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<v ->Performance metrics.</v>
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In this lesson, we're going to talk
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all about performance metrics and how they're used
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to ensure network availability.
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Now, network performance metrics are a large part
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of network monitoring.
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Network performance monitoring is the end-to-end
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network monitoring of your end user experience.
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This differs from traditional monitoring though,
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because traditional monitoring is focused on performance
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between two points, like a switch and a router,
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but with network performance monitoring,
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we're going to look at the overall end user experience
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by monitoring the performance
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from the end user's workstation to the final destination
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that they're trying to reach.
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So to help us monitor network performance,
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there's really going to be three key metrics
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that we're going to use.
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These are latency, bandwidth and jitter.
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The first metric is latency.
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Now, latency is the measure of the time that it takes
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for data to reach its destination across a network.
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Usually we measure network latency as the round trip time
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from a workstation to the distant end
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and back to the workstation.
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We report this time in milliseconds.
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Now, for example, let's say you open up your command prompt
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and you enter the command ping 8.8.8.8, and you hit enter.
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You're going to get a response that tells you
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how long it took for an ICNP packet to leave your computer
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reach the Google DNS server located at eight dot 8.8.8.8
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and return to your computer again.
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In my case, this took an average time of 38.2 milliseconds
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when I did it for four repetitive ping requests.
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Now it's important to measure the round trip delay
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for network latency because the computer that uses
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a TCP IP network can send only a limited amount of data
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to its destination at one time,
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and then it sits and waits for an acknowledgement
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that that data was received before it sends out
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more data across the network.
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So if you have high latency or a long round trip delay,
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this can drastically slow down
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your overall network performance for your end users.
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Now, if you're seeing consistent delays
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or even just spikes in the delay time in your network,
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this could indicate a major performance issue
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that's going to be occurring.
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For regular web traffic these delays
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aren't usually noticeable,
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but if you're using streaming video applications,
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things like voiceover IP, or you're playing video games,
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these delays are extremely noticeable
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and they can cause a lot of problems for your end users.
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Our second metric media monitor is known as bandwidth.
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Now bandwidth is the maximum rate of data transfer
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across a given network.
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Now, technically bandwidth is actually
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a theoretical concept that measures how much data
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could be transferred from a source to a destination
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under ideal conditions.
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But in reality, when we're talking about our networks
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and our connections, they're rarely operating
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at the perfect or ideal conditions.
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Therefore we often measure something known as throughput
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instead of bandwidth to monitor our network performance.
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Throughput is the actual measure of data
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as it's being successfully transferred
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from the source to the destination,
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but you'll often hear people use the terms
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bandwidth and throughput interchangeably,
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not realizing there is a difference.
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Technically, bandwidth is a theoretical limit
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where throughput is the reality of what you're achieving.
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So if you want to a bandwidth speed test
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or more accurately, a throughput test for your network,
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you can go to something like speedtest.net and click on go,
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and you'll have a series of downloads and uploads
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that'll occur from your workstation
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to their server and back.
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Then it will report to you how fast or slow
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your connection was in terms of throughput.
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In this example, you can see my results indicate
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I have a throughput with a top download speed
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of 240 megabits per second,
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and a top upload speed of around 241 megabits per second.
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The problem with that is that my actual bandwidth,
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my theoretical, limit should be 650 megabits per second
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for downloads and 310 megabits per second for uploads.
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So why is my throughput so much less?
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Well, when I was doing this test,
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I connected to my office network using my wifi adapter
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and not directly connecting through a wired switch.
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At the same time, there's other people
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in the office using the connection,
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and all of these factors lead to a less than ideal
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environment and this makes my throughput much lower
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than my expected bandwidth.
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As I make different changes to my network,
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I can retest the throughput to see if those changes
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help or hurt my overall throughput.
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For example, if I switched
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from a wireless internet connection
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to a wired internet connection,
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I'll be able to see a dramatic increase
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in overall throughput that I wouldn't see
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over that wireless connection.
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The third metric we need to monitor is known as jitter.
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Jitter is the network condition that occurs
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when a time delay in the sending of the data packets
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over a network connection is occurring.
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Now jitter is really a big problem
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for any real-time applications that you may be supporting
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on your network.
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If you're doing things like video conferences
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and voiceover IP, and virtual desktop infrastructure,
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all of these are negatively affected by jitter.
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Basically a jitter is simply a variation
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in the delay of the packets.
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And this can cause some really strange side effects,
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especially for your voice and video calls.
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If you've ever been in a video conference
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and somebody starts speaking,
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and then all of a sudden you hear their voice
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start speeding up for about five or 10 seconds,
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and then it returns back to normal speed,
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that usually is because of jitter on their network.
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If you have a good quality of service management in place,
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you shouldn't experience a lot of jitter,
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but if you're not doing QoS properly,
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then jitter will occur.
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You see, when your network suffers from congestion,
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the network devices like your routers and switches
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are going to be unable to send the equivalent amount of traffic
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as what they're receiving.
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This causes their packet buffers to start to fill up,
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and eventually they'll start to drop packets
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if they have too much in the buffers.
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This is known as packet loss.
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Now, when this happens, your TCP packets
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are going to get resent, and this causes
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increased network load again.
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Now on the other hand, if the buffer begins to fill up,
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but then the network congestion eases up
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those buffers will be able to quickly send
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all of their contents to the destination.
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The destination will then try to process them all,
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but usually it can't do that.
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And this leads to delays in processing
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that can result in jitter on the endpoint device as well.
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So to prevent jitter, we want to ensure our network
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is using quality of service properly.
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We want to make sure we're categorizing
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and prioritizing our voice and video traffic over
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the other types of traffic.
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Also, we need to verify our network connections
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and our devices are large enough to support
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the amount of data that we're trying to transfer.
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As a network administrator,
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it's your responsibility to always monitor
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your network's performance and the three key metrics
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you always should be keeping track of
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are latency, bandwidth or throughput and jitter.
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(logo buzzing)
