Network Performance Analysis Using Cisco VIRL

Routing is an important process in computer networks, which involves selecting a path for packet data. The routing mechanism is governed by protocols at layer three of the OSI model or layer two of the TCP/IP model. Routing can be static or dynamic, with dynamic routing algorithms automatically determining the best path in the network. One commonly used dynamic routing protocol is Open Shortest Path First (OSPF), which automatically selects the optimal route for data packets based on received network topology information. The need for more advanced and scalable routing solutions led to the development of Multi-Protocol Label Switching (MPLS), introducing the concept of label switching for faster and more efficient data forwarding. This research aims to analyze the performance of OSPF and OSPF MPLS routing scenarios regarding TCP and UDP services. Through evaluation of Quality of Service (QoS) parameters, including throughput, packet loss, delay, and jitter. This study simulates the network using GNS3 software, with TCP and UDP services transmitted for durations of 15, 30, and 60 seconds. The results showed performance for both scenarios with an average index value of 3.75, indicating "good" network performance according to TIPHON standards across all measurement sessions. Although MPLS OSPF showed slightly lower throughput compared to non-MPLS OSPF, both scenarios showed very low packet loss, low delay, and stable jitter. The evaluation also highlights that the use of OSPF MPLS for UDP services reduces the time delay. Both non-MPLS OSPF and MPLS OSPF effectively support TCP and UDP services, with MPLS OSPF showing advantages in certain QoS parameters, especially for UDP services.

Demand for mobile data services and people’s dependence on 4G LTE networks continue to increase. However, the quality of service (QoS) of this network still requires improvement, especially regarding the effect of QoS on throughput at specific frequencies. The research gap lies in the lack of indepth analysis of the impact of QoS parameters on network performance at frequencies of 2,100 MHz and 2,300 MHz. This study evaluates the effect of QoS parameters, such as delay, jitter, and packet loss, on throughput in 4G LTE networks at both frequencies. The research methodology uses an experimental approach with throughput, delay, jitter, and packet loss measurements in various network conditions. The results showed that delay (17.2174 ms to 37.0322 ms), jitter, and packet loss significantly influence throughput, ranging from 624.5 Kbps to 1,322.4 Kbps. The 2,100 MHz frequency tends to show better performance than 2,300 MHz. This study concludes that optimizing QoS parameters, especially delay and jitter, can significantly improve 4G LTE network performance. These findings provide practical contributions for mobile operators in improving network quality and customer satisfaction and open opportunities for further research on other frequencies or newer network technologies.

Objective: The main objective of the work is to enhance the routing performance of the network by increasing the quality of service parameters. Increasing the routing performance will lead to increase in the network performance and the efficiency of the existing network. So an algorithm is designed which lead to the higher performance of the network. Method Analysis: In this an algorithm is proposed which is designed on the basis of the residual power of the nodes and the other resources available to the nodes. Using the residual power and other resources information the nodes tries to manage the packet loss in the network by routing the packet through another available path in the network. Findings: The proposed algorithm is implemented using the AODV and AOMDV routing protocol. The AODV and AOMDV routing protocols are analyzed on the basis of various performance metrics and the problem of route failure which leads to packet loss in the network and the increase in the overheads in the network is demolished to some extent. This will lead to better performance of the network and more reliable communication in the environment. There is an incredible increase in the performance metrics of the network which will ultimately leads to network with higher efficiency rate. Improvement: The proposed methodology is an advance version of the previous existing work and there by providing with better results and network scenario which will have higher quality of service parameters. Keywords: AOMDV (Ad-hoc on demand multipath distance vector routing protocol), AODV (Ad-hoc on demand distance vector), EOM (efficient optimization module), Path Recovery, Residual Power

Adaptive quality management schemes in Voice over IP (VoIP) Network have a constructive impact on the improvement of voice quality level received by end-users. These schemes are used for persistent monitoring of quality level leading to better management of network services. In this paper, a model is introduced to improve the voice quality level and optimize resource utilization. In the proposed model, the VoIP network makes the decision for configuration adaptively and modifies itself by means of Hidden Markov Model (HMM). Depending on the quality level and consumed bandwidth evaluated in the receiver side this adaptive self-modification takes place. During this procedure, acceptable weights for Quality and Quality of Service (QoS) parameters, namely, End-to-End Delay, Packet Loss, and Jitter are determined. These acquired weights help HMM model predict next optimum state properly. Another important objective of this work is to find a relation between Quality of Experience (in terms of Mean Opinion Score) and Quality of Service (in terms of End-to-End Delay and Jitter) parameters. This mapping provides a tangible understanding of how QoS parameters affect MOS. Simulation results demonstrate that the proposed technique lessens the impact of jitter and packet loss on voice quality level.

This study presents a novel comparative analysis employing Random Forest regression to quantify the relative importance of key Quality of Service (QoS) parameters—packet loss, delay, and jitter—in Multiprotocol Label Switching (MPLS) and Software-Defined Wide Area Network (SD-WAN) architectures. Using empirical data collected from controlled simulations of multimedia traffic, the feature importance scores reveal that packet loss overwhelmingly dominates as the critical factor influencing network performance, with scores of 0.8620 in SD-WAN and 0.7259 in MPLS, indicating an 18.76% increase in SD-WAN’s sensitivity to packet loss. Delay exhibits moderate relevance in MPLS, with an importance score of 0.2205, but shows markedly reduced significance in SD-WAN at 0.1341 (a 39.21% decrease). At the same time, jitter demonstrates negligible influence across both networks, with scores below 0.054. These findings confirm that SD-WAN’s dynamic path optimisation effectively mitigates delay effects, whereas packet loss remains the principal constraint on performance. This work constitutes the first methodical Random Forest-based comparative evaluation of QoS parameter importance across MPLS and SD-WAN, delivering robust, data-driven insights tailored to each architecture’s operational characteristics. The framework provides network operators with critical guidance for targeted QoS optimisation, prioritising packet loss mitigation strategies, particularly within SD-WAN environments. Overall, this research establishes an empirical foundation for architecture specific QoS management, advancing intelligent network performance assessment through machine learning techniques.

In this paper, we propose a new approach for video quality control for multimedia networks. Our new approach is based on video quality measure that combines both the network quality of service (QoS) as well as the user quality of experience (QoE). The proposed approach improves the end-to-end traditional video quality control for multimedia network by including the human perception of video data, which is major concern for the video client, along with the network quality of service (QoS) measurements. We will show that the proposed QoS-QoE based video quality control algorithm can reflect both the condition of the network environment and the human perception of the received networked video data stream. According to both QoS and QoE parameters, rather than using only QoS parameter, video quality control action will satisfy the user needs more than relying only on the network conditions. Since our proposed QoE parameter Self-Reference Complex Wavelet Video Structural Similarity Index (SRCW-VSSIM) can be obtained with no reference (NF) video data, it satisfies the requirement of real-time video transmission.

Problem statement: The demand for multimedia applications in WiMAX networks is growing at a rapid pace. A method for guaranteeing Quality of Service (QoS) for different classes of traffic is therefore gaining importance. Hence desi gning and analyzing multimedia traffic and QoS parameters has become central to this problem. In t his study, we propose a cross layer frame work in which a coordination between the Medium Access Control layer (MAC) and the Physical layer (PHY) is established for guaranteeing the QoS requirement in a multiclass traffic WiMAX environment. Two traffic classes' real time Polling Service (rtPS) a nd non real time Polling Service (nrtPS) are considered for analyzing the performance. Approach: The objective of study is to guarantee QoS for multiple service class traffic in a multiple connec tion environment in WiMAX network. A cross layer design approach is used for this purpose. A priorit y scheduler at the MAC layer schedules the traffic based on channel state information. The Adaptive modulation and coding scheme is used at the physical layer that adapts to the scheduled traffic to stabilize the QoS requirements of different tra ffic classes. Results: The Priority value is estimated using the Friis eq uation that calculates the received power and determines the SNR. The average throughput, average bytes received and the packet loss are plotted against time. This indicated that these QoS parameters are stable over a period of time. Conclusion: A cross layer frame work was developed based on a scheduler for QoS stability that uses CSI at the MAC layer and an AMC at the PHY layer in WiMax Networks. The scheduler enjoyed flexibility and scalability, whose performance was evaluated against existing systems through simulati on.

Software-Defined Networking (SDN) is a new efficiently idea of programmable networks that separates the control plane from data plane of all network devices. Internet service provider is responsible for all the control decisions and communication among the forwarding elements from centralized controller. SDN provides the various optimized services. Quality of service (QoS) routing is a path computation method that is suitable for the different traffics generated by several applications, while utilization of network resources has increased. This agreement of service is defined by QoS requirements such as throughput, delay, jitter and packet loss etc. Multimedia applications often require assured from multi QoS constrained, causing the NP-complete problem which cannot be simply solved in polynomial time and high management complexity in the transition network. SDN is able to reduce complexity and it is used to efficiently implement traffic, hence SDN significantly values to development QoS routing. In this paper, we propose QoS routing algorithm called Quantized Level Balance (QLB) for SDN that considers one or many QoS parameters relating to the network application. To satisfy the requirements, QLB selects QoS parameters depending to the level of appropriate application service quality. We have replicated our algorithm on simulate topology with Scalable Video-streaming Evaluation Framework (SVEF). We measure the Peak Signal-to-Noise Ratio (PSNR) and Mean Opinion Score (MOS) of Scalable Video Coding (SVC) at the receiver. Our propose algorithm is improved than single-metric approach that may choose poor QoS parameter paths.

Accurate modeling of VoIP traffic QoS parameters in current and future networks with multifractal and Markov models

Information technology is developing rapidly, especially in the field of computer science, telecommunications and information systems. In the early days of telecommunications, voice networks could not be combined with data networks. However, more recently it became possible to integrate data, audio and video services onto a single network, known as a multimedia network. The capabilities of the Internet are also rapidly increasing, and now, in addition to data packets (email, web browsing) the Internet transfers audio and video packets. Voice over Internet Protocol (VoIP) is a multimedia service. This technology can be used to transmit audio, video, and data packets. Consequently, it is becoming the most preferred communications technology. VoIP will eventually replace the use of traditional telephony – Public Switched Telephone Networks (PSTN), although the switchover process is challenging. A chief concern it that, while PSTN transmits audio packets over a closed network, VoIP sends audio packets using an open network – the Internet. In PSTN, eavesdroppers must have direct access to physical network to obtain information from communications. Whereas with VoIP, eavesdroppers can monitor data packets from they are connected to the Internet. Hence, security and privacy are important considerations when switching to VoIP systems. One solution is anonymous systems, which can be used to implement security and privacy protocols. However, this typically reduces the Quality of Service (QoS) of the VoIP. This empirical research study examines VoIP performance in an anonymous network – The Onion Routing (Tor). Two scenarios are implemented using the real Tor network to investigate three QoS metrics for VoIP: latency, jitter and packet loss. As recommended, latency in VoIP should be less than 400 ms, jitter should be less than 50 ms and packet loss should not be more than 5%. In addition, the research calculates the probability of attackers in the two scenarios implemented and evaluate Tor network forecasting. Experiments were conducted in reference to two scenarios. The first scenario is VoIP calls routed through a Tor network with three Tor relays (default Tor). And the second scenario is VoIP calls routed through a Tor network with two Tor relays. Experiments were performed in three periods; December 2012, July 2013, and October 2013. During each experimental time period, a hundred calls were captured for each scenario. Experimental results show that the QoS of VoIP over the two Tor relays was better than the VoIP over the three Tor relays. However, a probability of attackers calculation found that the VoIP with three Tor relays returned better anonymity than that with two Tor relays. The best results were returned over the experimental period in July 2013, when the acceptable calls using two Tor relays reached 64 calls at 5% packet loss. Meanwhile, the worst experimental results were returned in October 2013, when acceptable calls numbered 11 using three Tor relays at 1% packet loss. At the end of 2013, the actual data for relay numbers and bandwidth approached forecast result with time series analysis. Meanwhile, the numbers of Tor users differed from the Tor users forecast. In August 2013, Tor users increased dramatically; this is attributed to the fact that BotNet attack was using the Tor network to attack their target leading to a fivefold increase.

In this paper, we analyze the jitter and packet loss behavior of voice over Internet protocol (VoIP) traffic by means of networks measurements and simulations results. As result of these analyses, we provide a detailed characterization and accurate modeling of these Quality of Service (QoS) parameters. Our studies have revealed that VoIP jitter can be modeled by selfsimilar and multifractal models. We present a methodology for simulating packet loss. Besides, we found relationships between Hurst parameter (H) with packet loss rate (PLR).

Bandwidth load balancing is a critical technique for improving the quality of service (QoS) in local area networks (LANs). Insufficient load balancing and QoS utilization can lead to network congestion, reduced performance, latency, packet loss, and an overall degradation of network performance. This paper presents an overview of load balancing techniques, load balancing algorithms, QoS parameters, load balancing strategies, and considerations for calculating bandwidth. The challenges of implementing load balancing algorithms and preferences for their implementation are discussed. Accurate network measurements and monitoring, coordination between network devices, security and privacy considerations, and careful evaluation of load balancing algorithms are among the challenges faced. The paper also discusses optimization and monitoring of bandwidth use, including options such as increasing bandwidth, bandwidth throttling, and data transfer throttling. The findings presented in this paper provide insights into the importance of load balancing and QoS in LANs and offer guidance for addressing related challenges.

The concept of military operations introduced in 1990 is centered on a network for sharing information that provides an essential element for military missions around the world. Communication and network technology has undergone several significant changes in the utilization of an application service. Building practical communication technology is increasingly being considered by military organizations in which, mobile communication is highly prioritized at the center of tactical military operations. This study aims to develop a SIP mobile application that will be applied according to the needs of military operations to provide information needed by squad commanders, making it easier to develop a strategy in forest battles. The ability of SIP or Session Initiation Protocol is very appropriate to be used as communication between military forces in forest area battle areas. In addition, the application that is built has been tested using Quality of Service parameters, including delay, jitter and packet loss. In this study the data analysis method used quantitative methods using statistical tests. The results of the Quality of Service according to ITU-T G.114 standards measurement on the SIP Mobile application in Forest Areas it can be seen that the delay was 8,56ms, the jitter was 7,29ms and the packet loss was 0,02%. Thus it can be concluded that the network quality according to TIPHON standards was found in Good quality. It is expected that the utilization of SIP mobile application can support communication of military personnel, especially in combat conditions in forest areas without any internet network.

The digitization of user services has increased, in line with the need for VPS and cloud computing services, which are rampant among application and platform developers. Quite a few companies that create applications or application users have servers to handle user needs. Testing is carried out using the ICMP Protocol to get real-time results and can be measured. From Scenario 1, carrying out 20 test requests, we get a packet loss of 5% with RTT Avarage of 195,838ms and Mdev 4,103ms. If you apply DNS failover in scenario 2, the client will likely access the web a little slower, as evidenced by the packet loss being 25% greater in value. Compared to scenario 1, having a high standard deviation (Mdev) of round-trip times is not desirable. This variation is also known as jitter. Increased jitter can cause a bad user experience, especially in real-time audio and video streaming applications. However, this is still understandable because it only has a 1-5 second effect on the service. Next, in scenario 3, we can see that private and public servers have relatively high gab with 0% packet loss, which has a small Mdev value of 0.309ms. Therefore, the DNS failover method is a solution for network administrators who have problems related to server migration between public servers and private servers so that services can run even if a server is maintaining or downlinking.

Under the current TCP/IP architecture, all TCP traffics compete for network resources completely fairly, which makes it difficult to satisfy applications' versatile communication requirements. This paper presents an improved TCP congestion control mechanism where the congestion window becomes ω(1 - b)W rather than (1 - b)W for every window W containing a packet loss. Theoretical analysis and simulation results show that it can be easily implemented with less additional overhead and can easily perform network resource allocation by weighted parameter ω for traffics under the similar communication environments, which can efficiently lead to guaranteed relative quality of services and improve network performances.