper, we propose a routing optimization algorithm to achieve traffic equilibrium while reducing the potential penalty of the network operators at the same time. Experiments show that our algorithm can achieve traffic equilibrium and effectively reduce the network operator's potential penalty.

The conventional forwarding rule used by IP networks is to always choose the path with the shortest length – in terms of administrative link weights assigned to the links – to forward traffic. Lately, it has been proposed to use shortest-path-first routing to implement Traffic Engineering in IP networks, promising with a big boost in the profitability of the legacy network infrastructure. The idea is to set the link weights so that the shortest paths, and the traffic thereof, follow the paths designated by the operator. Unfortunately, traditional methods to calculate the link weights usually produce a bunch of superfluous shortest paths, often leading to congestion along the unconsidered paths. In this paper, we introduce and develop novel methods to increase the accuracy of this process and, by means of extensive simulations, we show that our proposed solution produces remarkably high quality link weights.

First hop redundancy protocols are an essential tool for improving the availability of IP networks. Several protocols exist to rapidly configure redundant paths in the event of a link failure but a review of literature provides little guidance on their relative performance. This paper proposes a method for appraising the effectiveness of three common first hop redundancy protocols within IP networks and critically analyses the approach. Results from a typical network topology are presented as well as a discussion around the validity of the results and potential sources of error. Consideration is given to the potential of advanced numerical methods to minimise error, particularly in large scale networks. Recommendations are made regarding the optimal configurations to be used when comparing first hop redundancy protocols, as well as further research required to improve the accuracy of the results obtained.

In this paper, we propose an extensive analysis of different co-located router network topologies within the same POP. This analysis in done in the context of the POP topological optimization problem in IP networks. This problem consists of selecting the number of routers and their type, selecting the interface card types and connecting the access and the backbone links to the ports. The goal is to find the minimum cost POP. Three topologies are analyzed: tree, ring and full-meshed topologies. For each of those topologies we propose a complexity analysis, feasibility conditions, and valid inequalities.

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.

With the development of big data and cloud computing technology, the traditional software-defined network is facing new challenges (e.i., ubiquitous accessibility, higher bandwidth, more flexible management and greater security). Using a proprietary protocol or encoding format is a way to improve information security. However, the flow, which carried by proprietary protocol or code, cannot go through the traditional IP network. In addition, ultra- high-definition video transmission service once again become a hot spot. Traditionally, in the IP network, the Serial Digital Interface (SDI) signal must be compressed. This approach offers additional advantages but also bring some disadvantages such as signal degradation and high latency. To some extent, HD-SDI can also be regard as a proprietary protocol, which need transparent transmission such as optical channel. However, traditional optical networks cannot support flexible traffics . In response to aforementioned challenges for future network, one immediate solution would be to use NFV technology to abstract the network infrastructure and provide an all-optical switching topology graph for the SDN control plane. This paper proposes a new service-based software defined optical network architecture, including an infrastructure layer, a virtualization layer, a service abstract layer and an application layer. We then dwell on the corresponding service providing method in order to implement the protocol-independent transport. Finally, we experimentally evaluate that proposed service providing method can be applied to transmit the HD-SDI signal in the software-defined optical network.

One of the great challenges nowadays when managing IP networks is to guarantee proper Quality of Service, using network infrastructure on optimized way. One of the proposed solutions is traffic engineering with MPLS. However, the characterization of the demands and of the network state are difficult tasks, considering that the demands and the data traffic are random, consequently, the network state changes dynamically and in a random way. In this work we propose a connection admission controller that uses fuzzy logic based on linguistic rules to treat the inaccurate information in IP over MPLS networks with the purpose of offering Quality of Service to the users. In accordance with the simulation results, we concluded that the use of fuzzy logic allows a large flexibility in the connection admission process and the possibility to include more network and traffic information when making a decision without increasing considerably the controller complexity.

The rapid deployment of technologies that can share data brings, in addition to the positive aspects, also technologies that an attacker can use to misuse personal data. Vulnerabilities in the network can be divided according to the type of attack into horizontal attacks from the perspective of the attacker and vertical attacks from the perspective of the victim. An important goal of organizations is to successfully defend against such an attack. The search for attacks on the network infrastructure can be ensured by implementing machine learning that can capture the current attack. This work describes methods of searching for attacks using graphical representation of data and also using machine learning of the high count type at different sizes of the investigated segments.

The TCP/IP internetworking technology has proven to be the means to build highly scalable, heterogeneous information infrastructures, as exemplified by the global Internet. Originally designed for data transport, the internet is being used increasingly for transport of multimedia information such as voice and video. With the increase in sophistication of desktop computers, and the availability of networked multimedia applications, private IP networks as well as the public Internet is likely to see more of this type of traffic. This type of usage, however, is ad-hoc in that the IP network architecture has been designed for “best-effort” delivery, without any guarantees on throughput or delay. To adequately support real-time traffic flows with bandwidth and delay requirements, the following features are needed: new service classes beyond best-effort to provide certain guarantees on throughput and delay to applications; a mechanism that allows applications to signal to the network their quality of service (QoS) requirements and to reserve network resources to obtain the desired QoS; traffic management mechanisms in hosts and routers, and a routing architecture that allows the dynamic discovery of QoS-accommodating paths in the network for data flows, in the presence of changes in network topology and loading.

Recent studies on end-to-end congestion control in IP networks have demonstrated the difficulties in compromising between stability and fairness objectives. In this paper, we introduce a new congestion avoidance model, which can meet the key criteria of both objectives with enhanced source behavior. Our new scheme provides local system stability which is scalable to arbitrary delays, capacities and network topologies. The tradeoff between system stability and response speed is controlled by single parameter tuning at source. At equilibrium, the obtained resource allocation is weighted proportional fair or weighted TCP friendly fair with appropriate choice of parameter at source. Not like the arrangement in current TCP scheme, in which large round-trip time (RTT) are treated with bias, our resulting weighted allocation scheme favors long delay flows.