Network performance QoS estimation

The question of quality of service in networks of the latest generation is studied. The impact of heterogeneous traffic on the main quality of service indicators reflecting the probability of packet loss and delivery delay over time is evaluated. A model of quality of service in information communication networks has been developed to ensure low traffic delay. Taking into account the concept of quality in information communication networks, the issue of quality of service in networks of the latest generation becomes important from the standpoint of ensuring high reliability and reliability of information transmission, which significantly affects the quality of services provided to the user. This, of course, determines the need to control various parameters of equipment and communication lines, which are determined both by the features of the transmission system and by the parameters of leased channels and tracts. The basis for determining the quality of service, as mentioned above, is the provision of the international concept of quality of service (QoS). According to the requirements of the compliance control model proposed in the concept, the approach to quality indicators is structurally related to the OSI reference model. The main advantages of modern technologies of information communication networks are undoubtedly 5G technologies, which already support such resource-intensive services as massive machine-to-machine communication, ultra-reliable communication and low-latency communication. All of them require high data transfer rates, low millisecond latency, and connectivity of millions of devices. At the same time, it is necessary to ensure a high level of quality of service (QoS), which plays an important role in 5G technologies. The proposed model makes it possible to evaluate not only the quality of service of the combined traffic flow of Internet of Things, tactile Internet, and augmented reality services, but also the delivery delay and the probability of packet loss. The simulation results showed that the parameters of the quality of service of the traffic of the Internet of Things and other modern applications differ significantly during the service of the aggregate flow. When studying the service of aggregated traffic, it was established that the probability of packet loss is higher for it than for Internet of Things traffic. In addition, this difference becomes more significant when the intensity of the incoming load increases.

Trends in teleco mmunications show that customers require still mo re and more bandwidth.If the telecommun ication operators want to be successful, they must invest a lot of money to their infrastructure and they must ensure required quality of service.The telecommun ication operators would devote to development in this area.The art icle deals with quality of service in IP networks.Problems of quality of service can be solved through admission control methods based on measurements.These admission control methods take care of control of inco ming traffic load.New flow can be accepted only if needed quality of service is ensured for it and without quality of service breach causing of already accepted flows.In the article were made description of simu lations and results of simu lations for Voice over IP, constant bit rate and video sources.Simu lations were realized in Net work simu lator 2 environ ment.These simu lations were evaluated on the base of some parameters such as: estimated bandwidth, utilization and loss rate.

Background. The introduction of fifth-generation (5G) networks creates new opportunities for fast and continuous data exchange, but there are still some problems with the quality of voice services in such networks. With the rapid development of technology and the further spread of 5G, there is a need to understand the impact of key aspects of 5G on voice quality. This requires research that can systematically analyse the features of 5G networks that affect the quality of voice services. Objective. Identification of ways to improve the quality of voice services in 5G networks. Assessment of key indicators of voice service quality in 5G networks. Determination of the best option for the gradual transition to the Standalone mode and the use of VoNR technology in the fifth generation networks. Methods. Analysis of factors affecting the quality of voice services in fifth-generation networks. Analysis of well-known publications on the implementation of 5G networks. Comparison of the implementation of Non-Standalone and Standalone modes in the 5G network. Testing of the modern EVS codec, which provides an opportunity to improve the customer experience. Results. Confirmation that 5G networks can significantly improve the quality of voice services compared to previous mobile communication technologies such as 4G and 3G. Certain factors that may affect the quality of voice services and require additional attention when planning and deploying 5G networks are identified. The optimal steps for the transition to the Standalone mode and the use of VoNR technology in fifth-generation networks are determined. The main differences in the QoS architecture between LTE and 5G are identified, and the purpose of DRB flows for separating traffic types and services is established. Conclusions. It has been confirmed that 5G networks can significantly improve the quality of voice services compared to previous technologies such as 4G and 3G. This is possible due to the broadband capabilities of 5G networks, improved data transmission, low latency, the use of an advanced EVS codec and reduced response time. However, certain factors, such as network coverage, optimisation level and traffic characteristics, can affect the quality of voice services and require additional attention when planning and deploying 5G networks. The QoS management architecture consists of QoS flows, which allow separating packet assignment to flows (managed by the CN) from the assignment of DRB flows (managed by the RAN). As 5G networks are being rolled out gradually, it is important to properly integrate the 5G domain into the existing telecoms provider's network. The transition from Non-Standalone to Dual connectivity is a necessary step for the implementation of VoNR technology in Standalone mode. Using the modern EVS codec allows not only improving the customer experience, but also introducing new voice services.

Years of research on QoS architectures for IP networks have delivered sophisticated proposals, which have nevertheless not found broad commercial use. The reasons are not lack of technical soundness or insurmountable technological complexity, but insufficient attention to other, non-QoS-specific matters. First among them is the lack of a commercialization model for the Internet together with the necessary accounting and charging architecture. Another crucial issue is the assurance of end-to-end QoS coherence in the face of multiple intervening parties (network and content providers, users). Furthermore, the practical requirements imposed by those parties to any successful QoS architecture have not been fully taken into account: Ease of management, simplicity and measurable guarantees are some of the main ones. In this paper, the overall constraints on and conditions for the successful deployment of QoS in IP networks are analyzed and some possible directions explored.

Following the success of the Internet, over the last five years or so IP has firmly established itself as the networking protocol of choice for a wide range of both traditional and emerging applications. More recently, among business users, there has been much interest in adding real-time applications, including voice over IP (VoIP) and interactive videoconferencing. A major driver for this is the desire to consolidate all applications on to a single, multi-service platform. This chapter focuses specifically on the DiffServ approach to QoS. DiffServ involves the segregation of traffic into a small number of classes, but unlike either the ATM QoS approach, or the IETF IntServ approach, there is no signalling control plane to look at end-to-end behaviour. Instead the approach relies upon per-class capacity planning at each router individually (the so-called per hop mechanism) in order to ensure that, overall, each class gets the standard of service required.

The Differentiated Services architecture has been proposed by IETF to implement QoS in IP networks. However, research shows that there exists great unfairness in the DiffServ network, especially between responsive and unresponsive aggregate traffic. This paper argues that responsive and unresponsive flows should be treated differently only to their out-of-profile traffic, in addition, traffic responsiveness should be identified by its behavior rather than simply by protocol. Then this paper defines the TCP-friendly fairness in the DiffServ network by introducing the idea of TCP friendliness into the DiffServ network to guarantee the fairness between responsive and unresponsive aggregate flows. Moreover, two markers, srTCM and trTCM, with eight mapping schemes, are examined by elaborate simulations, however, none could achieve the TCP-friendly fairness. Therefore, this paper propose the direct congestion control scheme (DCCS) to achieve fairness between responsive and unresponsive aggregate flows.

Session Initiation Protocol (SIP) is a session signaling protocol designed to establish multimedia sessions (voice, video and data). Meanwhile multimedia applications are sensitive to the Quality of Service (QoS) provided by the underlying network. Therefore SIP could manage QoS for multimedia sessions by interacting with QoS mechanisms deployed in IP networks, and especially in DiffServ domain. In this paper, we present QoS management mechanisms to be deployed by SIP over an IP-MPLS network. In fact, we use SIP to achieve QoS on a session basis, in which the over all activity of the user during the session is considered. We show the implementation of our mechanisms as well as an experimental evaluation for different kind of sessions.

This paper presents the design of the automated resource management tool (ARMT), a software-based network management tool that automates the potentially complex manual task of global resource management in a mission-oriented, intranet environment. It allows the network operator to specify high level policies defining the network quality of service (QoS) that must be offered to the different applications sharing the IP network, under each operating mode of the network. The tool automatically handles resource conflicts in accordance with the high-level QoS policies defined by the network operator. It also monitors the network to detect events that may lead to deviations from the specified service policy and automatically implements proactive actions to ensure that the network delivers the required QoS. ARMT implements an innovative policy-driven QoS control technique which ensures that multimedia intranet applications will have an adaptive environment that permits them to coexist with other mission applications. The paper presents our experience with the implementation of a rapid prototype of the ARMT in a laboratory testbed environment.

It is estimated that the Internet currently connects millions of hosts and users worldwide, and the number will continue to increase exponentially for years to come [1]. Such a huge popularity of the Internet has enabled the widespread use of IP networks everywhere. As real-time, multimedia applications as well as new emerging applications demanding stringent quality of service (QoS) becomes common in the Internet, a single best-effort service class is no longer adequate in delivering predictable and bounded QoS performance. This drives an urgent need to provide some kind of QoS in the IP network. In addition, as IP networks are rapidly attracting pervasive commercial uses from competitive business and cooperates who are willing to pay more substantially higher price to get the best possible service level, the framework for QoS guarantee and differentiation become increasingly more important than ever. We consider offline versions of path provisioning and path protection problems for MPLS networks. The objective is to split the traffic of a broken working path to multi back up protection paths. We consider label switch path (LSP)- based protection where, in case of a single LSP failure, all the traffic utilizing the failed link can be rerouted to a pre computed set of paths.

Currently the most relevant problem is how to improve the quality of service in IP networks. In order to secure proper quality of service, the network must react adequately to the service generated load, intended to the network and contribute proper amount of resources. In this paper we analyze the quality assurance of multiple services in IP network using differentiated services. We introduce the analytical differentiated flow service model in the router and propose assumptions to evaluate the different queuing disciplines in this paper. In the differentiated services network the classification particularity of complex flow is reflected by the management of queues. Therefore the imitation model was constructed which allows to evaluate different disciplines which serve queues for three priorities flows. There was estimated the intensity effect of different class services to analyzing classes of flows delays using FIFO, PQ, CBQ and WFQ disciplines. Ill. 14, bibl. 7 (in Lithuanian; summary in Lithuanian, English, Russian).

Wireless Sensor Networks (WSNs) comprise of groups of tiny sensor nodes that are deployed for collaborative missions such as environmental monitoring, target tracking and surveillance. Due to the miniature size of the nodes, they are typically deployed in large numbers and communicate via multiple hops through a wireless shared communication channel. The successful implementation of such networks is dependent on the enabling technologies (such as digital electronics and wireless communications), as well as the provisioning of Quality of Service (QoS) in the network. While there have been many efforts in QoS provisioning in traditional networks such as the Internet and Mobile Ad Hoc Networks (MANETs), these networks have very different characteristics from that of WSNs. Consequently, the QoS models and protocols that have been designed for the Internet and MANETs cannot be directly applied to WSNs. In this paper, we look at some of the existing QoS mechanisms in the networking literature, and the inherent characteristics of WSNs which make it challenging to provision for QoS in the network. We then identify some key performance metrics for WSN QoS and outline some mechanisms to achieve QoS in the sensor network. Finally, we propose WISER - a framework to enhance QoS in WIreless SEnsoR networks.

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

In an era of rapid urbanization, the quality of network services in urban environments is becoming increasingly important. This article explores efforts to improve network service quality in several sampang cities through an evaluation of Quality of Service (QoS) and user perceptions of Quality of Experience (QoE). The research is based on a survey conducted at various locations in sampang city to collect data on the reliability, speed, and availability of network services. The survey involved a diverse sample of users, covering a wide range of ages, professions, and levels of technology usage. In addition, we also analyzed users' perceptions of service quality based on their experiences of using the network in an urban environment. The results highlighted several key challenges in improving QoS, such as user density, signal interference, and network infrastructure limitations. However, we also found that strategies such as the use of advanced network technologies, such as 5G and beamforming, as well as traffic prioritization based on application type, can help to significantly improve QoS. In addition, we found a strong positive correlation between improved QoS and improved QoE, suggesting that improved quality of service has a direct impact on improving user experience. These findings provide valuable insights for network operators and policy makers in their efforts to improve network services in dense urban environments. The research also recommends further investment in network technologies and the development of policies that support more effective management of network traffic to ensure optimal quality of service.

Admission control in Flow-Aware Networking (FAN) architectures under GridFTP traffic

Internet traffic engineering using multi-protocol label switching (MPLS)