This document summarizes the results of a lab simulation comparing the performance of G.711, G.729, and G.723 codecs. The simulation measured FTP and email download response times, voice traffic received, end-to-end delay, and packet delay variation. Based on the results, the G.729 codec had the best overall performance with minimal impact on data traffic, acceptable delay and jitter ranges, and significantly lower network bandwidth usage compared to G.711. Though G.723 had the lowest bandwidth, its delays were unacceptable. Therefore, the document recommends implementing G.729 for voice traffic.
This document summarizes the steps taken in a Wireshark lab to analyze VoIP call setup and media transfer using SIP and RTP protocols. The student is instructed to examine SIP messages like INVITE, 180 Ringing, 200 OK and ACK to establish a call, and RTP and RTCP packets to observe audio payload and call quality metrics. Screenshots are included to show the analyzed packet details.
The document discusses circuit switching vs packet switching networks, with circuit switching reserving bandwidth for constant real-time communication but being less efficient than packet switching which allocates bandwidth on demand. It also describes how botnets can compromise many devices to perform distributed denial of service attacks by recruiting devices to target servers.
This document analyzes the RC4 encryption algorithm and examines how its performance is affected by changing parameters like encryption key length and file size. Experimental tests were conducted to measure encryption time for different key lengths and file types. The results show encryption time increases with longer keys and larger files, and are modeled mathematically. The document also provides background on encryption methods, how RC4 works, and compares stream and block ciphers.
Investigation of dhcp packets using wiresharkjpratt59
This document summarizes research investigating DHCP packets using the network analysis tool Wireshark. The researchers captured DHCP packets between a client and server to analyze the contents and parameters exchanged. There are four DHCP packets exchanged: DHCPDISCOVER, DHCPOFFER, DHCPREQUEST, and DHCPACK. The researchers used Wireshark to investigate each packet type in detail, observing the information carried like transaction IDs, IP addresses, and MAC addresses. The goal was to understand the complex process of how a DHCP client and server communicate to automatically assign an IP address.
I am Norman H. I am a Computer Networking Assignment Expert at computernetworkassignmenthelp.com. I hold a Master's in Computer Science from, McMaster University, Canada. I have been helping students with their assignments for the past 15 years. I solve assignments related to Computer Networking.
Visit computernetworkassignmenthelp.com or email support@computernetworkassignmenthelp.com.
You can also call on +1 678 648 4277 for any assistance with Computer Networking Assignment.
A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet ...IJECEIAES
This document summarizes a proposed statistical approach to adaptive playout scheduling in Voice over Internet Protocol (VoIP) communication. The approach estimates the optimal buffer delay for each packet based on network statistics, packet loss rate, and buffer availability. It uses a window-based method to track recent network conditions and estimate delay for the current packet. Buffer delay is calculated based on estimated jitter, a delay factor accounting for surrounding packet arrival, and late packet loss rate. Experimental results show this approach allocates buffer delay with the lowest late packet loss rate compared to other algorithms.
This document summarizes the steps taken in a Wireshark lab to analyze VoIP call setup and media transfer using SIP and RTP protocols. The student is instructed to examine SIP messages like INVITE, 180 Ringing, 200 OK and ACK to establish a call, and RTP and RTCP packets to observe audio payload and call quality metrics. Screenshots are included to show the analyzed packet details.
The document discusses circuit switching vs packet switching networks, with circuit switching reserving bandwidth for constant real-time communication but being less efficient than packet switching which allocates bandwidth on demand. It also describes how botnets can compromise many devices to perform distributed denial of service attacks by recruiting devices to target servers.
This document analyzes the RC4 encryption algorithm and examines how its performance is affected by changing parameters like encryption key length and file size. Experimental tests were conducted to measure encryption time for different key lengths and file types. The results show encryption time increases with longer keys and larger files, and are modeled mathematically. The document also provides background on encryption methods, how RC4 works, and compares stream and block ciphers.
Investigation of dhcp packets using wiresharkjpratt59
This document summarizes research investigating DHCP packets using the network analysis tool Wireshark. The researchers captured DHCP packets between a client and server to analyze the contents and parameters exchanged. There are four DHCP packets exchanged: DHCPDISCOVER, DHCPOFFER, DHCPREQUEST, and DHCPACK. The researchers used Wireshark to investigate each packet type in detail, observing the information carried like transaction IDs, IP addresses, and MAC addresses. The goal was to understand the complex process of how a DHCP client and server communicate to automatically assign an IP address.
I am Norman H. I am a Computer Networking Assignment Expert at computernetworkassignmenthelp.com. I hold a Master's in Computer Science from, McMaster University, Canada. I have been helping students with their assignments for the past 15 years. I solve assignments related to Computer Networking.
Visit computernetworkassignmenthelp.com or email support@computernetworkassignmenthelp.com.
You can also call on +1 678 648 4277 for any assistance with Computer Networking Assignment.
A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet ...IJECEIAES
This document summarizes a proposed statistical approach to adaptive playout scheduling in Voice over Internet Protocol (VoIP) communication. The approach estimates the optimal buffer delay for each packet based on network statistics, packet loss rate, and buffer availability. It uses a window-based method to track recent network conditions and estimate delay for the current packet. Buffer delay is calculated based on estimated jitter, a delay factor accounting for surrounding packet arrival, and late packet loss rate. Experimental results show this approach allocates buffer delay with the lowest late packet loss rate compared to other algorithms.
Toward an Understanding of the Processing Delay of Peer-to-Peer Relay NodesAcademia Sinica
Peer-to-peer relaying is commonly used in realtime applications to cope with NAT and firewall restrictions and provide better quality network paths. As relaying is not natively supported by the Internet, it is usually implemented at the application layer. Also, in a modern operating system, the processor is shared, so the receive-process-forward process for each relay packet may take a considerable amount of time if the host is busy handling some other tasks. Thus, if we happen to select a loaded relay node, the relaying may introduce significant delays to the packet transmission time and even degrade the application performance.
In this work, based on an extensive set of Internet traces, we pursue an understanding of the processing delays incurred at relay nodes and their impact on the application performance. Our contribution is three-fold: 1) we propose a methodology for measuring the processing delays at any relay node on the Internet; 2) we characterize the workload patterns of a variety of Internet relay nodes; and 3) we show that, serious VoIP quality degradation may occur due to relay processing, thus we have to monitor the processing delays of a relay node continuously to prevent the application performance from being degraded.
IMPROVING IPV6 ADDRESSING TYPES AND SIZEIJCNCJournal
This document discusses proposed modifications to IPv6 addressing types and address size. It suggests that multicast addressing can mimic anycast and limited broadcast addressing, making those types unnecessary. It also proposes reducing the IPv6 address size from 128-bits to decrease packet overhead, while ensuring the new size supports future internet growth. A formula is presented to predict IP address exhaustion dates for different address sizes based on current usage and population projections.
This document contains block diagrams and specifications for a Quanta Computer Inc. motherboard project. It includes:
1) A block diagram of the system showing the main components like the Ivy Bridge processor, Panther Point PCH, memory, graphics, SATA, USB, networking ports.
2) Specifications for the power delivery including voltages supplied to different components and the power states of those components.
3) A more detailed block diagram of the Ivy Bridge processor showing the connections to the PCH and graphics via DMI and PCIe interfaces.
4) Signal specifications and design notes for components like the processor, display ports, and graphics compensation.
The document describes DHCP snooping Option 82 configuration examples. It provides an overview of Option 82, which allows DHCP servers to assign IP addresses based on location information added to client requests. It then gives an example configuration where a DHCP snooping device adds unique Option 82 information to requests from three client groups, allowing the DHCP server to assign each group a separate address range from the 192.168.10.0/24 network. The DHCP snooping device and server are configured to implement this address assignment based on Option 82.
1. UDP is used for voice and video traffic instead of TCP because TCP introduces delays that break data streams and UDP does not have mechanisms for retransmitting lost packets. TCP is preferred for transmitting data files because it is more reliable and requires lost packets to be retransmitted.
2. To facilitate secure intranet access for remote workers, an enterprise IT department would use VPN and user authentication.
3. The purpose of the Cisco Enterprise Architecture is to divide the network into functional components while still maintaining the concept of Core, Distribution, and Access Layers.
This document discusses local area network (LAN) technologies, with a focus on Ethernet. It outlines the following objectives:
- Briefly discuss dominant wired LANs including Ethernet and other media types.
- Describe Media Access Control (MAC) and Carrier Sense Multiple Access with Collision Detection (CSMA/CD).
- Explain the Address Resolution Protocol (ARP) and bridges.
- Discuss switched Ethernet and virtual LANs (VLANs).
The document then provides details on Ethernet frames, MAC addresses, CSMA/CD, cabling standards and specifications.
The document discusses bandwidth requirements for IP telephony solutions in Bangladesh. It examines factors that influence bandwidth needs like codecs, sample rates, packet overhead. The G.729A codec requires the least bandwidth at 16 kbps per call. To serve a 50 seat call center would require a 800 kbps connection. The government needs to plan adequate bandwidth to support growing IP telephony demands.
This document provides an overview of a networking lesson that teaches participants how to analyze and troubleshoot common IP and ARP problems using Wireshark. The lesson covers ARP and IP addressing, fragmentation issues, routing problems, duplicate IP addresses, and DHCP configuration errors. Specific troubleshooting techniques are demonstrated, such as using ARP requests to diagnose connectivity problems, analyzing packet captures to find fragmented packets, and identifying duplicate IP addresses through gratuitous ARP messages. The overall goal is for participants to gain skills in isolating and resolving basic IP and ARP issues using network analysis tools like Wireshark.
THE FIGHT AGAINST IP SPOOFING ATTACKS: NETWORK INGRESS FILTERING VERSUS FIRST...ijsptm
The IP(Internet Protocol) spoofing is a technique that consists in replacing the IP address of the sender by
another sender’s address. This technique allows the attacker to send a message without being intercepted
by the firewall. The most used method to deal with such attacks is the technique called "Network Ingress
Filtering". This technique has been used, initially, forIPv4 networks, but its principles, are currently
extended toIPv6 networks.Unfortunately, it has some limitations, the main is its accuracy. To improve
safety conditions, we applied the "First-Come First-Serve (FCFS)" technique, applied for IPV6 networks,
and developed by the "Internet Engineering Task Force (IETF)" within its working group "Source Address
Validation Improvements (SAVI)", which is currently being standardization. In this paper, we remember
the course of an attack by IP Spoofing and expose the threats it entails.Then, we explain the "Network
Ingress Filtering" technique. Next, We present the FCFS SAVI method and methodology that we have
adopted for its implementation.Finally, we, followingthe results, discuss and compare the advantages,
disadvantages andlimitations of the FCFSSAVI methodto thoseknown in the "Network Ingress Filtering"
technique. FCFS SAVI method is more effective than the technique of "Network Ingress Filtering", but
requires some improvements, for dealing with limitations it presents.
7.2.1.8 lab using wireshark to observe the tcp 3-way handshakegabriel morillo
The PC host initiates a TCP three-way handshake with the Google web server to establish a connection. The PC's IP address is its source address, and its MAC address can be found using ipconfig. Frame 15 shows the DNS query from the PC to resolve www.google.com. Frame 16 is the response from the DNS server with the IP address. Frame 17 shows the flags set in the initial request from the PC to start a session. The source port is classified as ephemeral and the destination port is classified as well-known.
Leonardo Nve Egea - Playing in a Satellite Environment 1.2Jim Geovedi
This document discusses techniques for intercepting unencrypted satellite communications. It begins by providing background on satellite types and orbits, as well as common transmission standards. It then describes how to capture satellite signals using a DVB card and Linux tools. Specific techniques covered include identifying packet IDs to create virtual interfaces, DNS spoofing, TCP hijacking, and attacking GRE tunnels. The document explains how these techniques could allow intercepting passwords, cookies, emails and other sensitive transmitted data. It emphasizes that while uplink data cannot be captured from satellites, attacking protocols like GRE could enable some uplink sniffing.
Abstract: The Dynamic Host Configuration protocol (DHCP) is a protocol that is designed to help in automate the process of IP configuration and the rest of network parameters to the host in the network. The DHCP has a unique and important features which are make its address administration very efficient especially nowadays with the proliferation of mobile devices with the patterns that have a transient network access. With a large network or with a mobile ad-hoc network, the administrator will face an impossibility to configure the IP and the rest of network parameters of the host in the network because there will be many wrongs or there will be no infra-structure.
One of the most important features of DHCP is that the same IP will not be allowed to be used at the same time between two hosts or network cards in DHCP mechanism.
The misconfigurations or misbehavior of the host will prevent the DHCP to work properly. Our focus in this paper is to discuss the address administration of DHCP over performance and vulnerabilities in operational networks today. Moreover, we will try to display how the misconfigurations of the host could affect the DHCP and how we will be able to get rid or reduce these misconfigurations.
This document summarizes the study of parameters that determine the quality of service of various Voice over IP (VoIP) clients. The study measured parameters like bandwidth requirement, delay, packet size and observed how clients behaved under different network conditions. Key findings were that bandwidth, jitter, latency and packet loss most affected quality of service. The VoIP clients tested included Google Talk, Skype, VQube, Windows Live Messenger and Yahoo Voice Messenger. Network Address Translation (NAT) types and Simple Traversal of UDP through NAT (STUN) were also explained.
The document discusses different types of Ethernet cables including straight-through cables, crossover cables, and rollover cables. Straight-through cables connect pins on one end of the cable to the same pins on the other end, allowing connection between a computer and a switch, hub, or another computer. Crossover cables have crossed wire pairs and are used to connect like devices such as two computers or two routers. Rollover cables have opposite wiring on each end and connect a device to a router or switch's console port for programming. The document also discusses IP addressing and subnetting concepts.
This was a talk, largely on Kamaelia & its original context given at a Free Streaming Workshop in Florence, Italy in Summer 2004. Many of the core
concepts still hold valid in Kamaelia today
SmartShare's Dynamic Quality of Service (QoS) technology automatically prioritizes time-sensitive traffic like VoIP and video conferencing to ensure flawless performance. It analyzes traffic patterns continuously and allocates extra bandwidth within fractions of a second without needing configuration. Benchmarks showed SmartShare significantly outperformed a Cisco router on a congested ADSL connection, with zero packet loss for VoIP versus 5.34% loss on Cisco, and average packet delay of 119ms versus 782ms on Cisco.
Toward an Understanding of the Processing Delay of Peer-to-Peer Relay NodesAcademia Sinica
Peer-to-peer relaying is commonly used in realtime applications to cope with NAT and firewall restrictions and provide better quality network paths. As relaying is not natively supported by the Internet, it is usually implemented at the application layer. Also, in a modern operating system, the processor is shared, so the receive-process-forward process for each relay packet may take a considerable amount of time if the host is busy handling some other tasks. Thus, if we happen to select a loaded relay node, the relaying may introduce significant delays to the packet transmission time and even degrade the application performance.
In this work, based on an extensive set of Internet traces, we pursue an understanding of the processing delays incurred at relay nodes and their impact on the application performance. Our contribution is three-fold: 1) we propose a methodology for measuring the processing delays at any relay node on the Internet; 2) we characterize the workload patterns of a variety of Internet relay nodes; and 3) we show that, serious VoIP quality degradation may occur due to relay processing, thus we have to monitor the processing delays of a relay node continuously to prevent the application performance from being degraded.
IMPROVING IPV6 ADDRESSING TYPES AND SIZEIJCNCJournal
This document discusses proposed modifications to IPv6 addressing types and address size. It suggests that multicast addressing can mimic anycast and limited broadcast addressing, making those types unnecessary. It also proposes reducing the IPv6 address size from 128-bits to decrease packet overhead, while ensuring the new size supports future internet growth. A formula is presented to predict IP address exhaustion dates for different address sizes based on current usage and population projections.
This document contains block diagrams and specifications for a Quanta Computer Inc. motherboard project. It includes:
1) A block diagram of the system showing the main components like the Ivy Bridge processor, Panther Point PCH, memory, graphics, SATA, USB, networking ports.
2) Specifications for the power delivery including voltages supplied to different components and the power states of those components.
3) A more detailed block diagram of the Ivy Bridge processor showing the connections to the PCH and graphics via DMI and PCIe interfaces.
4) Signal specifications and design notes for components like the processor, display ports, and graphics compensation.
The document describes DHCP snooping Option 82 configuration examples. It provides an overview of Option 82, which allows DHCP servers to assign IP addresses based on location information added to client requests. It then gives an example configuration where a DHCP snooping device adds unique Option 82 information to requests from three client groups, allowing the DHCP server to assign each group a separate address range from the 192.168.10.0/24 network. The DHCP snooping device and server are configured to implement this address assignment based on Option 82.
1. UDP is used for voice and video traffic instead of TCP because TCP introduces delays that break data streams and UDP does not have mechanisms for retransmitting lost packets. TCP is preferred for transmitting data files because it is more reliable and requires lost packets to be retransmitted.
2. To facilitate secure intranet access for remote workers, an enterprise IT department would use VPN and user authentication.
3. The purpose of the Cisco Enterprise Architecture is to divide the network into functional components while still maintaining the concept of Core, Distribution, and Access Layers.
This document discusses local area network (LAN) technologies, with a focus on Ethernet. It outlines the following objectives:
- Briefly discuss dominant wired LANs including Ethernet and other media types.
- Describe Media Access Control (MAC) and Carrier Sense Multiple Access with Collision Detection (CSMA/CD).
- Explain the Address Resolution Protocol (ARP) and bridges.
- Discuss switched Ethernet and virtual LANs (VLANs).
The document then provides details on Ethernet frames, MAC addresses, CSMA/CD, cabling standards and specifications.
The document discusses bandwidth requirements for IP telephony solutions in Bangladesh. It examines factors that influence bandwidth needs like codecs, sample rates, packet overhead. The G.729A codec requires the least bandwidth at 16 kbps per call. To serve a 50 seat call center would require a 800 kbps connection. The government needs to plan adequate bandwidth to support growing IP telephony demands.
This document provides an overview of a networking lesson that teaches participants how to analyze and troubleshoot common IP and ARP problems using Wireshark. The lesson covers ARP and IP addressing, fragmentation issues, routing problems, duplicate IP addresses, and DHCP configuration errors. Specific troubleshooting techniques are demonstrated, such as using ARP requests to diagnose connectivity problems, analyzing packet captures to find fragmented packets, and identifying duplicate IP addresses through gratuitous ARP messages. The overall goal is for participants to gain skills in isolating and resolving basic IP and ARP issues using network analysis tools like Wireshark.
THE FIGHT AGAINST IP SPOOFING ATTACKS: NETWORK INGRESS FILTERING VERSUS FIRST...ijsptm
The IP(Internet Protocol) spoofing is a technique that consists in replacing the IP address of the sender by
another sender’s address. This technique allows the attacker to send a message without being intercepted
by the firewall. The most used method to deal with such attacks is the technique called "Network Ingress
Filtering". This technique has been used, initially, forIPv4 networks, but its principles, are currently
extended toIPv6 networks.Unfortunately, it has some limitations, the main is its accuracy. To improve
safety conditions, we applied the "First-Come First-Serve (FCFS)" technique, applied for IPV6 networks,
and developed by the "Internet Engineering Task Force (IETF)" within its working group "Source Address
Validation Improvements (SAVI)", which is currently being standardization. In this paper, we remember
the course of an attack by IP Spoofing and expose the threats it entails.Then, we explain the "Network
Ingress Filtering" technique. Next, We present the FCFS SAVI method and methodology that we have
adopted for its implementation.Finally, we, followingthe results, discuss and compare the advantages,
disadvantages andlimitations of the FCFSSAVI methodto thoseknown in the "Network Ingress Filtering"
technique. FCFS SAVI method is more effective than the technique of "Network Ingress Filtering", but
requires some improvements, for dealing with limitations it presents.
7.2.1.8 lab using wireshark to observe the tcp 3-way handshakegabriel morillo
The PC host initiates a TCP three-way handshake with the Google web server to establish a connection. The PC's IP address is its source address, and its MAC address can be found using ipconfig. Frame 15 shows the DNS query from the PC to resolve www.google.com. Frame 16 is the response from the DNS server with the IP address. Frame 17 shows the flags set in the initial request from the PC to start a session. The source port is classified as ephemeral and the destination port is classified as well-known.
Leonardo Nve Egea - Playing in a Satellite Environment 1.2Jim Geovedi
This document discusses techniques for intercepting unencrypted satellite communications. It begins by providing background on satellite types and orbits, as well as common transmission standards. It then describes how to capture satellite signals using a DVB card and Linux tools. Specific techniques covered include identifying packet IDs to create virtual interfaces, DNS spoofing, TCP hijacking, and attacking GRE tunnels. The document explains how these techniques could allow intercepting passwords, cookies, emails and other sensitive transmitted data. It emphasizes that while uplink data cannot be captured from satellites, attacking protocols like GRE could enable some uplink sniffing.
Abstract: The Dynamic Host Configuration protocol (DHCP) is a protocol that is designed to help in automate the process of IP configuration and the rest of network parameters to the host in the network. The DHCP has a unique and important features which are make its address administration very efficient especially nowadays with the proliferation of mobile devices with the patterns that have a transient network access. With a large network or with a mobile ad-hoc network, the administrator will face an impossibility to configure the IP and the rest of network parameters of the host in the network because there will be many wrongs or there will be no infra-structure.
One of the most important features of DHCP is that the same IP will not be allowed to be used at the same time between two hosts or network cards in DHCP mechanism.
The misconfigurations or misbehavior of the host will prevent the DHCP to work properly. Our focus in this paper is to discuss the address administration of DHCP over performance and vulnerabilities in operational networks today. Moreover, we will try to display how the misconfigurations of the host could affect the DHCP and how we will be able to get rid or reduce these misconfigurations.
This document summarizes the study of parameters that determine the quality of service of various Voice over IP (VoIP) clients. The study measured parameters like bandwidth requirement, delay, packet size and observed how clients behaved under different network conditions. Key findings were that bandwidth, jitter, latency and packet loss most affected quality of service. The VoIP clients tested included Google Talk, Skype, VQube, Windows Live Messenger and Yahoo Voice Messenger. Network Address Translation (NAT) types and Simple Traversal of UDP through NAT (STUN) were also explained.
The document discusses different types of Ethernet cables including straight-through cables, crossover cables, and rollover cables. Straight-through cables connect pins on one end of the cable to the same pins on the other end, allowing connection between a computer and a switch, hub, or another computer. Crossover cables have crossed wire pairs and are used to connect like devices such as two computers or two routers. Rollover cables have opposite wiring on each end and connect a device to a router or switch's console port for programming. The document also discusses IP addressing and subnetting concepts.
This was a talk, largely on Kamaelia & its original context given at a Free Streaming Workshop in Florence, Italy in Summer 2004. Many of the core
concepts still hold valid in Kamaelia today
SmartShare's Dynamic Quality of Service (QoS) technology automatically prioritizes time-sensitive traffic like VoIP and video conferencing to ensure flawless performance. It analyzes traffic patterns continuously and allocates extra bandwidth within fractions of a second without needing configuration. Benchmarks showed SmartShare significantly outperformed a Cisco router on a congested ADSL connection, with zero packet loss for VoIP versus 5.34% loss on Cisco, and average packet delay of 119ms versus 782ms on Cisco.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
This manual provides solid practical advice on application, implementation and, most importantly, troubleshooting Voice Over IP (VOIP) systems.
MORE INFORMATION: http://www.idc-online.com/content/practical-fundamentals-voice-over-ip-voip-21?id=151
- The document describes a project to design an acoustic modem system to allow students in a classroom to communicate without WiFi when the internet is down. It would use students' laptop sound cards and microphones.
- An objective tree was created identifying key needs of the system: being easy to use, high quality audio, and low cost. Engineering requirements were identified and tradeoff matrices were made.
- Two design options were considered: using .wav or .txt files. Simulations showed signals could be sent and received at 9600 baud without WiFi, but the received message could not be decoded. The project deadline was not met as decoding was not achieved.
Cse 318 Project Report on Goethe Institut Bangladesh Network DesignMaksudujjaman
1. The student designed a network for the Goethe-Institut in Dhaka to simulate their networking needs. The network included servers, PCs, wireless access points, switches, routers, IP phones, and other devices across three floors.
2. Physical and logical network diagrams were created showing the layout and connections between devices. Key features of the design included separate networks for each floor, wireless connectivity, remote device management, and security features.
3. A cost analysis was conducted calculating the total price of network devices, coming to a total of over 13 million BDT. Common network protocols like DNS, FTP, and SMTP were configured. The project taught the student about network concepts and skills in areas
Iwsm2014 performance measurement for cloud computing applications using iso...Nesma
This document discusses measuring performance of cloud computing applications using ISO 25010 standard characteristics. It presents a case study of a private cloud hosting a Microsoft Exchange application. The study collected performance log data from nodes over one week. It analyzed the data focusing on the time behavior characteristic. It calculated statistics on measures like transmission rate and created a performance index to identify peaks and valleys in system performance over time. The study demonstrated mapping measures to ISO characteristics but noted challenges in data collection, processing and representation for large cloud infrastructures.
CASE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS IN NON-MOBILITY SCENARIOSijcsity
IEEE 802.11 is the most popular standard for WLAN networks. It offers different physical transmission
rates. This paper focuses on this multi transmission rate of 802.11 WLANs and its effect on speech quality.
In non-adaptive systems, when the physical layer switches from a higher transmission rate to a lower one,
different than the one that the VoIP flow needs, the switching may result in congestion, high delay and
packet loss, and consequently speech quality degradation. However, there are some algorithms that adapt
the transmission parameters according to the channel conditions. In this study we demonstrate how
choosing parameter (different codec and packet size) can affect the voice quality, network delay and packet
loss. Further, this study presents a comparison between adaptive and non-adaptive methods. The adaptive method has also been evaluated for different congestion level from perceived speech quality point of view.
This document provides a template for a written assessment submission. It includes instructions for students to fill in their name, student number, and to respond to assignment questions without directly copying from other sources. The template is divided into multiple pages with space for responses. Students are asked to complete tasks such as viewing IP and MAC addresses on their computer and converting them to binary, using ping commands to test network reachability, drawing a basic network diagram, and answering questions about networking concepts. They are instructed to cite any external sources used and are warned that directly copying content will result in lost marks.
“Let me show you how OMEGAMON XE for Mainframe Networks can help isolate problems to the application, z/OS Communications Server or the network.”
Network:
• Routers
• Switches
• WAN links
OMEGAMON XE for Mainframe Networks:
- Monitor all layers of the network stack
- Correlate application, z/OS and network data
- Isolate problems to application, z/OS or network
- Integrate with z/OS monitoring for end-to-end views
Network systems programmer:
"That sounds very helpful. Please show me some examples of how it can help isolate problems."
What do data center operators need to know when deploying Hadoop in the Data Center? Multi-tenancy, network topology, workload types, and myriad other factors affect the way applications run and perform in the data center. Understanding performance characteristics of the distributed system is key to not only optimize for Hadoop, but allows Hadoop to seamlessly operate side-by-side existing applications.
This lab uses Riverbed Modeler to simulate a home network with three PCs connected to the internet through different connection types. Four scenarios are created by changing the download speed of the WAN link to the internet from 20 kbps, 40 kbps, 512 kbps, and 1.544 Mbps (T1 line). The results show that higher speed connections significantly reduce WAN link utilization and improve web application response times. However, there is little benefit to upgrading from 512 kbps to a T1 line for the current network usage. Comparing all the scenarios provides insight into how performance is affected by varying the WAN link data rate.
Internet2: VoIP Phone Codec Testing White PaperJoshua Reola
This document summarizes a study that tested the effects of packet loss on voice quality for various Voice over IP (VoIP) codecs. Key findings include:
- The G.711 codec maintained acceptable audio quality (MOS >3.0) up to 5% packet loss, more than typically claimed by vendors.
- The G.729 codec degraded more quickly above 5% packet loss.
- The GIPS codec significantly outperformed standard codecs, maintaining quality up to 15% packet loss.
- There were no significant differences between vendors in codec performance. The study recommends the G.711 codec for networks with ample bandwidth and considering the GIPS codec. Further research on additional codecs and factors like packet jitter was advised
The document describes a student project to implement the User Datagram Protocol (UDP) in hardware using two FPGA development boards. The goals were to include a proper checksum calculation, demonstrate multiplexing and demultiplexing of ports, and introduce errors. UDP segments were sent between the boards using a simple bus protocol to simulate network transmission. The project provides a platform for exploring how UDP works and customizable hardware implementation of transport protocols.
The document discusses network integration considerations for Hadoop data centers. It addresses traffic types, job patterns, network attributes, architecture, availability, capacity, flexibility, management and visibility. It provides examples of buffer usage on switches and recommendations for dual 1GbE or 10GbE NIC configuration for Hadoop servers.
This document provides best practices for implementing SIP with the Aspect Unified IP environment. It defines key SIP terms and components. It describes the SIP module hierarchy and outlines steps to configure the Server Configurator with machine names, IP addresses, SIP web services, Aspect SIP proxies, and TAs. It emphasizes adding all SIP service machines to TA host files for proper call setup and resolution.
end to end delay performance analysis of video conferencing over lteINFOGAIN PUBLICATION
Mental development to use the data, such as multimedia, video and online games led to the development of a technique called LTE long term evolution. The goal of this paper is to analyze the quality of service (QoS) performance and its effects when video is streamed over LTE .Using OPNET (Optimized Network Engineering Tool). the performance can be simulated having Different scenarios for video conferencing . in addition to we also measured the performance of packet End-to-End delay .
nZDM (Near Zero Downtime Maintenance) and nZDT (Near Zero Downtime Technology) are methods to minimize downtime during SAP system updates and conversions. nZDM reduces downtime for SUM updates by performing more tasks while the system is still available, though it increases runtime. nZDT is a service offered by SAP that uses cloning to perform conversion tasks on a clone system during uptime and only requires a brief downtime for final switching to the new system. While both help lower downtime, nZDT can achieve even lower business downtimes of 8-16 hours depending on validation needed.
Microsoft RemoteFX promises to enhance the user QoE for rich media applications running on remote desktops and IPQ can be a key technology to help deliver on that promise.
5 maximazing networkcapacity_v4-jorge_alvaradoSSPI Brasil
This document discusses how to maximize network capacity through bandwidth optimization and data compression techniques. It provides an agenda that covers defining wireless link optimization, maximizing network capacity for internet access, VPN networks, UDP traffic, corporate applications, and cellular backhaul. Specific scenarios and case studies are presented where XipLink's optimization solutions have reduced bandwidth usage by 18-60% for various application types including internet, VPNs, VoIP, video surveillance, and file transfers. The solutions provide a typical return on investment of less than 4 months.
1. NETW320 Lab for WAN Codec Pg. 1
Lab # 7 Citrix Platform
Greg Pubill
Professor Noel Broman
December 14, 2014
Required Lab Questions for NETW320, Codec Selection for Campus Network (This section is
worth 75% of your grade for this lab.)
1. In the Results browser, expand FTP and select Download Response Time (sec). Change the
view from As Is to time_average and select the Show button. Zoom in on the last two-thirds of
the graph to eliminate start-up oscillation time and to get better granularity of the results. Copy
and label this graph to your lab report. Then use this graph to answer the following questions.
1. For the G.711 run, estimate the FTP response time.
The G.711 FTP response time was 17.2 seconds
2. For the G.729 run, estimate the FTP response time.
The G.729 FTP response time was 15.2 seconds
3. For the G.723.1 run, estimate the FTP response time.
The G.723 FTP response time was 23.7 seconds
FTP Download Response Time Graph
2. Close the FTP tree. Expand E-mail and select Download Response Time (sec). Change the
view from As Is to time_average and select the Show button. Zoom in on the last two-thirds of
the graph to eliminate start-up oscillation time and to get better granularity of the results. Copy
and label this graph to your lab report. Then use this graph to answer the following questions.
4. For the G.711 run, estimate the E-mail Download response time.
2. NETW320 Lab for WAN Codec Pg. 2
Lab # 7 Citrix Platform
Greg Pubill
Professor Noel Broman
December 14, 2014
The G711 email download response time was: 14.9 seconds
5. For the G.729 run, estimate the E-mail Download response time.
The G729 email download response time was: 11.3 seconds
6. For the G.723.1 run, estimate the E-mail Download response time.
The G723 email download response time was: 18.9 seconds
Email Download Response Time Graph
3. NETW320 Lab for WAN Codec Pg. 3
Lab # 7 Citrix Platform
Greg Pubill
Professor Noel Broman
December 14, 2014
3. Close the E-mail tree. Expand Voice and select Traffic Received (bytes/sec). Change the view
from As Is to time_average and select the Show button. Zoom in on the last two-thirds of the
graph to eliminate start-up oscillation time and to get better granularity of the results. Copy and
label this graph to your lab report. Then use this graph to answer the following questions:
7. For the G.711 run, estimate the Traffic Received in bytes/sec.
The traffic received for G.711 was: 91,388 bytes/sec
8. For the G.729 run, estimate the Traffic Received in bytes/sec.
The traffic received for G.729 was: 11,425 bytes/sec
9. For the G.723.1 run, estimate the Traffic Received in bytes/sec.
The traffic received for G.723 was: 7,673 bytes/sec
Bytes Received Graph
4. Go to results and select Compare Results. Expand Voice and select Packet End-to-End Delay
(sec). Change the view from As Is to time_average and select the Show button. Zoom in on the
last two-thirds of the graph to eliminate start-up oscillation time and to get better granularity of the
results. Copy and label this graph to your lab report. Then use this graph to answer the following
questions.
10. For the G.711 run, estimate the End-to-End Delay for voice packets. Then use this answer
to get the overall delay by taking into account the frame size delay and look ahead delay as
outlined in the introduction. Note: a common mistake is to add millisecond to seconds. Be sure
you convert your data to the same units (i.e., add seconds to seconds or milliseconds to
milliseconds).
Overall delay = End-to-End delay + Frame Size delay + Look Ahead delay
263 + 0.125 + 0 = 263.125 milliseconds = Total G.711 overall end-to-end delay
4. NETW320 Lab for WAN Codec Pg. 4
Lab # 7 Citrix Platform
Greg Pubill
Professor Noel Broman
December 14, 2014
11. For the G.729 run, estimate the End-to-End Delay for voice packets. The use this answer
to get the overall delay by taking into account the frame size delay and look ahead delay as
outlined in the introduction. Be sure you convert your data to the same units (i.e., add seconds to
seconds or milliseconds to milliseconds).
240 + 10 + 5 = 255 milliseconds = Total G.729 overall end-to-end delay
12. For the G.723.1 run, estimate the End-to-End Delay for voice packets. Then use this
answer to get the overall delay by taking into account the frame size delay and look ahead delay
as outlined in the introduction. Be sure you convert your data to the same units (i.e., add seconds
to seconds or milliseconds to milliseconds).
373 + 30 + 7.5 = 410.5 milliseconds = Total G.729 overall end-to-end delay
End-to-End Delay Graph
5. Go to results and select Compare Results. Expand Voice and select Packet Delay Variation.
Change the view from As Is to time_average and select the Show button. Zoom in on the last
two-thirds of the graph to eliminate start-up oscillation time and to get better granularity of the
results. Copy and label this graph to your lab report. Then use this graph to answer the following
questions:
13. For the G.711 run, estimate the Packet Delay Variation for voice packets.
The jitter total was: 0.178 for G.711
14. For the G.729 run, estimate the Packet Delay Variation for voice packets.
The jitter total was: 0.160 for G.729
15. For the G.723.1 run, estimate the Packet Delay Variation for voice packets.
The jitter total was: 0.379 for G.723
Voice Packet Delay Variation (Jitter) Graph
5. NETW320 Lab for WAN Codec Pg. 5
Lab # 7 Citrix Platform
Greg Pubill
Professor Noel Broman
December 14, 2014
6. NETW320 Lab for WAN Codec Pg. 6
Lab # 7 Citrix Platform
Greg Pubill
Professor Noel Broman
December 14, 2014
Required Lab Summary Report for NETW320, Codec Selection for Campus Network
Which of the codecs used in this lab would you recommend that the network administrator
implement for the organization under analysis?
The results of this simulation, support the performance values listed and recommended by the
ITU the tables (see tables on last page of report). From these results it can be concluded that
the network shall use the G.729 codec for VoIP communication. This codec will have minimal
email, ftp and http traffic impact, it will also result in acceptable delay and jitter ranges. It will
also result in a significant reduction of network traffic with a total of 11.5 KBps when compared
to 91 KBps generated by G.711. It is true the G.723 carries only 7.6 KBps but the delays fall in
the unacceptable range and it has the lowest MOS value. The G.729 codec should be prefered
as it still maintains an MOS value of 3.92 which is only slightly lower than the 4.1 value of the
G.711 codec.
a. How did particular codec implementations affect FTP download response time and e-
mail download response time? Were any of these delays significant enough to cause
FTP users and e-mail users to become dissatisfied?
Specifically for FTP and email response time the choice of G.729 codec actually improved the
response time. The data collected from the simulation shows that the average response time
was 15.2 for FTP and 11.3 seconds for email using this codec. The next set of results shows
that the second best performing codec was G.711. Email response time for email was 17.2
seconds and FTP response time was 14.9 which can be considered comparable to the G.729
performance. By far the worst perfoming codec was G.723 with the highest response times.
Email and FTP traffic is more tolerant than VoIP traffic. Voice traffic can tolerate response/delay
time of up to 200 ms. Delays higher than this value cause appreciable loss of voice quality, the
delays on data traffic althought they are much higher will still be tolerable in end user
experience.
b. Compare the amount of voice packets generated by the various codecs. What effect
does generating more voice packets for a particular codec have on voice quality?
Why?
Generating more voice packets could have effects that may either be desirable or undesirable
depending on the bandwidth available. Codecs that generate more voice packets will produce
sound of higher quality and less delay due to the fact that less time is spent compressing and
decompressing voice packets. However, when bandwidth is limited this codec can cause
degradation of the conversation as packets will be dropped by the UPD protocol, end users will
need continue to repeat portions of the conversation due to these dropped packets. Data user
will also notice congestion as data traffic will be delayed since TCP will continue to retransmit
until acknowledgements is received.
c. Use the overall delay you calculated for the three codecs used in this lab to rate their
overall delay times with the ITU-T benchmark outlined in the introduction to this lab.
7. NETW320 Lab for WAN Codec Pg. 7
Lab # 7 Citrix Platform
Greg Pubill
Professor Noel Broman
December 14, 2014
Do all codecs fall within recommended parameters? If not, how do you think this
delay will affect the speech quality of the VoIP traffic?
The calculated delays on this simulation show that both the G.711 and G.729 codecs fall within
the acceptable range for One-Way Overall Delay Times recommended by the ITU-T. The G.729
codec had the better performance with a delay of 255 ms, the G.711 codec resulted in a delay
of 263 ms. Since the ITU table shows that an acceptable range exist between 150 to 400 ms.
We can consider these two codecs as suitable. It is always desirable to have delays less than
150 ms in every case, however these results still fall within the range manageable by
administrators. G.723 performance would not be acceptable with a total delay of 410.5 ms.
d. Comment on the effect of your measured packet delay variation for the various
codecs analyzed in this lab. See the comments on packet delay variation (jitter)
outlined in the introduction to this lab. Which codecs are most affected by jitter in the
Internet environment? Do you believe the jitter value for any of the codecs is high
enough to degrade speech quality?
The effects of these codecs on packet-delay variation where very favorable in this simulation.
As the introduction stated a variation of more than 50 ms is considered excessive jitter because
it degrades voice quality. However, from the results we see that even the worst performing
codec which was G.723 produced a variation of only 37 milliseconds. The variation displayed by
the G.711 was 17 ms and G.729 was 16 ms which represents a similar range. All of these
values are well below the 50 ms recommendation. In terms of jitter all of these produced results
satisfactory according to the recommended value.
End